# 24bit vs 16bit, the myth exploded!



## gregorio (May 29, 2022)

It seems to me that there is a lot of misunderstanding regarding what bit depth is and how it works in digital audio. This misunderstanding exists not only in the consumer and audiophile worlds but also in some education establishments and even some professionals. This misunderstanding comes from supposition of how digital audio works rather than how it actually works. It's easy to see in a photograph the difference between a low bit depth image and one with a higher bit depth, so it's logical to suppose that higher bit depths in audio also means better quality. This supposition is further enforced by the fact that the term 'resolution' is often applied to bit depth and obviously more resolution means higher quality. So 24bit is Hi-Rez audio and 24bit contains more data, therefore higher resolution and better quality. All completely logical supposition but I'm afraid this supposition is not entirely in line with the actual facts of how digital audio works. I'll try to explain:

 When recording, an Analogue to Digital Converter (ADC) reads the incoming analogue waveform and measures it so many times a second (1*). In the case of CD there are 44,100 measurements made per second (the sampling frequency). These measurements are stored in the digital domain in the form of computer bits. The more bits we use, the more accurately we can measure the analogue waveform. This is because each bit can only store two values (0 or 1), to get more values we do the same with bits as we do in normal counting. IE. Once we get to 9, we have to add another column (the tens column) and we can keep adding columns add infinitum for 100s, 1000s, 10000s, etc. The exact same is true for bits but because we only have two values per bit (rather than 10) we need more columns, each column (or additional bit) doubles the number of vaules we have available. IE. 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024 .... If these numbers appear a little familiar it is because all computer technology is based on bits so these numbers crop up all over the place. In the case of 16bit we have roughly 65,000 different values available. The problem is that an analogue waveform is constantly varying. No matter how many times a second we measure the waveform or how many bits we use to store the measurement, there are always going to be errors. These errors in quantifying the value of a constantly changing waveform are called quantisation errors. Quantisation errors are bad, they cause distortion in the waveform when we convert back to analogue and listen to it.

 So far so good, what I've said until now would agree with the supposition of how digital audio works. I seem to have agreed that more bits = higher resolution. True, however, where the facts start to diverge from the supposition is in understanding the result of this higher resolution. Going back to what I said above, each time we increase the bit depth by one bit, we double the number of values we have available (EG. 4bit = 16 values, 5bit = 32 values). If we double the number of values, we halve the amount of quantisation errors. Still with me? Because now we come to the whole nub of the matter. There is in fact a perfect solution to quantisation errors which completely (100%) eliminates quantisation distortion, the process is called 'Dither' and is built into every ADC on the market.

 Dither: Essentially during the conversion process a very small amount of white noise is added to the signal, this has the effect of completely randomising the quantisation errors. Randomisation in digital audio, once converted back to analogue is heard as pure white (un-correlated) noise. The result is that we have an absolutely perfect measurement of the waveform (2*) plus some noise. In other words, by dithering, *all* the measurement errors have been converted to noise. (3*).

 Hopefully you're still with me, because we can now go on to precisely what happens with bit depth. Going back to the above, when we add a 'bit' of data we double the number of values available and therefore halve the amount of quantisation error. The result (after dithering) is a perfect waveform with half the amount of noise. To phrase this using audio terminology, each extra bit of data moves the noise floor down by 6dB (half). We can turn this around and say that each bit of data provides 6dB of dynamic range (*4). Therefore 16bit x 6db = 96dB. This 96dB figure defines the dynamic range of CD. (24bit x 6dB = 144dB).

 So, 24bit does add more 'resolution' compared to 16bit but this added resolution doesn't mean higher quality, it just means we can encode a larger dynamic range. This is the misunderstanding made by many. There are no extra magical properties, nothing which the science does not understand or cannot measure. The only difference between 16bit and 24bit is 48dB of dynamic range (8bits x 6dB = 48dB) and *nothing else*. This is not a question for interpretation or opinion, it is the provable, undisputed logical mathematics which underpins the very existence of digital audio.

 So, can you actually hear any benefits of the larger (48dB) dynamic range offered by 24bit? Unfortunately, no you can't. The entire dynamic range of some types of music is sometimes less than 12dB. The recordings with the largest dynamic range tend to be symphony orchestra recordings but even these virtually never have a dynamic range greater than about 60dB. All of these are well inside the 96dB range of the humble CD. What is more, modern dithering techniques (see 3 below), perceptually enhance the dynamic range of CD by moving the quantisation noise out of the frequency band where our hearing is most sensitive. This gives a percievable dynamic range for CD up to 120dB (150dB in certain frequency bands).

 You have to realise that when playing back a CD, the amplifier is usually set so that the quietest sounds on the CD can just be heard above the noise floor of the listening environment (sitting room or cans). So if the average noise floor for a sitting room is say 50dB (or 30dB for cans) then the dynamic range of the CD starts at this point and is capable of 96dB (at least) above the room noise floor. If the full dynamic range of a CD was actually used (on top of the noise floor), the home listener (if they had the equipment) would almost certainly cause themselves severe pain and permanent hearing damage. If this is the case with CD, what about 24bit Hi-Rez. If we were to use the full dynamic range of 24bit and a listener had the equipment to reproduce it all, there is a fair chance, depending on age and general health, that the listener would die instantly. The most fit would probably just go into coma for a few weeks and wake up totally deaf. I'm not joking or exaggerating here, think about it, 144dB + say 50dB for the room's noise floor. But 180dB is the figure often quoted for sound pressure levels powerful enough to kill and some people have been killed by 160dB. However, this is unlikely to happen in the real world as no DACs on the market can output the 144dB dynamic range of 24bit (so they are not true 24bit converters), almost no one has a speaker system capable of 144dB dynamic range and as said before, around 60dB is the most dynamic range you will find on a commercial recording.

 So, if you accept the facts, why does 24bit audio even exist, what's the point of it? There are some useful application for 24bit when recording and mixing music. In fact, when mixing it's pretty much the norm now to use 48bit resolution. The reason it's useful is due to summing artefacts, multiple processing in series and mainly headroom. In other words, 24bit is very useful when recording and mixing but pointless for playback. Remember, even a recording with 60dB dynamic range is only using 10bits of data, the other 6bits on a CD are just noise. So, the difference in the real world between 16bit and 24bit is an extra 8bits of noise.

 I know that some people are going to say this is all rubbish, and that “I can easily hear the difference between a 16bit commercial recording and a 24bit Hi-Rez version”. Unfortunately, you can't, it's not that you don't have the equipment or the ears, it is not humanly possible in theory or in practice under any conditions!! Not unless you can tell the difference between white noise and white noise that is well below the noise floor of your listening environment!! If you play a 24bit recording and then the same recording in 16bit and notice a difference, it is either because something has been 'done' to the 16bit recording, some inappropriate processing used or you are hearing a difference because you expect a difference.

 G

 1 = Actually these days the process of AD conversion is a little more complex, using oversampling (very high sampling frequencies) and only a handful of bits. Later in the conversion process this initial sampling is 'decimated' back to the required bit depth and sample rate.

 2 = The concept of the perfect measurement or of recreating a waveform perfectly may seem like marketing hype. However, in this case it is not. It is in fact the fundamental tenet of the Nyquist-Shannon Sampling Theorem on which the very existence and invention of digital audio is based. From WIKI: “In essence the theorem shows that an analog signal that has been sampled can be *perfectly* reconstructed from the samples”. I know there will be some who will disagree with this idea, unfortunately, disagreement is NOT an option. This theorem hasn't been invented to explain how digital audio works, it's the other way around. Digital Audio was invented from the theorem, if you don't believe the theorem then you can't believe in digital audio either!!

 3 = In actual fact these days there are a number of different types of dither used during the creation of a music product. Most are still based on the original TPDFs (triangular probability density function) but some are a little more 'intelligent' and re-distribute the resulting noise to less noticeable areas of the hearing spectrum. This is called noise-shaped dither.

 4 = Dynamic range, is the range of volume between the noise floor and the maximum volume.


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## Dr. Strangelove

Quite an excellent write up. I am eager to see the rebuttal.


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## NiceCans

wOw!
 Very interesting read, and I expect there to be some very interesting responses.

 Thank you very much for the time, effort, and research that went into this post.


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## David_N

_Excellent_ thread! This will surely help a lot of people, I'm interested to see where this thread goes


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## Bojamijams

W00t! Thank you for that indepth explanation.. NOW I KNOW MORE!


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## Oublie

so the reality is we should all be using non oversampling dacs from a 16bit source?

 I run my computer audio at 16bit 44100khz so i guess i'm doing things right.

 Interesting reading and it makes sense. this is the first time i've read this info explained so clearly thanks.

 waiting for an opposing opinion


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## userlander

cliffs?


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## majkel

1) The Nyquist theorem does not include amplitude quantization meaning infinite resolution, so it doesn't discuss quantization effects at all.
 2) There is no audio system in the world giving more than 20 clear bits of signal due to resistance and semiconductor noise chracteristics. 
 3) You cannot "recreate" a single thing by dithering, just make it sounding more natural to the ears, especially when using noise shaping filters for the dither signal. 
 4) It's not true ADC's do any dithering. Some of them do some lowpass filtering with noise shaping involved when delta-sigma type which happens not for the dithering purposes but for the signal itself.
 5) You cannot increase dynamic range by dithering.
 6) It's not dynamics killing people and affecting hearing but sound pressure with the given numbers of 140dB = pain, 160~180dB = death, respectively. You can listen to the signal with the 144dB dynamics not exceeding safe sound pressure limits, just set the volume appropriately. Sure, you won't hear the bottom of your dynamic range then.


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## gregorio

Quote:


  Originally Posted by *Oublie* /img/forum/go_quote.gif 
_so the reality is we should all be using non oversampling dacs from a 16bit source?_

 

In my original post I only really dealt with bit depth rather than with sampling rates. Bit depth is relatively simple to explain and is entirely predictable both in theory and in practice. Sampling rates are not so simple to explain in practice, as the defining feature of sampling rates is the anti-alias filter which has to be used. How these filters are employed from one model of ADC to another and indeed how the signal is re-constructed back to analogue in a DAC varies and can be quite complicated to understand. The person I learnt from is called Nika Aldrich and is regarded as one of the world's leading authorities on digital audio. If you want to understand about how these filters work here is a link to probably the best paper written on the subject: Digital Audio Explained

 In short, oversampling DACs can (possibly) make a difference to how one perceives the audio quality. This refers of course only to the sampling rate side of things. Increasing the bit depth will not make any difference as explained in my original post.

 Interestingly, a very good authority on digital audio is Dan Lavry who has his own forum here on Head-Fi. He published a very well regarded paper on sampling rates, if you want to know more about this side of digital audio: http://www.lavryengineering.com/docu...ing_Theory.pdf

 G


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## majkel

I see Nika Aldrich didn't explain the oversmapling reason. The purpose of oversampling is to shorten the output "steps" of the DAC. When you look at the Shannon-Kotielnikov theorem, you have perfect analog recreation when *output impulses are indefinitely short*. Instead of it, you get "bars" neighbouring each other, without silence gaps between them. So, after lowpass filtering, instead of the original signal you obtain the signal with obvious sin(x)/x bandwidth distortion. This kind of distortion makes the treble response rolled-off which some audiophiles call "musical" because of less piercing highs on inexpensive equipment but actually it's further from the original than oversampled signal. 
 For oversampling, you need a FIR filter moving ultrasonic content of the DAC output to much higher frequencies. This will guarantee the "bars" won't be of equal amplitude ahd thus work for the sin(x)/x distortion removal in the audible bandwidth. Another advantage is that you need milder lowpass filters after the DAC, inducing less phase distortion.


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## Lamenthe

Very interesting thread, I'll be following the discussion between majkel and gregorio attentively


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## gregorio

Quote:


  Originally Posted by *majkel* /img/forum/go_quote.gif 
_1) The Nyquist theorem does not include amplitude quantization meaning infinite resolution, so it doesn't discuss quantization effects at all.
 2) There is no audio system in the world giving more than 20 clear bits of signal due to resistance and semiconductor noise chracteristics. 
 3) You cannot "recreate" a single thing by dithering, just make it sounding more natural to the ears, especially when using noise shaping filters for the dither signal. 
 4) It's not true ADC's do any dithering. Some of them do some lowpass filtering with noise shaping involved when delta-sigma type which happens not for the dithering purposes but for the signal itself.
 5) You cannot increase dynamic range by dithering.
 6) It's not dynamics killing people and affecting hearing but sound pressure with the given numbers of 140dB = pain, 160~180dB = death, respectively. You can listen to the signal with the 144dB dynamics not exceeding safe sound pressure limits, just set the volume appropriately. Sure, you won't hear the bottom of your dynamic range then._

 

1. True.
 2. True. Most people believe that their 24bit DAC is actually a 24bit DAC, just marketing I'm afraid.
 3. True. Dithering is just a process which should be used whenever a quantisation or re-quantisation is performed, to convert quantisation errors into un-correlated noise.
 4. This one is not true. All ADCs use dither. Some 24bit ADCs use self-dither, in other words because the digital noise floor is so low (-144dB) the noise generated by their own internal components is enough to dither, but one way or another, they all dither. Also, *all* ADCs use a low-pass brick wall filter (anti-alias filter). Noise-shaped dither is not and should never be used in an ADC or when mixing. As the recorded channels are mixed the re-distributed noise is summed and can cause problems. The only time noise-shaped dither should be applied is during the last quantisation process. This usually means when converting the 24bit master from the recording studio into 16bit for CD release.
 5. Sort of true. In an absolute sense CD has 96dB dynamic range, however if we move the noise that is down at the -96dB level to areas of the hearing spectrum where we are less sensitive (for example below 60Hz or above 12kHz). This gives a perceived improvement of dynamic range for 16bit. Bob Katz, the leading expert, reckons that about 120dB is the perceived dynamic range achievable with today's dithering technology.
 6. True. Though of course by turning down your amp and not hearing the quietest sounds, then you are not hearing all the detail or the whole dynamic range, so it rather defeats the whole purpose of more dynamic range (more bits) in the first place.

 G


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## majkel

Oops, in point 4. you are obviously right as I was thinking about DAC's.


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## linuxworks

slight diversion but its related to the main ideas, here.

 people often worry about the DIGITAL side of things thinking there is 'god stuff' that occurred before the encoding from analog to digital (at the studio).

 guess what - most musicians and engineers do NASTY things to your 'perfect sound' way before its even in the digital domain.

 many years ago, I was toying around with pro audio and I was learning about 'compressors'. I bought a thing called the RNC (really nice compressor). it was a few hundred dollars and the pros raved about it. but it was an ANALOG COMPRESSOR!

 people get all nutty about op-amps 'in the path' but how many pros are 100% discrete class A in their path?

 probably none.

 people are fussing here more than most pros who CREATE the music are fussing.

 they use compressors in their chain.

 does that blow your mind? 
	

	
	
		
		

		
		
	


	




 or at least temper the 'no op amps!' mantra I hear all too often.

 way before you are at 16/24 bits - you are 'destroying the sound' in compressors, equalizers and other 'effects boxes'. very few recordings are untouched and recorded with no processing at all.

 it sometimes helps to understand where the data comes from and not just assume god dropped it on your DAC for you


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## gregorio

Quote:


  Originally Posted by *linuxworks* /img/forum/go_quote.gif 
_slight diversion but its related to the main ideas, here.

 people often worry about the DIGITAL side of things thinking there is 'god stuff' that occurred before the encoding from analog to digital (at the studio).

 guess what - most musicians and engineers do NASTY things to your 'perfect sound' way before its even in the digital domain.

 many years ago, I was toying around with pro audio and I was learning about 'compressors'. I bought a thing called the RNC (really nice compressor). it was a few hundred dollars and the pros raved about it. but it was an ANALOG COMPRESSOR!

 people get all nutty about op-amps 'in the path' but how many pros are 100% discrete class A in their path?

 probably none.

 people are fussing here more than most pros who CREATE the music are fussing.

 they use compressors in their chain.

 does that blow your mind? 
	

	
	
		
		

		
		
	


	




 or at least temper the 'no op amps!' mantra I hear all too often.

 way before you are at 16/24 bits - you are 'destroying the sound' in compressors, equalizers and other 'effects boxes'. very few recordings are untouched and recorded with no processing at all.

 it sometimes helps to understand where the data comes from and not just assume god dropped it on your DAC for you 
	

	
	
		
		

		
		
	


	


_

 

Partly true. These days most processing occurs in the digital domain, using plug in compressors, etc. Where outboard gear is used, it obviously depends on the studio, a world class studio will of course use the very best compressors/EQ and they will certainly be Class A.

 Also, we don't use compressors or other processing to destroy the sound quality, generally we use it for the opposite reason. For instance to correct the EQ of a recorded track to help with separation. Another example; usually recorded vocals have quite a wide dynamic range, even from syllable to syllable, when we try to mix this in the with rest of the track bits of the vocal are slightly too loud and bits of it too quiet, compression equals out these variations so we can hear a nice present vocal line. 

 As a general rule the more processing done, the more the sound quality suffers. So one of the main concerns for the producer is to balance the amount of processing to improve separation (and other factors) and the amount of processing which too negatively affects the SQ. Production is not an exact science, it is an art and virtually always involves some level of compromise.

 G


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## ILikeMusic

Quote:


  Originally Posted by *gregorio* 
_I know that some people are going to say this is all rubbish, and that “I can easily hear the difference between a 16bit commercial recording and a 24bit Hi-Rez version”._

 

Ya think..?


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## mark_h

Very interesting thanks. I use R2R DACs or Vinyl so guess I'm safe...phew!


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## linuxworks

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Partly true. These days most processing occurs in the digital domain, using plug in compressors, etc. Where outboard gear is used, it obviously depends on the studio, a world class studio will of course use the very best compressors/EQ and they will certainly be Class A._

 

but what about my treasured beatles collection? that era of music - what tech was used back then? what was 'average' and what was 'world class' ?

 my collection is more of stuff that existed in the analog days and very little that I listen to comes from 'today'. when I hear the multiple levels of hiss that sometimes accompany the start of a song, I realize that the gear I have now is way better than the stuff they used to CREATE it on, in the first place. my noise floor is below theirs!

 so my point is, no amount of worrying if a dac is 16bit or 24bit or even if the analog was copied IN 16bit or 24bit format - the original is still 'lossy as hell' compared to even mid-fi op amp specs of today.

 and what exactly does getting 'better resolution' buy you? on a hissy distorted (by today's standards) source will sound just as bad, but just in higher resolution so you can hear *more* of the hiss and noise and distortion (lol).

 too much worrying about 'the last mile', imho. your playback gear is almost always better than the combo of what was finally mixed and released. certainly true for older material.


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## gregorio

Quote:


  Originally Posted by *mark_h* /img/forum/go_quote.gif 
_Very interesting thanks. I use R2R DACs or Vinyl so guess I'm safe...phew!_

 

Using 24bit or even using an upsampling DAC and going from 16bit to 24bit is not likely to be detrimental, it's just that there won't be any benefit either.

 Going the other way, 24bit to 16bit could be quite detrimental unless a good quality dither is used as part of the process. Most consumer programs will truncate when going from 24 to 16bit. In other words, the last 8bits are just hacked off. Truncation is not good, it introduces quantisation distortion which is correlated to the audio material and it's results are unpredictable. It could mean that you get unwanted tones or harmonics in the mix which may be noticeable. Some consumer programs 'round' the result, still not good but better than truncation. The effects of rounding are unlikely to be heard by most people but the chances are that some audiophiles would notice. Dither is the only real option if you are serious about SQ.

 G


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## krmathis

Really interesting! **bookmarked**


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## gregorio

Quote:


  Originally Posted by *linuxworks* /img/forum/go_quote.gif 
_but what about my treasured beatles collection? that era of music - what tech was used back then? what was 'average' and what was 'world class' ?

 my collection is more of stuff that existed in the analog days and very little that I listen to comes from 'today'. when I hear the multiple levels of hiss that sometimes accompany the start of a song, I realize that the gear I have now is way better than the stuff they used to CREATE it on, in the first place. my noise floor is below theirs!

 so my point is, no amount of worrying if a dac is 16bit or 24bit or even if the analog was copied IN 16bit or 24bit format - the original is still 'lossy as hell' compared to even mid-fi op amp specs of today.

 and what exactly does getting 'better resolution' buy you? on a hissy distorted (by today's standards) source will sound just as bad, but just in higher resolution so you can hear *more* of the hiss and noise and distortion (lol).

 too much worrying about 'the last mile', imho. your playback gear is almost always better than the combo of what was finally mixed and released. certainly true for older material._

 

True to an extent. A fair bit of the Beatles stuff was done at Abbey Road Studios, which was and is still one of the best studios in the world. Abbey Road is a multi-million pound facility which uses the very highest quality audio equipment on the market and is often used as a test bed for the very high end manufacturers, Neve for example. One of the Beatles albums (think it was Sargent Pepper) is about the first example of multitrack recording being used for a commercial product. Having said all this, was the SQ possible in Abbey Road Studios in the '60s as good as modern replay systems? The answer in my opinion is probably not that far off a good system today! However, unless you can get hold of the original master (fat chance!) the copies available are likely to sound quite weak compared to modern recordings.

 If transferring a vinyl to digital, there may be some merit in using 24bit, for the same reason as using 24bit when recording in a studio. 24bit gives you tons of headroom as usually you try to set peak levels in 24bit to -18dB (-22dB is also a good peak level). This gives you plenty of space for any transient 'overs' which you might get using 16bit. You can always dither back down to 16bit when you're done or leave it at 24bit if storage space is not an issue.

 G


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## lamikeith

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_It seems to me that there is a lot of misunderstanding regarding what bit depth is and how it works in digital audio._

 

This is a great post. Nice to see these concepts explained in a way that I can understand easily.

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I know that some people are going to say this is all rubbish, and that “I can easily hear the difference between a 16bit commercial recording and a 24bit Hi-Rez version”._

 

It is my understanding that some (many?) recordings that are released in both 16 bit and 24 bit are not the same. The 24 bit release may have been produced and/or mastered differently. Also, there may be differences in the digital to audio conversion process within the end-users gear to consider. So it may well be true that people hear differences, but it will not be due to bit depth alone.


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## gregorio

Quote:


  Originally Posted by *lamikeith* /img/forum/go_quote.gif 
_It is my understanding that some (many?) recordings that are released in both 16 bit and 24 bit are not the same. The 24 bit release may have been produced and/or mastered differently. Also, there may be differences in the digital to audio conversion process within the end-users gear to consider. So it may well be true that people hear differences, but it will not be due to bit depth alone._

 

It is possible under certain circumstances that a difference could be heard. This is because 24bit releases also often have a higher sample frequency (say 96kFs/s). With certain equipment and certain music it may be possible to notice a marginal difference with the higher sample frequency. The potential difference is mainly in the ADC, where a relatively cheap ADC has been used which possibly has a poor implementation of the anti-alias filter. Upping the sample frequency in this case uses a much smoother anti-alias filter with fewer artefacts and it maybe possible, with very good hearing and a good system, to hear the effects of a better implemented filter at 96kFs/s. However, if the music was recorded with a high end professional ADC, the filters at 44.1kFs/s are generally much better implemented and then telling 44.1k from 96k is much more difficult (read impossible) regardless of equipment and hearing ability.

 AFAIK, there has never been a DBT between 16bit and 24bit (under controlled conditions) using the same sample frequency, where anyone has been able to tell the difference with any more accuracy than would be expected from chance.

 Also, what you said about the 24bit and 16bit releases is entirely true. There is no way of knowing the processes that each mix has gone through or even by who has done it and with what degree of care. EG. The Studio, the Mastering Engineer, The Record label even an assistant in one of these businesses who is effectively stealing!

 G


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## manaox2

Quote:


  Originally Posted by *mark_h* /img/forum/go_quote.gif 
_Very interesting thanks. I use R2R DACs or Vinyl so guess I'm safe...phew!_

 

Your never safe from the recording engineer, especially in today's music.


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## fjf

My bad!....There goes the SACD myth down the drain....We just saved a whole bunch of money...


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## linuxworks

Quote:


  Originally Posted by *manaox2* /img/forum/go_quote.gif 
_Your never safe from the recording engineer, especially in today's music.
	

	
	
		
		

		
		
	


	


_

 

NO one is safe from the recording engineer. his chief weapon is surprise...surprise and fear...fear and surprise.... his two weapons are fear and surprise...and ruthless efficiency.... his *three* weapons are fear, surprise, and ruthless efficiency...and an almost fanatical devotion to removing all impurities from the signal chain.


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## geremy

I don't know anything about anything, so I am not criticizing the initial arguement, but I don't follow it so much either.

 I'm used to working with Volts. So say you have a 16-bit ADC with a reference voltage and your rail-to-rail voltage is 0.5 to 4.5V. Then each bit of the 2^16 possible combination represents a 4V/2^16 value in volts (in this case 6.1e-5 volts). If you use the same rail-to-rail range with a 24-bit ADC, each bit represents 4/2^24 volts or 2.38e-7 volts (with the same sample clock).

 It seems like the original post is stating that each bit can only represented a fixed amount (in my case volts), and what increases by going from a 16-bit to a 24-bit ADC is they rail-to-rail measurable voltage. While it is true that this is possible, it is also possible measure the same voltage swing with increased resolution. I believe I am misunderstanding the original post.

 Thanks.


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## gregorio

Quote:


  Originally Posted by *linuxworks* /img/forum/go_quote.gif 
_NO one is safe from the recording engineer. his chief weapon is surprise...surprise and fear...fear and surprise.... his two weapons are fear and surprise...and ruthless efficiency.... his *three* weapons are fear, surprise, and ruthless efficiency...and an almost fanatical devotion to removing all impurities from the signal chain. 
	

	
	
		
		

		
		
	


	


_

 

And now for something completely different 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *geremy* /img/forum/go_quote.gif 
_I don't know anything about anything, so I am not criticizing the initial arguement, but I don't follow it so much either.

 I'm used to working with Volts. So say you have a 16-bit ADC with a reference voltage and your rail-to-rail voltage is 0.5 to 4.5V. Then each bit of the 2^16 possible combination represents a 4V/2^16 value in volts (in this case 6.1e-5 volts). If you use the same rail-to-rail range with a 24-bit ADC, each bit represents 4/2^24 volts or 2.38e-7 volts (with the same sample clock).

 It seems like the original post is stating that each bit can only represented a fixed amount (in my case volts), and what increases by going from a 16-bit to a 24-bit ADC is they rail-to-rail measurable voltage. While it is true that this is possible, it is also possible measure the same voltage swing with increased resolution. I believe I am misunderstanding the original post.

 Thanks._

 

I'm not sure I fully understand your question. The actual window of voltage variations represented by the 16bit (or 24bit window) is not fixed. For example, many ADCs are calibrated so that 0dBu which equals 0.775v (line level) registers as -18dBFS (dBFS meaning the digital Full Scale where 0dBFS is the maximum value of all bits set to 1). However, some systems are calibrated to 0dBv = -14dBFS and in film it's often set to -20 or -22dBFS.

 In theory, the 144dB dynamic range of 24bit allows us to quantise down to just a few nano-volts! However, at this level we are talking about the noise level generated by a single resistor. So in practice, many of the LSBs (Least Significant Bits) when recording in 24bit contain just system noise. In other words, the theoretical noise floor of a 24bit digital system is far in excess of what is actually possible in the real world of noisy electronics. That is why 24bit DACs are not able to actually resolve 24bits of dynamic range. If you read an earlier post I mention that some ADCs self dither, this is because the random noise generated by even the finest grade electronics is easily enough to cause the dithering effect.

 I'm not sure this has answered your question?

 For those not used to thinking in decibels scroll down this page for some examples: http://www.jimprice.com/prosound/db.htm

 A very rough way to think of it; if the maximum level of a digital system were set at the sound of a truck going by from 10ft away then 144dB quieter (in 24bit) would be roughly the level of noise produced from two hydrogen atoms colliding!!

 G


----------



## geremy

Yes you answered my question. I was confused about the voltage range of the input. I don't mean to say that the ADC fixed any range of voltage, more along the lines of "the signal we are trying to measure is between X and Y volts". Thanks.


----------



## gregorio

Quote:


  Originally Posted by *fjf* /img/forum/go_quote.gif 
_My bad!....There goes the SACD myth down the drain....We just saved a whole bunch of money_

 

Mmmm, maybe. The technology used on SACD is closely related to digital audio as found on CDs and DVDs but is different. PCM (Pulse Code Modulation) is what is used in 16bit and 24bit digital audio and is what I have discussed in this thread. DSD (Direct Stream Digital) is the technology used on SACD. Basically this technology uses a bit depth of 1 bit but very high sample rates in the megahertz range (2.82mFs/s to be exact). In this sense DSD is very similar to PCM during the initial stages of A to D conversion.

 There are both theoretical advantages and disadvantages of DSD over CD and the professional audio world is largely undecided about which is better. In practice though SACD usually sounds better than CD. This probably isn't due to DSD being better but for other reasons:

 1. DSD technology is relatively expensive so only the higher class studios are capable of creating DSD based recordings.

 2. SACD players are relatively expensive and generally only brought by those consumers really serious about sound quality.

 3. Baring in mind 1 & 2 above, the quality of recording, production and mastering tends to be much higher on SACD releases because the recording industry realises that SACD consumers generally have a higher expectation of the sound quality.

 I don't know how long the SACD format is going to survive but at this point in time SACD probably represents the highest audio quality currently available to the consumer.

 Sorry if I've just cost you a "whole bunch of money"! 
	

	
	
		
		

		
		
	


	




 G


----------



## nealric

A quick look at wikipedia seems to discount the 160db sound is deadly thing. 

 If that were true, anyone who had fired a M1 rifle would be dead or deaf. 
Sound pressure - Wikipedia, the free encyclopedia


----------



## Clutz

Quote:


  Originally Posted by *nealric* /img/forum/go_quote.gif 
_A quick look at wikipedia seems to discount the 160db sound is deadly thing. 

 If that were true, anyone who had fired a M1 rifle would be dead or deaf. 
Sound pressure - Wikipedia, the free encyclopedia_

 

You can also draw your hand through a flame for a fraction of a second without any damage (I know this because I dragged my hand through a bunsen burner this morning, and all I got from it was burned off hair), that doesn't mean that the flame isn't damaging. 160dB from a rifle would last a microsecond. Whether or not an energy source causes damage will be a function of the intensity of the energy source, the length of exposure, and the ability of the energy's destination to dissipate that energy.


----------



## gregorio

Quote:


  Originally Posted by *nealric* /img/forum/go_quote.gif 
_A quick look at wikipedia seems to discount the 160db sound is deadly thing. 

 If that were true, anyone who had fired a M1 rifle would be dead or deaf. 
Sound pressure - Wikipedia, the free encyclopedia_

 


 There are no absolutes with these dB figures when it comes to the level at which someone feels pain, goes deaf or is killed, we are all individuals. For example, try firing an M1 rifle 1m away from your granny and see if she dies!!

 It's usually accepted that at about 120dB - 140dB pain will be felt and permanent hearing damage is likely to occur. Persumably anyone firing an M1 rifle would have to be wearing some kind of hearing protection to avoid serious hearing damage. 180dB is usually the figure quoted for causing death but presumably someone with any kind of heart disorder could be killed by significantly less than this.

 In my original post I mentioned that actually expeiencing 144dB above a 50db noise floor (194dB) would likely kill you.

 G


----------



## apatN

Saving that wall of text until I find myself to be a little brighter. 
 Thanks for your work.


----------



## gregorio

Quote:


  Originally Posted by *ILikeMusic* /img/forum/go_quote.gif 
_Ya think..? 
	

	
	
		
		

		
		
	


	


_

 

Strange, I was expecting more dissenters!?

 G


----------



## oldschool

Here's a link to two music files, both from the same song and mastering session, in standard 16/44 and hi-res 24/96. You tell me if there's difference:

LINK


*When comparing the files, be sure your system is not performing any sort of resampling and/or dithering to either one. Some will by default, either upsample the 16/44 or downsample and dither the 24/96.*


----------



## nick_charles

...


----------



## ILikeMusic

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Strange, I was expecting more dissenters!_

 

You did too good a job with your explanation! But try writing a similar treatise on why copper conductors can't sound 'warmer' than silver and see what happens...


----------



## gregorio

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_I have downloaded these samples in the past, they are not identical, they are not perfectly aligned and there is a recording artifact on the high rez sample. When you analyze them.....

 By either measure there is an average 4db difference between the two samples._

 

Hi Nick, thanks for pointing that out. I have to say that most of the examples I've seen, where people are asked to compare 24bit and 16bit are in some way slightly 'loaded'. They usually have a point to prove or a product to sell.

 I remember when the consumer demand for 24bit started. Most of us thought that consumers were a bit deluded but obviously there were/are those who feel it's an opportunity for a new marketing strategy and that there's money to be made. This whole hi-rez thing is a great opportunity for those who want to make money for nothing, not dissimilar to some of the cable retailers out there! 
	

	
	
		
		

		
		
	


	




 I've been using higher than 16bit recording since 1992 but we just called it 20bit or 24bit. 24bit only started being called 'hi-rez' nearly a decade later and co-incidentally when consumer demand started!!

  Quote:


  Originally Posted by *ILikeMusic* /img/forum/go_quote.gif 
_... try writing a similar treatise on why copper conductors can't sound 'warmer' than silver and see what happens... 
	

	
	
		
		

		
		
	


	


_

 

I'd love to but unfortunately, my expertise is only in the field of audio, rather than psychology!! 
	

	
	
		
		

		
		
	


	




 G


----------



## euphoracle

Wow, that cleared a lot up. Thanks


----------



## nick_charles

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Hi Nick, thanks for pointing that out. I have to say that most of the examples I've seen, where people are asked to compare 24bit and 16bit are in some way slightly 'loaded'. They usually have a point to prove or a product to sell.
_

 

oops, I noticed an error in my comparison which is why I removed my post. However if you load both up in Audacity as 16/44 (downsampling the 2496) and then again as 24/96 (upsampling the 1644) (I used triangular dither) and plot the spectra of both samples in both cases using a 2048 or 4096 FFT, export the figures to a text file load up in Excel and graph them and compare you get some interesting artifacts...


----------



## gregorio

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_oops, I noticed an error in my comparison which is why I removed my post. However if you load both up in Audacity as 16/44 (downsampling the 2496) and then again as 24/96 (upsampling the 1644) (I used triangular dither) and plot the spectra of both samples in both cases using a 2048 or 4096 FFT, export the figures to a text file load up in Excel and graph them and compare you get some interesting artifacts..._

 

There are always going to be slight measurable differences between the 24bit and 16bit versions. However, if the conversion is done correctly these differences will be undetectable when listening. Now, how they got to 16bit from the 24bit master is another issue. Has it been truncated, rounded or dithered, has a good quality noise-shaped dither been used? Has the 24bit version been mastered separately from the 16bit version? Has noise-shaped dither been used on the 24bit version and again on the 16bit version? How has the sample rate conversion been handled? A professional would not use 96kFs/s for a product that will be used on CD. If a higher sample rate is required (due to weak 44.1k filers) then 88.2kFs/s should be used as there is less math involved in getting back to 44.1k and less chance of errors.

 There are all kinds of issues which could (theoretically) make a 16bit version sound (potentially) worse than a 24bit version but none of them due to an inherent weakness of 16bit format compared to 24bit.

 G


----------



## nick_charles

Quote:


  Originally Posted by *ILikeMusic* /img/forum/go_quote.gif 
_You did too good a job with your explanation! But try writing a similar treatise on why copper conductors can't sound 'warmer' than silver and see what happens... 
	

	
	
		
		

		
		
	


	


_

 

I cannot say with any certainty that any copper cable cannot have a more rolled off high frequency response than any silver cable.

 What I can tell you with some certainty is that the Frequency response for the cables (several stranded copper, several solid copper, one Silver plated copper and one stranded silver cable) I have empirically tested in my system using my CD player and my ADC have not shown significant measurable deviations either from a notional flat frequency response or from each other's frequency reponses either in amplitiude or frequency terms.

 This allows me to conclude that in my system none of these cables exhibit notably different frequency "signatures". Make of that what you will.


----------



## JaZZ

Quote:


  Originally Posted by *oldschool* /img/forum/go_quote.gif 
_Here's a link to two music files, both from the same song and mastering session, in standard 16/44 and hi-res 24/96. You tell me if there's difference:

LINK_

 

Thanks for the link! Finally some native 24/96 files to play with...

 I was using my E-MU 1212M for auditioning. The two files vary slightly in maximum amplitude as well as with arbitrarily tested sample points, but the deviation is within 0.04 dB. To exclude any difference other than such induced by the format, I additionaly upsampled and downsampled (...downsampled and upsampled) the two files, using WaveLab Lite. For valid listening tests the E-MU 1212M has to be set to either 44.1 or 96 kHz.

 At 16/44.1 both files sounded identical to me: the (low-rez) original and the file downsampled from 24/96. At 24/96, the hi-rez original sounded more 3-dimensional and had a finer overtone sparkle on the upmost treble. It's certainly not a night-and-day difference and only noticeable after intensive trial, but it's there. It's the same (kind of) difference I notice by comparing DVD-As with CDs.

 With respect to the thread topic -- I don't think the difference has much to do with the higher bit depth, rather with the increased sample rate (actually I forgot testing dynamic resolution separately).

 Headphone used: AKG K 701, connected directly to the (modified) E-MU line out.
.


----------



## Gamerphile

Just asked an hearing aid researcher and expert I know, he asked around a little to be sure. The generall consensus is that there is very little reason to believe spatiel cues we hear should exsist above 20-22KHz since there is not really any natural source which would provide the high energy sound needed to for the ear to physically be able to receive it. The very same reason is why we don't hear that well compared to some animals. While generally sound energy is manly generated within our hearing spectrum many animals communicate at much higher frequency - so they have to do both well.

 How ever as its writte by many experts in the field and a few said here already there is techincal reasons both in recording and playback which can make high sampling rate easyer to sound better.

  Quote:


 Finding out how high a sample rate you need to get all the spatial cue's with you is hardly something anyone have been able to conclude on due to its on unconscious level and spatial perception is not fully understood at all. I'm studying this topic and I haven't at seen any research into this which concentrated on this at all.

 But I don't know everything and there is certainly many who knows more about this - I'm actually going to contact some experts and see if they know anything.


----------



## nick_charles

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_. The two files vary slightly in maximum amplitude as well as with arbitrarily tested sample points, but the deviation is within 0.04 dB._

 

On average yes, however there is an artifact at the beginning of the 2496 files that is a give-away, I was looking at these files a couple of years ago and was able to ABX them to 13/15.

 If you take the segment from 0.07s to 0.17s on both files (unaltered) and analyse them you get some massive variation viz







 Beyond this point the files are pretty much identical but this blip ruins the files for ABXing. If I just gave you this data you would have to conclude that these were not the same recordings at all.


----------



## JaZZ

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_On average yes, however there is an artifact at the beginning of the 2496 files that is a give-away, I was looking at these files a couple of years ago and was able to ABX them to 13/15.

 If you take the segment from 0.07s to 0.17s on both files (unaltered) and analyse them you get some massive variation viz...

 (...)

 Beyond this point the files are pretty much identical but this blip ruins the files for ABXing. If I just gave you this data you would have to conclude that these were not the same recordings at all._

 

That's odd. I haven't listened to the first few seconds, though. 

 Unfortunately I have already inserted some silence at the start (my standard procedure with sound editing), so I can't locate the corresponding position. However, for your purposes it would have been easy to just retouch the issue.
.


----------



## nick_charles

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_However, for your purposes it would have been easy to just retouch the issue.
._

 

Indeed, but the point is that the website says here are two samples that just show the difference between 24/96 and 16/44.1 and nothing else, same mix same recording i.e they should be identical apart from the minor differences between the formats.

 In fact without the blip they are functionally near identical and I would think very hard to tell apart in that test that we can mention here. But the blip is a dead give-away and makes the comparison easy(ish).


----------



## JaZZ

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Indeed, but the point is that the website says here are two samples that just show the difference between 24/96 and 16/44.1 and nothing else, same mix same recording i.e they should be identical apart from the minor differences between the formats._

 

True -- but probably said flaw is not threre on purpose (I can't imagine what for). So for people without a general mistrust agaist audiophile websites, attitudes and approaches the two samples are an interesting playground for format comparisons. Personally I have waited for such an occasion -- since my multiformat player usually doesn't output hi-rez signals.
.


----------



## nick_charles

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_True -- but probably said flaw is not threre on purpose (I can't imagine what for). So for people without a general mistrust agaist audiophile websites, attitudes and approaches the two samples are an interesting playground for format comparisons. Personally I have waited for such an occasion -- since my multiformat player usually doesn't output hi-rez signals.
._

 

I am not suggesting it is deliberate, I see cockup not conspiracy, and also I work on the basis that audiophile recording websites are being sincere. I have no gripe against this site and I agree it is nice to have High res samples for folks to try out.


----------



## scompton

Linn records has test samples for downloading. I have no idea if they are the exact same sample.

Download our testfiles


----------



## JaZZ

Quote:


  Originally Posted by *scompton* /img/forum/go_quote.gif 
_Linn records has test samples for downloading. I have no idea if they are the exact same sample.

Download our testfiles_

 

Thanks for the Linnk!

 I can't discern the original 24/192 file from a 16/44.1 downsampled version. Apparently this piano duet is not critical enough a recording for my ears.
.


----------



## gregorio

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Beyond this point the files are pretty much identical but this blip ruins the files for ABXing._

 

Strange, if the blip is on the 24/96 but not the 16/44.1 this would imply that the 16bit version was not made directly from the 24/96 master, otherwise it too would have the blip. Of course, for the perposes of this thread a more accurate comparison would be 24/44.1 and 16/44.1 or 24/96 and 16/96.

 If we have moved on to to mentioning 192kFs/s, there are some problems with this sample rate that if any difference can be detected it is likely to be a deterioration in SQ, rather than an improvement. Have a read of this:
http://www.lavryengineering.com/docu...ing_Theory.pdf

 Unfortunately, my system is setup for a project I'm working on at the moment and I can't actually test the files which have been posted.

 G


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_If we have moved on to to mentioning 192kFs/s, there are some problems with this sample rate that if any difference can be detected it is likely to be a deterioration in SQ, rather than an improvement. Have a read of this:
http://www.lavryengineering.com/docu...ing_Theory.pdf_

 

Note that this paper is 5 years old. Moreover it's just _one_ opinion -- stated as a fact.
.


----------



## gregorio

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Note that this paper is 5 years old. Moreover it's just one opinion -- stated as a fact.
._

 

Actually there are a lot of people who agree with him. Also, this opinion is worth more than most as he's regarded professionally as a leading expert in the field. Some of the principles I laid out in my original post are 30 years old, doesn't make them any less true now though.

 G


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Actually there are a lot of people who agree with him. Also, this opinion is worth more than most as he's regarded professionally as a leading expert in the field. are 30 years old, doesn't make them any less true now though._

 

My objection isn't addressing «some of the principles you laid out in your original post», but rather the generalization deduced from Lavry's pretension that a sampling rate of 192 kHz will forever be too high a challenge for any future ADCs or DACs -- this from a perspective from 2004. Even if he still may have some following in this matter (such as you, among others), that doesn't make the hypothesis any more reliable or plausible.
.


----------



## gregorio

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_My objection isn't addressing «some of the principles you laid out in your original post», but rather the generalization deduced from Lavry's pretension that a sampling rate of 192 kHz will forever be too high a challenge for any future ADCs or DACs -- this from a perspective from 2004. Even if he still may have some following in this matter (such as you, among others), that doesn't make the hypothesis any more reliable or plausible.
._

 

It's not just me, many, many others have respect for one of the leading experts on the planet. I realise this counts for nothing here on head-fi though, where everyone appears to know more than those who do it for a living or indeed know more than the leading experts.

 Of course, it's not just the ability to process the datastream but also the fact that there is nothing in those frequencies to capture. There has been arguments that audiophiles here on head-fi can apparently hear beyond 20kHz and are therefore different from normal human beings. Are we now going to have a discussion that head-fiers can now hear beyond 48kHz and need the frequency of digital audio to go up to 96kHz? If so you are wasting your time, no microphone in any recording studio goes anywhere near 96kHz, in fact very few of them go much beyond 20kHz, what about your speakers or cans, do they have a freq response of 96kHz? Anyone thinking there is anything that can either be captured or heard up there is completely fooling themselves.

 Just to make it clear, there could (in theory) be some benefit to 96kFs/s under certain conditions. 192kFs/s is a complete waste of time, it's even a waste of time for recording, let alone for listening.

 G


----------



## ILikeMusic

Quote:


 Are we now going to have a discussion that head-fiers can now hear beyond 48kHz and need the frequency of digital audio to go up to 96kHz? 
 

Only if it can be done with vacuum tubes.


----------



## mark_h

Quote:


  Originally Posted by *manaox2* /img/forum/go_quote.gif 
_Your never safe from the recording engineer, especially in today's music.
	

	
	
		
		

		
		
	


	


_

 






 I have had a really hard weekend at work, this thread has put me to sleep. Just what I needed, thanks G!

 Writes in sleep...trust your ears, ignore sound science...zzzzzzz


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_It's not just me, many, many others have respect for one of the leading experts on the planet. I realise this counts for nothing here on head-fi though, where everyone appears to know more than those who do it for a living or indeed know more than the leading experts._[/i]

 

That's not what I intended. Please read my post again. I do have respect for knowledgeable people. You're simply -- and arbitrarily -- misinterpreting my statement.


  Quote:


 _Just to make it clear, there could (in theory) be some benefit to 96kFs/s under certain conditions. 192kFs/s is a complete waste of time, it's even a waste of time for recording, let alone for listening._ 
 

Maybe -- but you can't be sure about that. Not more sure than other -- knowledgeable -- people. Or people with positive experience with 192 kHz.
.


----------



## chesebert

.


----------



## manaox2

Quote:


  Originally Posted by *mark_h* /img/forum/go_quote.gif 
_






 I have had a really hard weekend at work, this thread has put me to sleep. Just what I needed, thanks G!

 Writes in sleep...trust your ears, ignore sound science...zzzzzzz_

 

I made the mistake of trusting them with my wallet. 
	

	
	
		
		

		
		
	


	




 I think this thread is great, wish there were more like it.


----------



## Acix

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_It's not just me, many, many others have respect for one of the leading experts on the planet. I realise this counts for nothing here on head-fi though, where everyone appears to know more than those who do it for a living or indeed know more than the leading experts.

 Of course, it's not just the ability to process the datastream but also the fact that there is nothing in those frequencies to capture. There has been arguments that audiophiles here on head-fi can apparently hear beyond 20kHz and are therefore different from normal human beings. Are we now going to have a discussion that head-fiers can now hear beyond 48kHz and need the frequency of digital audio to go up to 96kHz? If so you are wasting your time, no microphone in any recording studio goes anywhere near 96kHz, in fact very few of them go much beyond 20kHz, what about your speakers or cans, do they have a freq response of 96kHz? Anyone thinking there is anything that can either be captured or heard up there is completely fooling themselves.

 Just to make it clear, there could (in theory) be some benefit to 96kFs/s under certain conditions. 192kFs/s is a complete waste of time, it's even a waste of time for recording, let alone for listening.

 G_

 

Maybe you're the one that's fooling yourself, here. There are professional studio monitors like the Adam and the Yamaha that go up to 35-40kHz. The thing is, you don't have to be able to hear up to 40kHz in order to perceive the information there. And of course, this knowledge is proven through lab equipment that measures the frequencies. 

 Now practically, in the studio, these headphones and speakers that have an extended frequency response will have more space to the sound and therefore the music will be more, precise, smooth, airy and dynamic with a better details in the image. This is sound resolution and it's the same with digital. You don't need to be able to hear up tp 96 or 192 kHz, but there is information that is stored there. Of course you need to work with a chain of studio equipment that supports high resolution. Eventually when you're going to dither down to 16/44, or just a lousy mp, your mix will be much more rich and this is the benefit of working in high resolution.

 Here is more info about dithering http://www.apogeedigital.com/pdf/UV22osquick.pdf


----------



## gregorio

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Maybe you're the one that's fooling yourself, here. There are professional studio monitors like the Adam and the Yamaha that go up to 35-40kHz. The thing is, you don't have to be able to hear up to 40kHz in order to perceive the information there. And of course, this knowledge is proven through lab equipment that measures the frequencies. 

 Now practically, in the studio, these headphones and speakers that have an extended frequency response will have more space to the sound and therefore the music will be more, precise, smooth, airy and dynamic with a better details in the image. This is sound resolution and it's the same with digital. You don't need to be able to hear up tp 96 or 192 kHz, but there is information that is stored there. Of course you need to work with a chain of studio equipment that supports high resolution. Eventually when you're going to dither down to 16/44, or just a lousy mp, your mix will be much more rich and this is the benefit of working in high resolution._

 

 Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Maybe -- but you can't be sure about that. Not more sure than other -- knowledgeable -- people. Or people with positive experience with 192 kHz.._

 

If there is nothing there to record, if none of the equipment is capable of recording freqs in that range and if none of the playback equipment is capable of reproducing it then, yes I can be sure.

 Look, it's quite simple. No microphone can pick up anything much beyond 20kHz, yes some of their specs go to 35 or 40kHz but just look at the response roll-off. If there was anything up there, mics cannot record it. Secondly, yes, there are speakers that will in theory go up to 40kHz but what are they going to reproduce, nothing can be recorded there and you couldn't hear it if it were. Of course 40kHz can (in theory) be recorded using a sample rate of 96kFs/s. So now you want to double this sampling rate to 192kFs/s so your audio limit is now 96kHz. 96kHz is more than double what can either be recorded or that your system can reproduce.

 This is not just an opinion, it's very simple fact. No instrument produces any notes beyond about 8kHz, the only thing present anywhere near 20kHz is harmonics. Each subsequent higher harmonic is lower in amplitude than the previous harmonic. This is simple basic acoustics. By the time we get to around 20kHz the harmonics are so quiet that they are starting to disappear below the noise floor. By the time we get to 30kHz the harmonics are already way below the noise floor. Even the finest mics have very little response at 30kHz (let alone able to record something below the noise floor), so there is simply no physical way to record these harmonics. If there is something stored in these ultra-sonic frequencies it can *only* be system noise generated by electronics in the signal chain.

 If people think 192kFs/s sounds better, I'm sorry but there is no sensible alternative to the fact they are fooling themselves. The only possible alternative is that their DAC has some kind of malfunction which so negatively affects the re-construction of signals at 44.1kFs/s and 96kFs/s that 192kFs/s sounds better. It really is just another case of consumers' expectation that more data = better quality. 

 Just to make absolutely clear, there is *nothing* produced by any instrument that exists above the noise floor once we get to about 30kHz and there is certainly nothing at 96kHz. Even if there were, it is not possible to record it because it is way beyond the capability of studio mics. Even if something does exist and we could record it, your system could not reproduce it and even if it could, it's well beyond the hearing capabilities of a dog, let alone a human being! Remember also, there is no reliable proof that anyone can hear beyond 22kHz and we are talking here (with 192kFs/s) about extending the range of encodable frequencies from 48kHz to 96kHz!

 Lastly, in answer to the last sentence I quoted from Acix: If you decimate the sample rate from 192kFs/s to 44.1kFs/s a brickwall filter has to be applied to completely remove all audio frequencies above 22,050Hz. If all the frequencies above this point are not removed the re-sampling fails! So whatever may or may not be above 22kHz is totally and permanently removed. If there is nothing there, it cannot therefore make the CD sound "rich".

 G

 PS. For those who are not aware, the sample rate (correctly notated as kFs/s) is always double the highest audio frequency (notated as Hz or kHz). This is a basic tenet of the Nyquist-Shannon Sampling Theory.


----------



## linuxworks

what about 'scissors' so to speak?

 ie, the argument about speed of light - that nothing can go faster - yet if you combine 2 things that move at the speed of light and they move toward each other, you have a problem with a+b still not being greater than C 
	

	
	
		
		

		
		
	


	




 so, suppose you have 2 high freq waves that are at slightly diff frequencies. combine them on a scope and look at the places where they overlap - you need higher frequency components needed to be able to reproduce that, don't you?

 perhaps restated: which do you think is higher fidelity: recording 2 musicians with 1 mic (combined) or recording 2 musicians each with their own mic and each having their own speaker and amp for playback? I would think that each would be able to capture the single musician easier than 2x the amount of 'wave activity' in the air.

 so, it was never the single 20k tone that gives the need for higher capture rates but the fact that when you add more and more 'stuff' its harder and harder for that 20k to reproduce the entire set of sounds at record time.

 is this correct or is there a flaw in this logic?


----------



## nick_charles

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_This is not just an opinion, it's very simple fact. No instrument produces any notes beyond about 8kHz, the only thing present anywhere near 20kHz is harmonics. Each subsequent higher harmonic is lower in amplitude than the previous harmonic. _

 

That is not strictly true. The pattern is not linear , it tends downwards but some harmonics are higher than their predecessors.


  Quote:


 This is simple basic acoustics. By the time we get to around 20kHz the harmonics are so quiet that they are starting to disappear below the noise floor. 
 

That depends on which noise floor you are talking about. With CD the noise floor at -96db is low enough for harmonics at 20khz to be comfortably above the noise floor and in the cymbals crash test I did when the fundamental was at -5db the high harmonics were at ~ -63db at 20khz. Now I cannot hear that and you would need a quiet hall to get them but it is far from implausible and in a quiet listening room no problem. What renders them moot is masking, if you have a signal at -5db and one at -60db the one at -60db is not going to get noticed very much.

  Quote:


 Just to make absolutely clear, there is *nothing* produced by any instrument that exists above the noise floor once we get to about 30kHz 
 

Um, the Balinese Gamelan has harmonics above the noise floor extending way up beyond human audibility.


----------



## nick_charles

Quote:


  Originally Posted by *linuxworks* /img/forum/go_quote.gif 
_so, suppose you have 2 high freq waves that are at slightly diff frequencies. combine them on a scope and look at the places where they overlap - you need higher frequency components needed to be able to reproduce that, don't you?_

 

No, it does not work like that, the composite frequency is never higher than the higher of the two components , the waveform just becomes more complex but Shannon-Nyquest operates for any arbitrarily complex signal not just sine waves.


----------



## linuxworks

I'm trying to understand this - trying to visualize it.

 the wave gets 'more complex' but isn't that adding more harmonics than were there with just the 2 initial sound sources?

 and again, the thought experiment about recording 2 audio sources separately and playing them back at the same time vs combining them and playing back the composite. isn't the 2 going to be more accurate than one?

 another analogy: s-video vs composite vs component. if summing was non-destructive, as you say, then why do you get better fidelity with the separation of components, carrying them separately and them combining them at the final destination? composite video is 'a mess' and s-video breaks out Y/C into 2 wires; and component breaks it down even further.

 it just seems that the 'divide and conquer' gets you better accuracy in tracking complex waveforms than simply combining them and hoping you have enough 'total bw' to carry them without loss.


----------



## gregorio

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_That is not strictly true. The pattern is not linear , it tends downwards but some harmonics are higher than their predecessors.

 That depends on which noise floor you are talking about. With CD the noise floor at -96db is low enough for harmonics at 20khz to be comfortably above the noise floor and in the cymbals crash test I did when the fundamental was at -5db the high harmonics were at ~ -63db at 20khz. Now I cannot hear that and you would need a quiet hall to get them but it is far from implausible and in a quiet listening room no problem. What renders them moot is masking, if you have a signal at -5db and one at -60db the one at -60db is not going to get noticed very much.

 Um, the Balinese Gamelan has harmonics above the noise floor extending way up beyond human audibility._

 

I was not referring to the the theoretical noise floor of CD but to the actual noise floor when recording or replaying the signal and the actual practicalities of the equipment. IE. Good quality studio condensers (U87, etc), have a frequency cutoff around 20kHz. There are one or two which go up to 40kHz but again their roll off is usually quite severe above 20kHz. It's only usually laboratory mics that capture anything much above 30kHz and these are not suitable for studio use. And I haven't even mentioned mic-pres, EQ units, etc. Most producers and engineers would not want to capture anything above 20kHz anyway. Because these frequencies cannot be heard, engineers can't mix them with any degree of accuracy. So although there are the odd instruments (like Gamelan and some cymbals) which do produce harmonics above the theoretical noise floor of digital audio, they are not above the noise floor of the practical equipment or environment (including Brownian motion) in a recording and playback chain.

 Also, I was referring to sample rates of 192kFs/s, which gives us a theoretical audio frequency maximum of 96kHz. So the argument is, what is between 48kHz and 96kHz in a music performance that in practice can either be recorded or replayed? The answer is nothing! When we start to look at what humans are capable of hearing the figures start to get even more silly. 44.1kFs/s can encode pretty much to the limit of human hearing. 96kFs/s more than doubles the human limits and even exceeds the hearing limits of a dog. 192kFs/s goes more than double the hearing limitations of a dog and starts to approach the limits of what a bat can hear.

 G


----------



## nick_charles

Quote:


 This is not just an opinion, it's very simple fact. No instrument produces any notes beyond about 8kHz, the only thing present anywhere near 20kHz is harmonics. Each subsequent higher harmonic is lower in amplitude than the previous harmonic. 
  Quote:


 That is not strictly true. The pattern is not linear , it tends downwards but some harmonics are higher than their predecessors. 
 


 


  Quote:


 This is simple basic acoustics. By the time we get to around 20kHz the harmonics are so quiet that they are starting to disappear below the noise floor. 

  Quote:


 That depends on which noise floor you are talking about. With CD the noise floor at -96db is low enough for harmonics at 20khz to be comfortably above the noise floor and in the cymbals crash test I did when the fundamental was at -5db the high harmonics were at ~ -63db at 20khz. Now I cannot hear that and you would need a quiet hall to get them but it is far from implausible and in a quiet listening room no problem. What renders them moot is masking, if you have a signal at -5db and one at -60db the one at -60db is not going to get noticed very much. 
 


 


  Quote:


 Just to make absolutely clear, there is nothing produced by any instrument that exists above the noise floor once we get to about 30kHz 

  Quote:


 Um, the Balinese Gamelan has harmonics above the noise floor extending way up beyond human audibility. 
 


 



  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I was not referring to the the theoretical noise floor of CD but to the actual noise floor when recording or replaying the signal and the actual practicalities of the equipment. _

 

No, these are separate issues and I do not disagree with you on the pragmatics, but to say something is below the noise floor you now have to say what the noise floor is. If it is -50db then Gamelan harmonics are in play

  Quote:


 So although there are the odd instruments (like Gamelan and some cymbals) which do produce harmonics above the theoretical noise floor of digital audio, they are not above the noise floor of the practical equipment or environment (including Brownian motion) in a recording and playback chain. 
 

What is the ball park noise floor for a decent acoustic recording set-up ?. The ~ 20K cymbals harmonics I recorded from a 16/44.1 sample played back on an average CD player and recorded on a low end ADC were 30db above CD noise floor which is a lot , they may be above recording room noise floor ?.

 In the Oohashi study the Gamelan had harmonics at 50khz that were at -50db wrt the dominant tone and harmonics at 20K that were much much higher. The Oohashi study is highly flawed but I do not doubt that part.


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_If there is nothing there to record, if none of the equipment is capable of recording freqs in that range and if none of the playback equipment is capable of reproducing it then, yes I can be sure.

 Look, it's quite simple. No microphone can pick up anything much beyond 20kHz, yes some of their specs go to 35 or 40kHz but just look at the response roll-off. If there was anything up there, mics cannot record it._

 

You're so easy to refute -- I like it! 
	

	
	
		
		

		
		
	


	









 Here's a 96-kHz recording (below) and the same recording downsampled to 44.1 kHz, then upsampled to 96 kHz again (above) -- for comparability. Below you can see signals with a frequency content in the high 40 kHz, whereas the downsampled version is smoothed by the 22-kHz limitation. There are a lot of such signals in this recording («Dragon Boats», linked by _Oldschool_).


  Quote:


 _Secondly, yes, there are speakers that will in theory go up to 40kHz but what are they going to reproduce, nothing can be recorded there and you couldn't hear it if it were. Of course 40kHz can (in theory) be recorded using a sample rate of 96kFs/s. So now you want to double this sampling rate to 192kFs/s so your audio limit is now 96kHz. 96kHz is more than double what can either be recorded or that your system can reproduce._ 
 

Maybe it's not so important to hear it (–> ultrasonics), but rather to have a better preserved signal shape without distortion by the filter in proximity to the audible range (–> ringing). After all I do hear a difference in favor of hi-rez (also in the case at hand). 


  Quote:


 _This is not just an opinion, it's very simple fact. If people think 192kFs/s sounds better, I'm sorry but there is no sensible alternative to the fact they are fooling themselves. The only possible alternative is that their DAC has some kind of malfunction which so negatively affects the re-construction of signals at 44.1kFs/s and 96kFs/s that 192kFs/s sounds better. It really is just another case of consumers' expectation that more data = better quality._ 
 

So you're really sure? I'm (almost) about to believe you, since your reference is quite impressive _(trained as an orchestral musician at conservertoire and then played professionally with symphony orchestras for some years... own recording studio and probably been analytically listening to orchestral recordings since the late '70s...)._ But what kills your image is your attitude to display your knowledge and experience in an obtrusive manner. Apart from your absolutisms.


  Quote:


 _Just to make absolutely clear, there is *nothing* produced by any instrument that exists above the noise floor once we get to about 30kHz and there is certainly nothing at 96kHz. Even if there were, it is not possible to record it because it is way beyond the capability of studio mics. Even if something does exist and we could record it, your system could not reproduce it and even if it could, it's well beyond the hearing capabilities of a dog, let alone a human being! Remember also, there is no reliable proof that anyone can hear beyond 22kHz..._ 
 

And so on.
.


----------



## gregorio

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_No, these are separate issues and I do not disagree with you on the pragmatics, but to say something is below the noise floor you now have to say what the noise floor is. If it is -50db then Gamelan harmonics are in play

 What is the ball park noise floor for a decent acoustic recording set-up ?. The ~ 20K cymbals harmonics I recorded from a 16/44.1 sample played back on an average CD player and recorded on a low end ADC were 30db above CD noise floor which is a lot , they may be above recording room noise floor ?.

 In the Oohashi study the Gamelan had harmonics at 50khz that were at -50db wrt the dominant tone and harmonics at 20K that were much much higher. The Oohashi study is highly flawed but I do not doubt that part._

 

It's impossible to say what the noise floor of an average studio is. The recording environments of the top studios can be as low as 30dB but for much of the equipment in the chain figures beyond 20kHz are quite difficult to obtain. The big commercial studios may have a low noise floor but their live rooms are probably big enough to start introducing attenuation of higher frequencies through air absorption. +60dB is about the maximum that mic-pres go. The closer to this maximum one gets though, the more distortion is introduced. The real problem though is the mics, for the vast majority of mics -50dB @ 30kHz is just not possible. All dynamic mics are out of the question and all the commonly used studio condensers. Try EQ boosting above 20kHz and all that is present in the signal is noise. Also, cymbals and to an extent Gamelan in the higher frequencies are so full of both odd and even harmonics, the result is indistinguishable from white noise anyway and this is even in the audible frequency band. In other words, in practice we are (perceptually) adding to the noise floor rather than being able to distinguish sounds separate from the noise floor.

 I accept that for a limited few instruments there may be harmonic information present at 30kHz which would register above the theoretical noise floor of digital audio and even above the noise floor of some recording environments. But when we sum these two together and factor in mic response and the effects of everything else in the recording and playback chains, I still don't believe there would be much at 30kHz which could be captured. Regardless of this though, there is still no justification for people believing there is something to be gained by encoding audio frequencies of 96kHz.

 G


----------



## Acix

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_If there is nothing there to record, if none of the equipment is capable of recording freqs in that range and if none of the playback equipment is capable of reproducing it then, yes I can be sure.

 Look, it's quite simple. No microphone can pick up anything much beyond 20kHz, yes some of their specs go to 35 or 40kHz but just look at the response roll-off. If there was anything up there, mics cannot record it. Secondly, yes, there are speakers that will in theory go up to 40kHz but what are they going to reproduce, nothing can be recorded there and you couldn't hear it if it were. Of course 40kHz can (in theory) be recorded using a sample rate of 96kFs/s. So now you want to double this sampling rate to 192kFs/s so your audio limit is now 96kHz. 96kHz is more than double what can either be recorded or that your system can reproduce.

 This is not just an opinion, it's very simple fact. No instrument produces any notes beyond about 8kHz, the only thing present anywhere near 20kHz is harmonics. Each subsequent higher harmonic is lower in amplitude than the previous harmonic. This is simple basic acoustics. By the time we get to around 20kHz the harmonics are so quiet that they are starting to disappear below the noise floor. By the time we get to 30kHz the harmonics are already way below the noise floor. Even the finest mics have very little response at 30kHz (let alone able to record something below the noise floor), so there is simply no physical way to record these harmonics. If there is something stored in these ultra-sonic frequencies it can *only* be system noise generated by electronics in the signal chain.

 If people think 192kFs/s sounds better, I'm sorry but there is no sensible alternative to the fact they are fooling themselves. The only possible alternative is that their DAC has some kind of malfunction which so negatively affects the re-construction of signals at 44.1kFs/s and 96kFs/s that 192kFs/s sounds better. It really is just another case of consumers' expectation that more data = better quality. 

 Just to make absolutely clear, there is *nothing* produced by any instrument that exists above the noise floor once we get to about 30kHz and there is certainly nothing at 96kHz. Even if there were, it is not possible to record it because it is way beyond the capability of studio mics. Even if something does exist and we could record it, your system could not reproduce it and even if it could, it's well beyond the hearing capabilities of a dog, let alone a human being! Remember also, there is no reliable proof that anyone can hear beyond 22kHz and we are talking here (with 192kFs/s) about extending the range of encodable frequencies from 48kHz to 96kHz!

 Lastly, in answer to the last sentence I quoted from Acix: If you decimate the sample rate from 192kFs/s to 44.1kFs/s a brickwall filter has to be applied to completely remove all audio frequencies above 22,050Hz. If all the frequencies above this point are not removed the re-sampling fails! So whatever may or may not be above 22kHz is totally and permanently removed. If there is nothing there, it cannot therefore make the CD sound "rich".

 G._

 

If you want some reliable proof that anyone can hear beyond 22kHz. You can check out the SPL Phonitor...I was able to listen to the Phonitor and compare my previous Ultrasone PL650 to the K702. Check it out if you get a chance, you just might be able to hear something above the normal human range. If not, you can still hang on to your theory.

 Now, I get the impression that you don't really have experience working in HR sound environment...just theoretical concepts. Even for a theoretical concept dude whose beliefs are stronger than his experiences, you still believe that the leading expert at ADAM and Yamaha and SPL and all the other manufacturers have just created this range out of thin air to use as a marketing tool.


----------



## gregorio

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_If you want some reliable proof that anyone can hear beyond 22kHz. You can check out the SPL Phonitor...I was able to listen to the Phonitor and compare my previous Ultrasone PL650 to the K702. Check it out if you get a chance, you just might be able to hear something above the normal human range. If not, you can still hang on to your theory.

 Now, I get the impression that you don't really have experience working in HR sound environment...just theoretical concepts. Even for a theoretical concept dude whose beliefs are stronger than his experiences, you still believe that the leading expert at ADAM and Yamaha and SPL and all the other manufacturers have just created this range out of thin air to use as a marketing tool._

 

1. As far as I'm concerned, I am willing to admit that it may be possible to perceive some sonic content above 20kHz, although there is only anecdotal evidence for it. However, if there is anything to be heard it is definitely lower than 40kHz, simply because there are no studio mics that can pick up anything above this point. A 40kHz signal can be encoded using a sample rate of 96kFs/s. There is however absolutely no advantage to a sample rate of 192kHz as there simply is nothing between 48kHz and 96kHz which can be recorded. If someone thinks they hear a difference with 192kFs/s, it cannot be anything to do with the recording, because no studio mics can record above 40kHz, so the only thing which can be in these higher frequencies on a recording is noise.

 2. Sorry but you are way off the mark with your second paragraph. I have been recording and mixing exclusively in higher than 16bit since the end of 1992, not long after the technology was first available (Yamaha DMR8 + DRU8). I switched over from 20bit to 24bit in about 1995, when multi-channel 24bit converters first became available (DigiDesign 888). So, there can't be that many engineers who have a longer practical experience than me in working with >16bit. Also, I've used higher resolution recording technology not just for music recording and production but also quite extensively for film and TV sound too. My understanding of the theory side of HR has come from a fair bit of research over the years and particular thanks need to go to Nika Aldrich who gave many hours of his time online to help iron out many of my misunderstandings.

 In fact, there is not much in the digital audio chain that I haven't thoroughly tested. Take dither for example, to start with only TDPF (Triangular Probability Density Functions) were available but I've used extensively Sony Super-Bitmapping, UV22, POWr, Waves L2 and DigiDesign. I've gone through or tested countless mics, mic-pres, cables, ADCs, DACs and speakers. Since the early '90s I must have spent around $500,000 on equipment and acoustics. I've also done work in many of this country's (UK) top studios and dubbing theatres as well as my own of course.

 G


----------



## Aleatoris

Nevermind, I'm a retard.


----------



## gregorio

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_... you still believe that the leading expert at ADAM and Yamaha and SPL and all the other manufacturers have just created this range out of thin air to use as a marketing tool._

 

No, as I understand it, the 24bit 192kFs/s standard was decided upon because it was way beyond what was ever likely to be needed. If you create a standard which is so ridiculously beyond what humans are capable of perceiving that will hopefully result in this format never going out of date or requiring a new format incompatible with old formats. Bare in mind this format was agreed before any converters existed which were capable of 192kFs/s.

 ADAM, Yamaha and SPL did not invent the format, as far as I know the first to support this format were DigiDesign. I can't remember off the top of my head the name or composition of the standards organization responsible. I should also mention, that professionally we've been using higher than 16bit for almost 20 years, although it's only relatively recently that it's been called Hi-Res and there has been a consumer demand for it. Now there is a market, I cannot blame Yamaha or ADAM for taking advantage of it, even if the demand is almost entirely superfluous for the consumer. If you are talking about professional studio monitors, then as mentioned previously 24bit is useful for recording and mixing, it's as a playback consumer format that 24bit is irrelevant.

 G


----------



## JaZZ

*Sennheiser MKH 8000 series:*






.


----------



## gregorio

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Sennheiser MKH 8000 series_

 

Yes, those Sennheiser mics look interesting. I haven't had chance to see them yet. They were only released last year and have only become available in the last few months. In fact, AFAIK some of the 8000 series models have still not been released yet. I presume they are designed to replace the MKH40 series, which I have used and which have a maximum frequency of 20kHz, good mics though.

 To be honest I'm looking forward to having a play with the 8040 and seeing exactly what their frequency response is. At the moment though I don't have access to one and I don't know anyone who has done more than just seen them at a trade show. Again though, all but the last few kHz of their (quoted) maximum frequency response (50kHz) can be encoded at 96kFs/s. They go nowhere near the 96kHz of 192kFs/s.

 As they are so new though, they cannot account for differences that consumers think they hear with existing 192kFs/s recordings.

 As I said before, there are a few mics which go up to 40kHz (Schoeps for example) and it looks like there is a new one which might go up to 50kHz. All the most commonly used studio mics though, max out at 20kHz or lower.

 You can try nit-picking and finding any scrap of information which may provide some proof that some relatively insignificant detail of what I have stated is disputable. But the main substance of what I have stated is correct, namely: 24bit is a complete waste of time for the consumer and so is 192kFs/s.

 G


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_...the main substance of what I have stated is correct, namely: 24bit is a complete waste of time for the consumer and so is 192kFs/s._

 

I'm not pretending the opposite -- since I have no experience with 24/196. But in contrast to you I just leave it open that the higher data resolution (from 96 to 192 kHz) might possibly provide a noticeable sonic advantage -- as 96 kHz does compared to 44.1 kHz, despite the fact that it shouldn't.
.


----------



## gregorio

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_I'm not pretending the opposite -- since I have no experience with 24/196. But in contrast to you I just leave it open that the higher data resolution (from 96 to 192 kHz) might possibly provide a noticeable sonic advantage -- as 96 kHz does compared to 44.1 kHz, despite the fact that it shouldn't.
._

 

Theoretically there is good a reason why 96kFs/s could sound better than 44.1kFs/s. The anti-alias filters at 44.1k have to be very steep (to stay out of the hearing range) and it's relatively difficult and expensive to create good steep brick wall filters without noticeable artifacts. The anti-alias filters are much smoother (less steep) with 96kFs/s and so are much easier and cheaper to implement without noticeable artifacts. So in theory it is potentially possible that a difference could be perceived. There is some anecdotal evidence to support this potential difference but so far as I'm aware no one has yet managed to get a significant result in DBT. Most good studios use good quality ADCs though, so in most commercial releases it is extremely unlikely (but not provably impossible!) that the anti-alias filter at 44.1kFs/s is going to have any noticeable artifacts which could be improved by the smoother filters in 96kFs/s.

 Again though, even if there is a perceivable difference with 96kFs/s it is not related to data resolution. I conducted my own tests in 2002 with both 96kHz and 192kFs/s and neither I nor any of my colleagues could tell a difference, except that my hard disk space seemed to get eaten up at an alarming rate and my processing ability was decimated! 
	

	
	
		
		

		
		
	


	




 These days, when recording for TV/Film I use 24bit-48kFs/s and then dither down to 16bit. For CD I generally use 24bit - 44.1kFs/s or if I'm not convinced with the quality of the ADC, I might use 24bit-88.2kFs/s then dither back to 16bit-44.1k. Beyond for testing purposes, I don't bother with 192kFs/s at all.

 G


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Theoretically there is good a reason why 96kFs/s could sound better than 44.1kFs/s. The anti-alias filters at 44.1k have to be very steep (to stay out of the hearing range) and it's relatively difficult and expensive to create good steep brick wall filters without noticeable artifacts. The anti-alias filters are much smoother (less steep) with 96kFs/s and so are much easier and cheaper to implement without noticeable artifacts. So in theory it is potentially possible that a difference could be perceived. There is some anecdotal evidence to support this potential difference but so far as I'm aware no one has yet managed to get a significant result in DBT. Most good studios use good quality ADCs though, so in most commercial releases it is extremely unlikely that the anti-alias filter at 44.1kFs/s is going to have any noticeable artifacts which could be improved by the smoother filters in 96kFs/s.

 Again though, even if there is a perceivable difference with 96kFs/s it is not related to data resolution._

 

We already had this. I think it is, nonetheless. Because in this case I'm referring to the «Dragon Boats» -- and the comparison between the original 96-kHz and the downsampled 44.1-kHz version. No hardware filtering involved.
.


----------



## linuxworks

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_then as mentioned previously 24bit is useful for recording and mixing, it's as a playback consumer format that 24bit is irrelevant.

 G_

 

exact same analog in photography. you want to capture at high bit depth, but even more important is STAYING at that depth during the processing. in photo, we start with a raw image (non compressed) and we keep data at 16bit/pixel and only at 'save as' time do we 'dither down' to 8bit jpg for regular old joe's use on the web or even for print.

 edit needs high res because of math roundoff. but final user presentation does NOT apply to the same standard and 8bit is just fine for photo use.

 I do believe that 16bit audio is 'just fine' for end consumer use. a GOOD clean 16 bits, but that's all it really takes.

 also, do remember, that in the real world, storage is not infinite. photos, songs, movies - they eat up disk. that's NOT a good thing (I'm thinking blue ray, which is hella expensive in storage for what it SHOULD be taking, but I digress..). so even if there's some slight test-equip measurable diff by going deeper in bit depth or sampling freq, you have to remember that portable players and even home disks have to store this stuff.

 a good technical design makes a tradeoff between absolute perfection and the real world. 24bit and 96k do not seem 'real world' to me as final formats for end user listening. still too expensive and they don't justify their piggish storage needs.


----------



## JaZZ

Quote:


  Originally Posted by *linuxworks* /img/forum/go_quote.gif 
_exact same analog in photography. you want to capture at high bit depth, but even more important is STAYING at that depth during the processing. in photo, we start with a raw image (non compressed) and we keep data at 16bit/pixel and only at 'save as' time do we 'dither down' to 8bit jpg for regular old joe's use on the web or even for print._

 

Yes. -- Pixel number and sharpness on the other hand would represent the sampling rate. So if you want the pictures to be enjoyed on a large poster format, better leave it in hi-rez. Some audiophiles listen to their music in poster format (think electrostatic headphones, HD 800, high-end speakers...).
.


----------



## gregorio

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_We already had this. I think it is, nonetheless. Because in this case I'm referring to the «Dragon Boats» -- and the comparison between the original 96-kHz and the downsampled 44.1-kHz version. No hardware filtering involved.
._

 

Filtering has to be involved, there is no choice or option. The Nyquist limit for 44.1kFs/s is 22,050Hz anything above this frequency must be removed otherwise the sampling (or re-sampling) process either fails entirely or results in alias images which cause serious artifacts. It's not hardware filtering, just as it isn't hardware filtering in ADCs but it must brick wall filter, this is a basic "law" of digital audio.

  Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Some audiophiles listen to their music in poster format (think electrostatic headphones, HD 800, high-end speakers...).
._

 

No they don't, there is no equivalent in audio to "large poster format". With bit depth it would mean the audiophile listening at a volume which would at the least make them permanently deaf and in sample rate it would mean pitch shifting the highest frequencies in the music down by more than 2 octaves, while leaving the lower frequencies unaltered (otherwise they would be too low to be heard!).

 G


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Filtering has to be involved, there is no choice or option. The Nyquist limit for 44.1kFs/s is 22,050Hz anything above this frequency must be removed otherwise the sampling (or re-sampling) process either fails entirely or results in alias images which cause serious artifacts. It's not hardware filtering, just as it isn't hardware filtering in ADCs but it must brick wall filter, this is a basic "law" of digital audio._

 

Now you lost me. Are you saying that WaveLab doesn't do the low-pass filtering right during downconversion? Which kind of filter(ing) would do it right, then, in your opinion? BTW, I also compared the original 44.1-kHz version of the «Dragon Boats» with the hi-rez version, to the same avail.

 It seems that you can't avoid brickwall filtering (no news) -- hence you can't avoid the flaws you attribute to it.
.


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_No they don't, there is no equivalent in audio to "large poster format". With bit depth it would mean the audiophile listening at a volume which would at the least make them permanently deaf and in sample rate it would mean pitch shifting the highest frequencies in the music down by more than 2 octaves, while leaving the lower frequencies unaltered (otherwise they would be too low to be heard!)._

 

Weird analogies -- and definitely inadequate. The equivalent to a high bit depth would be a high bit depth -- so high dynamic accuracy at ultra-low levels. The equivalent to increased sampling rate would be finer detail (e.g. by finer printing screen) and increased sharpness.

 The equivalent to your pitch shift in the field of photography would be a shift in the electromagnetic spectrum (e.g. green would turn to ultraviolet or the like).
.


----------



## gregorio

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Now you lost me. Are you saying that WaveLab doesn't do the low-pass filtering right during downconversion? Which kind of filter(ing) would do it right, then, in your opinion?BTW, I also compared the original 44.1-kHz version of the «Dragon Boats» with the hi-rez version, to the same avail.

 It seems that you can't avoid brickwall filtering (no news) -- hence you can't avoid the flaws you attribute to it.
._

 

"Doing it right" is a contentious issue. I know that a lot of design and processing power is thrown at "doing it right" in high end ADCs. What I do know is that it's very difficult to avoid artifacts with a steep brickwall filter. But brickwall filters aren't all created equal, there are quite a number of different ways an anti-alias filter can be implemented. I posted a link earlier in this thread to a really good paper on digital filters. The exact implementation of what is done by Wavelab or indeed inside ADCs with regards to brickwall filters is beyond my knowledge and to an extent probably considered by some to be a trade secret. To be honest Wavelab is not generally used in commercial studios and I don't believe is considered "professional" quality more of a pro-sumer product. I don't know what dithering algorithms are used by Wavelab either.

 Sorry, I haven't really answered your question. I would like to run the test file "Dragon boat" myself to find out what is really going on but my system is currently configured for a project I'm working on and I won't want to mess with it until I'm finished on the 7th April.

 G


----------



## linuxworks

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Yes. -- Pixel number and sharpness on the other hand would represent the sampling rate. So if you want the pictures to be enjoyed on a large poster format, better leave it in hi-rez. Some audiophiles listen to their music in poster format (think electrostatic headphones, HD 800, high-end speakers...).
._

 

shooting 'for posters' does NOT need high res!

 if you are standing next to the poster, sure. but photogs don't shoot 'for that' - they shoot regular slr style res and when you get noise, well, you get noise. posters, like high def tv's, only really look good when there is PROPER viewing distance. you can't judge a photo by what it looks like an inch away from your face (so called pixel-peeping).

 I might continue the analogy, though; that headphone listening IS more like looking at a photo closer-up than using home speakers would be. flaws in the material and the amp chain are more noticeable in phones than speakers.

 funny that I find that almost all my music sounds 'fine' on speakers but once I start to plug in phones, the flaws all come out and sound ugly. having TOO high a res can hurt you, in some situations (like playback)


----------



## gregorio

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Weird analogies -- and definitely inadequate. The equivalent to a high bit depth would be a high bit depth -- so high dynamic accuracy at ultra-low levels. The equivalent to increased sampling rate would be finer detail (e.g. by finer printing screen) and increased sharpness.

 The equivalent to your pitch shift in the field of photography would be a shift in the electromagnetic spectrum (e.g. green would turn to ultraviolet or the like).
._

 

Exactly, to hear the accuracy at very low levels in 24bit audio you would have to turn your amp up so high that when a sound came in near the high end of the dynamic range it would probably put the listener in a coma (no joke)!! But as I've already mentioned there is no equipment out there that can actually resolve 24bit audio. And yes, your analogy with using a higher sample rate would mean at 192kFs/s three quarters of the colours used would be beyond the light frequencies visible to a human being.

 In other words, a correct analogy would be a poster which included colours so bright that looking at them would cause permanent blindness and where three quarters of the colours used could only be seen or appreciated by bees!! Unless of course you are a 'visiophile' in which case those ultra-violet colours and x-rays definitely improves the percievable quality of the image 
	

	
	
		
		

		
		
	


	




 G


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Exactly, to hear the accuracy at very low levels in 24bit audio you would have to turn your amp up so high that when a sound came in near the high end of the dynamic range it would probably put the listener in a coma (no joke)!! But as I've already mentioned there is no equipment out there that can actually resolve 24bit audio. And yes, your analogy with using a higher sample rate would mean at 192kFs/s three quarters of the colours used would be beyond the light frequencies visible to a human being.

 In other words, a correct analogy would be a poster which included colours so bright that looking at them would cause permanent blindness and where three quarters of the colours used could only be seen or appreciated by bees!! Unless of course you are a 'visiophile' in which case those ultra-violet colours and x-rays definitely improves the percievable quality of the image 
	

	
	
		
		

		
		
	


	


_

 

Your examples are funny, but not adequate. No need to defend 24 bit or 14/16 bit in photography other than for editing purposes, but I definitely see an audible advantage in hi-rez sampling rates and don't even exclude a benefit from 192 kHz. Your reasoning isn't entirely convincing to me. Your excuses for the brickwall filter being difficult to implement properly rather come across as an attempt to defense your current point of view. If it's indeed just the implementation which is so hard to do perfectly, then it lacks perfection in every environment I've compared hi-rez to 44.1 kHz.
.


----------



## evilking

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_...and don't even exclude a benefit from 192 kHz. Your reasoning isn't entirely convincing to me._

 


 What? His reasonings for greater than 96khz recording isn't convincing?

 Let me guess, only when you've heard it?

 Say this never stops, sampling frequency only increases with time. Next year, Sennheiser discovers some new microphone construction that can recording upto 200khz! Wow, it's incredible, perfect straight frequency response right upto 200khz. Would you think a theoretical 384khz recording has some benefits? Even though those high frequencies are used today for finding fish? No, that would be ridiculous. Then why doesn't that apply for 192khz recording?








 EK


----------



## JaZZ

Quote:


  Originally Posted by *evilking* /img/forum/go_quote.gif 
_What? His reasonings for greater than 96khz recording isn't convincing?

 Let me guess, only when you've heard it?_

 

Indeed. I also need more convincing arguments than yours. 
	

	
	
		
		

		
		
	


	



.


----------



## CDBacklash

I am afraid I have to disagree.
 "There is no difference between 24bit and 16bit other than higher dynamic range". 
 The potential for a larger dynamic range is great and nothing to be scoffed at. As for there being "no additional resolution", nothing really could be further from the truth. Have you listened to any 8bit recordings recently? There is little high pitches to speak of (and definitely no sparkle), and in most encodes there is a pleasant rolloff.
 Why is this so? Well, high frequencies are harder to put in the digital realm and need more data to be represented accurately (or at all in some extreme cases). The spectrum is expanded from 16 bit when it is in 24bit. The resolution is "higher".
 16bit can extend beyond the range of human hearing, but there is little study on whether frequencies above conventional hearing have an impact on the overall sound on the rest of the song through constructive interferance where as removing a lot of highs will leave just the fundamental and things will sound nearly identical to eachother - you lose the character that is violin or flute (produce a similar wave form and have similar attacks decays so on). Logically, and in my opinion audibly, you have extra clarity and detail in 24bit PCM resolutions because of this.
 Here i become a little less certain of what I am talking about but i try to express normally. I think 16bit PCM is limited to 96khz sampling, and from a technical perspective you can see aliasing in lower frequencies than in 24bit PCM which allows 192 khz sampling. Even if you can not hear the harmonic being played by the headphones you will be able to hear the alias
 correct me if i am wrong on any of this, but I am quite sure this is how it works in the digital


----------



## Clutz

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_Have you listened to any 8bit recordings recently? There is little high pitches to speak of (and definitely no sparkle), and in most encodes there is a pleasant rolloff._

 

He's not arguing that there isn't a benefit between going from 8 bit to 16 bit, he's saying there isn't any benefit from going between 16 bit to 24 bit. 8 bits represents 2^8= 256 possible levels, 2^16= 65536, and 2^24 = 16777216. Those are totally different things. The benefit you get from increasing resolution (bit depth) is asymptotic. There is a maximum resolution that you are capable of discerning- whatever that may be- and storing the data (music in this case) at any higher resolution doesn't give you any additional benefit because you cannot discern it. The second issue is that as your resolution increases, background noise caused by the various electronic components sets a certain practical maximum level of resolution. This is the difference between accuracy and precision. gregorio is arguing that going from 16 bit to 24 bit provides no benefit because of both of these reasons. The nyquist-whatever sampling theory (and theory = law, it doesn't mean some random unproved ideas: e.g. the theory (law) of gravity). says that you need to sample a frequency at approximately 2X the frequency you wish to reconstruct, so if you want to be able to reconstruct a 20 kHz tone, you need to be able to sample that tone at 40 kHz (and CDs sample at 44.1 kHz). Unless you're like 12 years old, you can't hear much beyond 20 kHz. 

 Now come the arguments of people professing that they can hear greater than 20 kHz. Sorry, you probably can't - even if you think you can hear it. Take a tone generator, set your system to a given volume, and listen to tones generated at 100, 200, 500, 1000, 2000, 5000, 10000, 12000, 15000, 20000. Set the volume to a level where you would normally listen when playing either the 500, 1000, or 2000 hz tone, and then proceed through the rest without adjusting the volume. Now being honest, can you really hear 20,000 hz? Science suggests that if you're an adult male you probably can't. Simply saying that you can won't convince me either. I'm 31, and seven years ago when I had my ears tested I could hear maybe up to 20 kHz, but the sound was so faint that it didn't really matter much. 

 This is why medical researchers study the effect of drugs using double blind studies- so that neither the individual being measured, nor the individual measuring knows whether they are being given the real treatment or being given a placebo. Just wanting something to be true is enough in order to convince yourself that it is. Some small fraction of the adult population maybe able to hear 20 kHz without drastically increasing the volume, but it will be a very small percentage. 

  Quote:


 Why is this so? Well, high frequencies are harder to put in the digital realm and need more data to be represented accurately (or at all in some extreme cases). The spectrum is expanded from 16 bit when it is in 24bit. The resolution is "higher". 
 

Dude, he understands this - don't be patronizing. 

  Quote:


 16bit can extend beyond the range of human hearing, but there is little study on whether frequencies above conventional hearing have an impact on the overall sound on the rest of the song through constructive interferance where as removing a lot of highs will leave just the fundamental and things will sound nearly identical to eachother - you lose the character that is violin or flute (produce a similar wave form and have similar attacks decays so on). 
 

If you're going to engage in a scientific discussion and reference a paper supporting your position you are remiss if you do not cite it specifically. 

  Quote:


 Logically 
 

You haven't made any arguments from which any conclusions can be deduced 'logically'.


----------



## CDBacklash

.
 a. double blind testing isnt a good testing method
 b. Interference - Wikipedia, the free encyclopedia this applies to all independant frequencies and also harmonics within the same system (even those outside the hearing range no matter how subtle the impact is there).
 Got a headphone that can produce 5hz? Try boosting 5-15 hz really loud for me and see how it sounds 
	

	
	
		
		

		
		
	


	




 It wont be the same.
 I dont know about you, but since I am hearing the effect of harmonics and the like in real life I want to hear them at home, no matter how audible the effect is.
 c. please do not be rude to me because of my poor english next time
 d. I never mentioned anything abotu being able to hear above 20khz, although 2 years ago at last test i was reported to have had a range of 17hz-22.2 khz (the latter of which extends beyond the rule you provided for 44.1khz). Not that this really affects the "music". I am not listening for > 13 khz sounds when i listen to a song, but i hear them and the effect they have on other frequencies.


----------



## Clutz

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_.
 a. double blind testing isnt a good testing method_

 

According to whom? It is the gold standard in biomedical research. What do you propose as a better alternative?


  Quote:


 Got a headphone that can produce 5hz? Try boosting 5-15 hz really loud for me and see how it sounds 
	

	
	
		
		

		
		
	


	




 It wont be the same.
 I dont know about you, but since I am hearing the effect of harmonics and the like in real life I want to hear them at home, no matter how audible the effect is. 
 

But we're not talking about 5-15 hz, so this is irrelevant. 44.1kHz sampling already can deal with 5-15 hz. We're talking about high frequencies. I don't understand your point.

  Quote:


 c. please do not be rude to me because of my poor english next time 
 

I wasn't rude to you because of your poor English. I wasn't rude to you at all. I was rather pointedly arguing that you hadn't supported your conclusions.

  Quote:


 d. I never mentioned anything abotu being able to hear above 20khz, although 2 years ago at last test i was reported to have had a range of 17hz-22.2 khz (the latter of which extends beyond the rule you provided for 44.1khz). Not that this really affects the "music". I am not listening for > 13 khz sounds when i listen to a song, but i hear them and the effect they have on other frequencies. 
 

Except the whole point of higher frequency sampling rates is in order to get a better representation of higher frequency sounds.


----------



## Aleatoris

Quote:


  Originally Posted by *Clutz* /img/forum/go_quote.gif 
_(and theory = law, it doesn't mean some random unproved ideas: e.g. the theory (law) of gravity)._

 

No sir, theories are theories. Based on facts and provable experiments, yes. But they are not complete laws as we do not have a complete understanding of these things. Today's theories are tomorrow's jokes (some times).

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_.
 a. double blind testing isnt a good testing method
 b. Interference - Wikipedia, the free encyclopedia this applies to all independant frequencies and also harmonics within the same system (even those outside the hearing range no matter how subtle the impact is there)._

 

You need to read the first lines:
 Interference usually refers to the interaction of waves which are correlated or coherent with each other, either because they come from the same source or because they have the same or nearly the same frequency.

 Two non-monochromatic waves are only fully coherent with each other if they both have exactly the same range of wavelengths and the same phase differences at each of the constituent wavelengths.

  Quote:


 Got a headphone that can produce 5hz? Try boosting 5-15 hz really loud for me and see how it sounds 
	

	
	
		
		

		
		
	


	




 It wont be the same.
 I dont know about you, but since I am hearing the effect of harmonics and the like in real life I want to hear them at home, no matter how audible the effect is.
 c. please do not be rude to me because of my poor english next time
 d. I never mentioned anything abotu being able to hear above 20khz, although 2 years ago at last test i was reported to have had a range of 17hz-22.2 khz (the latter of which extends beyond the rule you provided for 44.1khz). Not that this really affects the "music". I am not listening for > 13 khz sounds when i listen to a song, but i hear them and the effect they have on other frequencies. 
 

I haven't heard of any headphones that can do 5Hz-15Hz. Of course, I haven't ever looked at >$1000 dollar phones or electrostats. Also "really loud" may end up breaking your phones because they weren't designed to... play those frequencies "really loudly".

 I don't think anyone is taking a jab at your english. I think it's just fine.


----------



## CDBacklash

@aleatoris there are plenty that will. Ultrasone proline, and dt770. I think HD650 goes down to 7 or 12 within the scale they used for their measurement (but will produce lower than taht i think).
 @clutz double blind is a bad test when it comes to audio. No one hears exactly the same as someone else (both in terms of actual hearing and what they perceive which is where subjectivity comes into it). If you use the same person they will hear differences that arent there (and also not hear ones that are) because they will know in most cases that it is a double blind test or not be paying attention to that area of the music enough. Results from it are never to be taken seriously, really whether they be "i heard a difference" or "I could not hear a difference".
 @aleatoris the lines about the waves being similar frequencies involves the aspect of "beats". Beats do not exist nearly as much as they do in that case (because they are out of sync for a longer period of time due to doppler-like effects when their frequencies are comparable). Beats always exist from a complex source but become "inaudible" (i.e. the smooth sound we hear in our ears). Superposition applies to every sound frequency in a given system. even wikipedia knows this "interferance >usually<" etc not to mention the sounds are coming from the same source when you listen to music
 @clutz the low frequency is an extreme example of what high frequencies will do to the sound because there is more energy behidn the production of lower frequencies and the result will be more extreme. A similar thing happens with high frequencies only to a lesser degree.


----------



## Aleatoris

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_@aleatoris there are plenty that will. Ultrasone proline, and dt770. I think HD650 goes down to 7 or 12 within the scale they used for their measurement (but will produce lower than taht i think).
 <..snip..>
 @aleatoris the lines about the waves being similar frequencies involves the aspect of "beats". Beats do not exist nearly as much as they do in that case (because they are out of sync for a longer period of time due to doppler-like effects when their frequencies are comparable). Beats always exist from a complex source but become inaudible. Superposition applies to every sound frequency in a given system._

 

EDIT: Yes, those phones can produce those signals, but the point of my post was that if you EQ the crap out of the low frequencies, the signals will most likely get distorted.

 huh? Superposition doesn't mean that high frequency signals affect lower frequency signals... the waveforms are just added together to produce something more complex. The audible bits are still audible (without much noticable change) and the inaudible bits are... inaudible. If you take out the highs altogether... well, nothing really happens to the audible signal.

 Edit, I'm a retard.


----------



## CDBacklash

Dont worry about it, it's confusing stuff. The effect when listening to recorded music is a bit more extreme than the real world on recordings where it was one room and omnidirectional microphones because it exists in each mic with sound bleeding and then it is multiplied really.


----------



## Clutz

Quote:


  Originally Posted by *Aleatoris* /img/forum/go_quote.gif 
_No sir, theories are theories. Based on facts and provable experiments, yes. But they are not complete laws as we do not have a complete understanding of these things. Today's theories are tomorrow's jokes (some times)._

 

I disagree because there are no such things as 'scientific laws'. The basic idea of science is that all ideas (hypothesis or theory) must be disprovable. Calling something a 'law' means that it impossible to disprove, and is therefore inherently a-scientific. What the public colloquially call 'scientific laws', are in fact 'scientific theories'. The 'Law of Gravity' results from Classic Newtonian Mechanics or General Relativity which are theories. I stand by my original statement.


----------



## scompton

Quote:


  Originally Posted by *Clutz* /img/forum/go_quote.gif 
_I disagree because there are no such things as 'scientific laws'. The basic idea of science is that all ideas (hypothesis or theory) must be disprovable. Calling something a 'law' means that it impossible to disprove, and is therefore inherently a-scientific. What the public colloquially call 'scientific laws', are in fact 'scientific theories'. The 'Law of Gravity' results from Classic Newtonian Mechanics or General Relativity which are theories. I stand by my original statement._

 

Not that I disagree in general, but I thought law was applied only to those theories that were proved mathematically. Newton came up with calculus just so he could prove this theories. I could be completely wrong


----------



## gregorio

Quote:


  Originally Posted by *Aleatoris* /img/forum/go_quote.gif 
_No sir, theories are theories. Based on facts and provable experiments, yes. But they are not complete laws as we do not have a complete understanding of these things. Today's theories are tomorrow's jokes (some times)._

 

I think you may have misunderstood the nature of digital audio and the theeory behind it. In many cases, theories are developed to explain how the universe around us works. This means that as our understanding and observational abilities of the universe improves, so sometimes the theories need to be changed or modified. However, this is *not* the case with digital audio. Digital audio was not discovered and then a theory invented to explain it, it's the other way around. The Nyquist theory was created in the 1920s and then when technology had improved in the early 1970s Digital Audio was invented based on the theory. In other words, if the Nyquist theory is wrong, digital audio would not exist.

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_I am afraid I have to disagree.
 "There is no difference between 24bit and 16bit other than higher dynamic range". 
 The potential for a larger dynamic range is great and nothing to be scoffed at. As for there being "no additional resolution", nothing really could be further from the truth. Have you listened to any 8bit recordings recently? There is little high pitches to speak of (and definitely no sparkle), and in most encodes there is a pleasant rolloff.
 Why is this so? Well, high frequencies are harder to put in the digital realm and need more data to be represented accurately (or at all in some extreme cases). The spectrum is expanded from 16 bit when it is in 24bit. The resolution is "higher".
 16bit can extend beyond the range of human hearing..._

 

Sorry, but I'm afraid you have misunderstood how digital audio works. The bit depth in digital audio encodes the dynamic range, it is the sample rate that is responsible for the frequency range. 8bit 44.1kFs/s has exactly the same frequency range as 24bit 44.1kFs/s, both max out at 22,050Hz. This cannot be disputed as it is a basic tenet of the Nyquist Sampling Theorem. With lower bit depths (say 8bit) the dynamic range (48dB) might be such that the dither noise is noticeable enough to mask the high frequency content. But with 16bit the noise floor is so low that it cannot be heard and therefore cannot mask the high frequency content. So in fact when you say 8bit recordings have no high frequencies you are mistaken (unless there is also a lower sampling rate). 8bit 44.1kFs/s actually has more high frequency content than 16bit 44.1kFs/s because there is the program material plus 48dB more noise distributed throughout the 22,050Hz frequency range.

 I also think you need to read my original post to understand why the greater dynamic range of 24bit compared to 16bit is neither desirable nor even possible.

 Hope this clears things up a bit,

 G


----------



## mape00

Quote:


  Originally Posted by *scompton* /img/forum/go_quote.gif 
_Not that I disagree in general, but I thought law was applied only to those theories that were proved mathematically. Newton came up with calculus just so he could prove this theories. I could be completely wrong_

 

Kind of.

 You can 'prove' laws within theories. For example, conservation laws follow from symmetries (Noether's theorem). For example, the law of conservation of energy exists because the laws of physics are (assumed to be) time invariant, i.e. the same today as they were yesterday and will be tomorrow. Momentum is conserved because of translational invariance. Angular momentum because of rotational invariance. Etc. However these 'laws' are just corollaries of the theory we started with, and aren't truer than the original theory in any sense.

 Proof in physics ultimately comes from experiments. Newton had lots of astronomical data and the famous Keplererian laws, which were based on empirical data and seemed very accurate, but were not really understood. Newton didn't need calculus to _prove_ his theory of mechanics and gravitational law, but it was necessary for the mathematical description of mechanics (velocities, accelerations etc.), and from the simple starting points of his three laws, he could finally derive Kepler's laws. In a way he 'proved' them, but the point was that it was a test of the validity of his theory.


----------



## Rempert

If a mosquito sneezes in a noisy factory, has it contributed anything to the noise problem?

 Mastering engineers claim to hear differences between various dithering algorithms and choose the best one for that particular job. These noise-shaped dithering algorithms are referenced in the original post. How is the mastering engineer supposed to choose something he cannot hear? If the mastering engineer can hear coloration to the piece caused by the dither, surely an audiophile with top line gear and a treated listening room could as well. If coloration from the dither is audible, and distortion from truncation if one chooses not to dither even worse, then one must conclude that an audible difference can be heard between 24 bit and 16 bit audio without raising peak levels so high as to cause deafness or deathness. That dither is even mentioned in this discussion suggests an acknowledgment that 16 bits worth of signal to noise ratio is not foolproof. If 96 db were a great overkill, truncation distortion would be a purely academic topic and dither would be some irrelevant relic of 8 bit audio.

 Don't get me wrong. If we pick a random pop song and make a proper blind test for this, my money says we all fail. But then, almost all of us would fail the -V0 mp3 test too...


----------



## Aleatoris

Quote:


  Originally Posted by *Rempert* /img/forum/go_quote.gif 
_If a mosquito sneezes in a noisy factory, has it contributed anything to the noise problem?_

 

No, but I want to be able to hear that moquito sneeze, dammit.


----------



## manaox2

Quote:


  Originally Posted by *Aleatoris* /img/forum/go_quote.gif 
_No, but I want to be able to hear that moquito sneeze, dammit._

 

You hear it. You get malaria. Bill Gates will be happy to help. http://www.foxnews.com/story/0,2933,488348,00.html


----------



## gregorio

Quote:


  Originally Posted by *Rempert* /img/forum/go_quote.gif 
_If a mosquito sneezes in a noisy factory, has it contributed anything to the noise problem?

 Mastering engineers claim to hear differences between various dithering algorithms and choose the best one for that particular job. These noise-shaped dithering algorithms are referenced in the original post. How is the mastering engineer supposed to choose something he cannot hear? If the mastering engineer can hear coloration to the piece caused by the dither, surely an audiophile with top line gear and a treated listening room could as well. If coloration from the dither is audible, and distortion from truncation if one chooses not to dither even worse, then one must conclude that an audible difference can be heard between 24 bit and 16 bit audio without raising peak levels so high as to cause deafness or deathness. That dither is even mentioned in this discussion suggests an acknowledgment that 16 bits worth of signal to noise ratio is not foolproof. If 96 db were a great overkill, truncation distortion would be a purely academic topic and dither would be some irrelevant relic of 8 bit audio._

 

Good points but not quite how it works in practice. There are a few reasons for this but they are not so easy to explain without using some terminology:

 1. Most applied dithering algorithms are incorporated into limiters. So when you master down to a distribution format you are not just applying dither and a re-quantising process but also applying compression with an infinte ratio (limiting). So differences between dithering programs also include the sonic qualities of the limiter, which in general are rather noticeable.

 2. There are a number of dithering algorithms used and they all have slightly different properties and uses. For example, Type 1 Noise Shaped Dither is a dither algorithm which most strongly re-distributes the dither noise to extremes of the frequency spectrum where it is less percievable. This gives CD a perceptual dynamic range of 120dB. In other words, the dither noise would appear to the listener to be 120dB lower than the loudest noise in the music track. However, we only use this noise shaped dither as a final process, if we used this type of dither whenever we processed a channel in the mix and then sum those channels together, we are potentially going to hear the dither noise because it is an accumulation of dither noise concentrated a small frequency band. So in this case we would use a non-noise shaped dither algorithm where the noise is spead evenly across the whole frequency spectrum.

 3. Sometimes, to check what is really happening, a small and very quiet section of the music is selected and this section is played back at high volume (with the amp whacked up!) to identify any artifacts near the noise floor. The point is that we can instruct the software to just playback the selection, so we avoid the risk of playing back much higher dynamic material which would possibly blow our monitors and/or our ears. So we often check what is near the noise floor even though we know the consumer won't be able to hear it. Not everyone does this but those of us with a lot of professional pride and who are really picky about SQ often do.

 Compared to noise-shaped dither, truncation distortion could sound 30dB louder. The other point is that truncation distortion is not un-correlated noise like dither, this means that truncation distortion can act (modulate) with the program material!! So although truncating distortion going from 24bit to 16bit is very unlikely to be heard, it's affects on the program material may be noticeable.

 Sorry this post was a bit more technical, hopefully though it answers your post?

 G


----------



## scompton

Good thing I put in the caveat that I could be completely wrong


----------



## Clutz

Quote:


  Originally Posted by *scompton* /img/forum/go_quote.gif 
_Not that I disagree in general, but I thought law was applied only to those theories that were proved mathematically. Newton came up with calculus just so he could prove this theories. I could be completely wrong_

 

I'm not certain this is true. If you prove a theory mathematically, all you've done is proven the mathematical validity of that theory. You could come up with a legitimate mathematical proof for a theory that is imperfect. There are many mathematical proofs for string theory, but all of those proofs are contingent upon the underlying physical understanding of the theory being correct. The math for the theory may be correct, but that doesn't mean it's a perfect representation of nature.

 [Note: I didn't see mape00's response, otherwise I wouldn't have responded. I did not intend to belabor the point]


----------



## CDBacklash

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Sorry, but I'm afraid you have misunderstood how digital audio works. The bit depth in digital audio encodes the dynamic range, it is the sample rate that is responsible for the frequency range. 8bit 44.1kFs/s has exactly the same frequency range as 24bit 44.1kFs/s, both max out at 22,050Hz. This cannot be disputed as it is a basic tenet of the Nyquist Sampling Theorem. With lower bit depths (say 8bit) the dynamic range (48dB) might be such that the dither noise is noticeable enough to mask the high frequency content. But with 16bit the noise floor is so low that it cannot be heard and therefore cannot mask the high frequency content. So in fact when you say 8bit recordings have no high frequencies you are mistaken (unless there is also a lower sampling rate). 8bit 44.1kFs/s actually has more high frequency content than 16bit 44.1kFs/s because there is the program material plus 48dB more noise distributed throughout the 22,050Hz frequency range.

 I also think you need to read my original post to understand why the greater dynamic range of 24bit compared to 16bit is neither desirable nor even possible.

 Hope this clears things up a bit,

 G_

 

Sampling rate does not eliminate highs it simply causes aliasing and misrepresentation of them. It is typical for encodes to apply a lowpass that eliminates the frequencies - and many will do this based on a combination of bit depth and sampling frequency. Typical sampling rates for 8 bit are 22.05khz, and not 44.1khz. PCM will accept 44.1 khz for 8bit audio, but it is hardly ever used: see game boys, the NES and the good old telephone. Sure, your numbers are correct in theory, but they are not default. As for your precious 16bit, a lot of studios are now recording in 24bit and then converting 16bit for CD media (oh no!).
 You do know amplitude quantisation occurs in all digital media where each waveform is not represented as a complete picture, but merely as a snapshot, right? The resolution in 24bit audio is much higher and I am pretty sure you can get a much smoother representation of dynamic (note: dynamic, not contour) whereas noise will be added to lower bitdepths to snap each sample to the supported resolution. It's not like 24bit audio is just adding more volume onto the audio. There is the capacity to use a larger number of different amplitudes per sample! There are more ticks to snap to in a 24bit digital recording and it will be a more accurate representation of the sound if done correctly. You can have a "louder" signal from 24bit audio which can then be scaled down (a lovely thing called the volume knob), to a level that is not disimilar to the same song in a 16bit audio stream. However the bits will _not_ be the same. There will be greater dynamic accuracy in the 24bit stream because you can use the extra room up the top and then audibly reduce the distance between each quantisation with both encode and the volume knob...
 As for the additional dynamic range of 24bit audio being undesirable, I think you are pressing it just a little to far with this statement. Welcome to the *sound engineer*. Many sound engineers are not 12 yearold kids who love dynamic range and are going to have passages that are terribly soft with drum hits that are amazingly loud. Most engineers will apply a brickwall (that is a compression filter with a small attack and release time) and then increase the volume of the track. My personal favourites will just record and slap it on tape. I doubt 24 bit audio will help us with regards to the loudness war. However, with regards to recording one shout and then putting it on media "as is" I think 24bit resolution is definitely worthwhile. Why would I want my drums to have less impact than they were intended to by the artists? Musicians do have ears.
 Thread is becoming a bit monotonous and boring... If you are afraid of change, or cant afford or get high resolution gear/audio at the moment dont worry about it, the price will come down, and if things pan out the way they have for the last while with digital media, 24bit will come to the fore later on in this life. 16 bit audio is not perfect and as such it does not do justice to music, albeit it performs very well. Digital storage mediums will _never_ do audio true justice because of the quantisation that occurs. Even then one could argue that the warm hum you get from analog reproduction is not doing it justice either (although i particularly like it). 24bit audio has the capacity to store orchestra hits or heaven forbid cannon blasts (from a particular epic) far more realisticly because of the dynamic range without the overall sound becoming too loud.

 We are also seeing a generation with large amounts of hearing loss due to pressing their volume switches upwards because thats what compressed music wants you to do. Have you ever noticed that you often turn down the volume on drums with good 'crack' to them? Perhaps, if the mainstream industry plays it fairly perhaps we may be able to avoid a generation of deaf people.

 Whether or not you or I can personally hear a difference at this point in time, the fact of the matter is there is a difference between 16 and 24 bit audio and frankly you are starting to smell a bit like a troll. Then again, maybe everything I learned in college is false and the big bad internet has one-upped me again.
 (sorry for bad english)
 **the bit-depth war is getting to me just as much as the loudness war. Let better technology come all it wants. It's not going to affect you at all.


----------



## Aleatoris

uh guy, he IS a sound engineer.


----------



## CDBacklash

uh, so was I. If you were to have a large amount of loudness followed by a very quiet section, with enough clarity in your equipment (which is always getting better) you may be able to pick up on the noise floor quite easily, especially as the sound approaches -80dBFS.
 What the OP wants you to believe is "great sound" is actually just a dither doing its job and there is nothing particularly special about that at all.
 **what he is saying is nothing particularly new or exciting. This has been a hot topic over at HAF for a long time, partciularly the cannon in teldecs 1812 overture which live exceeds the dynamic range by enough that can be considered acceptible in 16bit audio (in my opinion), where noise may become audible.
 **the main problem with with 24bit playback atm is that the SNR of most equipment in use at the moment is only slightly better than pure 16bit. You can stretch this through simple thermodynamics (that is, cooling the equipment such as treating it cryogenically) and get a much better noise resolution but as far as approaching 24bit at this point in time i think using 24bit for playback is a overreaching a bit. It's sort of like how 64bit operating systems came out too early for 90% of users to have more than 3gb of ram without knowing how to use it in a 32bit OS. In the studio 24bit is definitely the way to go at the moment for the dynamic range, although the noise floor you hear is more likely to come from the equipment than the signal.
 **if the OP still disagrees, perhaps he'd like to find and ABX an unshaped and undithered test tone... oh well, i'm done with this before it develops into a flame war...
 to save him time he could take it two ways from here
 a. disk space to which i have no response but get bigger storage and b. he could argue whether or not differences are audible outside of test tones (i.e. in music). In which case there is no definite answer at the moment. "no" would be the current answer, but it has not really been tested whether it is possible by a large number of people. There are those out there that believe emotional response is more real from higher resolution/sample rates due to the affect of high frequencies on the brain etc.


----------



## Arjisme

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_You do know amplitude quantisation occurs in all digital media where each waveform is not represented as a complete picture, but merely as a snapshot, right?_

 

Did you read the first post in this thread?

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_The problem is that an analogue waveform is constantly varying. No matter how many times a second we measure the waveform or how many bits we use to store the measurement, there are always going to be errors. These errors in quantifying the value of a constantly changing waveform are called quantisation errors._


----------



## gregorio

CD Backlash:- You need to read this thread carefully and understand it before you come out with a whole bunch of statements based on assumption which are incorrect. For example:

  Quote:


  Originally Posted by *CDBacklash;5553024





			
				CDBacklash[/b said:
			
		


			/img/forum/go_quote.gif 
Sampling rate does not eliminate highs it simply causes aliasing and misrepresentation of them.

Click to expand...

*
*


			
				CDBacklash[/b said:
			
		


			Oh please! You can't have "aliasing and misrepresentation" of the recorded waveform otherwise it's going to be a recording full of complete rubbish and no one would be using digital audio. This is why all ADCs have an anti-alias filter, it's not optional, it is an absolute requirement and it's not based on bit depth but purely the Nyquist Point, IE. roughly half the sampling frequency. What college did you go to that taught you the rubbish you are spouting?

  Quote:


  Originally Posted by CDBacklash /img/forum/go_quote.gif 
As for your precious 16bit, a lot of studios are now recording in 24bit and then converting 16bit for CD media (oh no!).

 

That's hardly a suprise to me, as you would know if you'd bothered to read this thread before trolling! I started using high bit rates for recording when you were probably still in diapers so read the damn thread!

  Quote:


  Originally Posted by CDBacklash /img/forum/go_quote.gif 
Digital storage mediums will _never_ do audio true justice because of the quantisation that occurs... You do know amplitude quantisation occurs in all digital media where each waveform is not represented as a complete picture, but merely as a snapshot, right? ...

 

And you do know that a dithering quantiser was invented probably before you were born which converts the interpolation errors during quantisation into un-correlated noise and results in a perfectly linear reproduction of the original waveform ("complete picture"), right? Obviously not!

  Quote:


  Originally Posted by CDBacklash /img/forum/go_quote.gif 
There are more ticks to snap to in a 24bit digital recording and it will be a more accurate representation of the sound if done correctly.

 

Now you are completely getting yourself screwed up. Ticks are the basic timing units derived from the BPM (PPQN) of your sequencer and have nothing to do with sampling rates, bit depth or indeed digital audio! Or maybe you are confusing the quantising of MIDI data with the quantising of digital audio, the same term but totally unrelated processes. If you mean there are more quantisation levels in 24bit then yes there are, as explained in my original post!!! 

  Quote:


  Originally Posted by CDBacklash /img/forum/go_quote.gif 
Most engineers will apply a brickwall (that is a compression filter with a small attack and release time) and then increase the volume of the track.

 

You've even got this wrong. I take it you are refering to brickwall compression, which if you knew anything about audio engineering is actually called limiting. The difference between a limiter and a compressor is not the attack and release times but the ratio, which for a limiter needs to be infinity:1. And you've really displayed your ignorance as if you compress or limit the track you are decreasing the dynamic range and therefore require a lower bit depth to encode it. I would call the college you attended and ask for your money back!!

  Quote:


  Originally Posted by CDBacklash /img/forum/go_quote.gif 
24bit audio has the capacity to store orchestra hits or heaven forbid cannon blasts (from a particular epic) far more realisticly because of the dynamic range...

 

Exactly, so hands up anyone out there that wants to put on headphones and hear the true volume of what a cannon being fired from 3ft away actually sounds like?

 To be honest, it's hard to find a single sentence in your post which bares any resemblance to how digital audio actually works. My guess is that you are a kid who has just left college and who has completely mis-heard or mis-understood what your lecturer has tried (but obviously failed) to teach you!

 G
		
Click to expand...

*


----------



## CDBacklash

I did, but my english is poor (and so is my memory).
 He makes some valid points however I feel that he is misguided. There is a difference between 16 bit and 24bit audio. Whether current hardware can present it well enough and whether our ears can "get it" well enough is a completely different story.
 There was no myth exploded. Everyone knows that it's much easier to ABX 8bit vs 16 bit (or even 12 bit) than it is to abx 16 vs 24. Some of his arguments like that there is "no difference other than dynamic range" are borderline moronic when there clearly are.
 exiting stage left
 edit: uh no, I'd appreciate if the personal attacks were kept to a minimum.
 A compression filter with 1 sample attack and release time does exactly the same thing as a limiter... You dont require a lower bit depth to do anything when you compress something. You can encode anything however the hell you want.
 As for the cannon being fired? Yes, I will take the real volume. Exposure time for 120 db is 8.5 minutes if i recall correctly and it is halved for a 5 db increase. A cannonblast amongst a loud orchestra will not cause permenant ear damage.
 "Ticks are the basic timing units derived from the BPM (PPQN) of your sequencer and have nothing to do with sampling rates"
 I was using tick as an example of where the quantisation occurs (that is vertically on the typical representation of a waveform), not of anything else. This is at best an english error and I would appreciate you to not make fun of my english as it is not my native language.
 "Oh please! You can't have "aliasing and misrepresentation" of the recorded waveform otherwise it's going to be a recording full of complete rubbish and no one would be using digital audio. This is why all ADCs have an anti-alias filter, it's not optional, it is an absolute requirement and it's not based on bit depth but purely the Nyquist Point, IE. roughly half the sampling frequency. What college did you go to that taught you the rubbish you are spouting?"
 I was talking about in the DAC process, not the ADC. Again i feel like you are making fun of my english... sigh
 When you tell me that a digital storage results in a perfect reproduction of the complete picture, you are telling lies. The picture you are hearing is not the original picture. Close, but not quite. Have a nice day, I don't feel like partaking in a flame war. HAF Go here and have a nice day... i'm sure you'll learn a lot.


----------



## Clutz

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_Close, but not quite. Have a nice day, I don't feel like partaking in a flame war._

 

Dude, this is weak. First you accuse him of being a troll- which was both unfounded and uncalled for, and then you say that he's starting a flame war? *You* threw the first stone. *You* accused him of not knowing what he was talking about, when the problem was that *you* couldn't be bothered to read the whole thread. He's been a member for over a year, and you've been a member for less than a month- and yet despite the fact that you started with unnecessary rhetoric, you accuse him of being a troll?

 I think *you* are a troll.

  Quote:


 This is at best an english error and I would appreciate you to not make fun of my english as it is not my native language. 
 

And stop trying to hide behind your poor English skills. You're out and out attacking people, and then hiding behind your poor English writing skills? That's way weak. Regardless of what your level of English proficiency, we are forced to understand what you mean as a function of what you write. Do you really expect everyone to read your posts and then spend a lot of time analyzing what you meant versus what you wrote? He did not intentionally misinterpret you, that much is clear. 

 Weak!


----------



## CDBacklash

What does join date have to do with anything? (other than nothing). 
 I am also aware that I said something about his opinion first, because I was pretty disgusted by how blatantly he was trying to force his opinion (not to mention itd be amazing if he could preempt my post) on people on an undoubtedly grey area (at this point in time). Since then the thread has degraded into tit-for-tat pseudo-flaming which is why I am now ejecting myself. I have said what I wanted to said and am going to leave it at that.
 Please keep your personal attacks away from me.


----------



## evilking

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_What does join date have to do with anything? (other than nothing). 
 I am also aware that I said something about his opinion first, because I was pretty disgusted by how blatantly he was trying to force his opinion (not to mention itd be amazing if he could preempt my post) on people on an undoubtedly grey area (at this point in time). Since then the thread has degraded into tit-for-tat pseudo-flaming which is why I am now ejecting myself. I have said what I wanted to said and am going to leave it at that.
 Please keep your personal attacks away from me._

 

You come in here, spew seven shades of ******** and now _you're_ leaving the thread?

 Thanks so much for your input. I'm sorry the language barrier has it made so difficult for you to have a rational debate.


 EK


----------



## gregorio

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_I was using tick as an example of where the quantisation occurs (that is vertically on the typical representation of a waveform), not of anything else. This is at best an english error and I would appreciate you to not make fun of my english as it is not my native language._

 

I'm not having a go at your English, I'm having a go at your use of audio engineering terminology. The term "Ticks" is used all over the world including in non-english speaking countries! The correct term for what you are trying to describe is the "sampling point". 

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_I was talking about in the DAC process, not the ADC. Again i feel like you are making fun of my english... sigh_

 

I'm not falling for your lame excuse that English is not your first language, to try and cover up the fact that you have posted a pile of nonsense. I'm a foreign language speaker too, or didn't you notice my name (Gregorio)? And if you were talking about a DAC and not an ADC then you were even more wrong! It is the ADC which performs the quantisation process, not a DAC. It is an ADC which sets the sampling rate and an ADC which requires an anti-alias filter to remove frequencies above the Nyquist Limit.

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_When you tell me that a digital storage results in a perfect reproduction of the complete picture, you are telling lies._

 

Go away and learn your facts, I'm not going to tell you again! I am not telling you lies, I am directly quoting Harry Nyquist and the Nyquist-Shannon Sampling Theorem:

 "In essence the theorem shows that an analog signal that has been sampled can be *perfectly* reconstructed from the samples if the sampling rate exceeds 2B samples per second, where B is the highest frequency in the original signal." - Nyquistâ€“Shannon sampling theorem - Wikipedia, the free encyclopedia

 Maybe you would like to write the word "Liar" on Nyquist's grave stone and while you're at it get a nobel prize for proving the theorem wrong and that digital audio does not exist?

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_A compression filter with 1 sample attack and release time does exactly the same thing as a limiter..._

 

And how many professional compressors have you used which have the option to set the attack and release times to 1 sample? How much the signal is compressed is set with the ratio, the attack and release times just set how quickly the compression ratio is achieved and how long it is held once the threashold has been reached. Again the difference between a limiter and a compressor is the ratio, which is variable on a compressor but fixed at infinity:1 on a limiter. Once the track has been compressed the dynamic range has been reduced and therefore can be encoded with a lower bit rate, what part of this simple explanation do you not understand?

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_edit: uh no, I'd appreciate if the personal attacks were kept to a minimum._

 

Could you please go and look up the english word "hypocrite". So far you've called me a moron, a 12 year old and a liar.

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_Exposure time for 120 db is 8.5 minutes._

 

OK, that has dealt with the maximum perceivable dynamic range of a CD. What does it say about 144dB (24bit). I think you'll find it says permanent hearing damage will occur instanteneously. It is also likely that an instantaneous level of 120dB will severely damage the hearing of a child (World Heath Organisation). So you tell me, how irresponsible would a record label have to be to sell a recording which will literally deafen the people who buy it, how long before they got sued out of existance and the directors imprisoned for corporate negligence? Even if you are completely ignorant of the facts, at least try to apply some logic.

 For anyone else reading this thread, please do *not* try and listen to 120dB signals, even for a short time!!

 G


----------



## CDBacklash

"Could you please go and look up the english work "hypocrite". So far you've called me a moron, a 12 year old and a liar."
 Another english attack. I never called you a 12 yearold, i never called you a moron or a liar. I said IMO your opinions were moronic, and that you'd be lying if you said you can perfectly recreate bits that should have been there from nothing (because what is lost will vary). Keep your opinions all you want, what i have said may have lacked the technical terms and eloquence but they are far from wrong. You cannot recreate what does not exist. Having it there in the first place is the superior option. Enjoy your 16bit audio forever. When higher fidelity sound becomes available for consumer playback and is plausible, I will probably enjoy it just as much


----------



## gregorio

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_"Could you please go and look up the english work "hypocrite". So far you've called me a moron, a 12 year old and a liar."
 Another english attack. Keep your opinions all you want, what i have said may have lacked the technical terms and eloquence but they are far from wrong._

 

You nearly got your English right but I think you meant to end your last sentence with "but they are far from right"?!

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_I was pretty disgusted by how blatantly he was trying to force his opinion on people on an undoubtedly grey area (at this point in time)._

 

I have not forced my opinion on anyone, I have mearly stated the facts of digital audio theory. It's not difficult to understand, the theory was invented and then the technology was developed from the theory and what do you know, the technology works! Where is the grey area and why should it be "at this point in time" when the theory is over 80 years old and the technology nearly 40 years old?

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_You cannot recreate what does not exist. Having it there in the first place is the superior option._

 

Have you ever actually heard of the Nyquist-Shannon sampling theorem? I put a link to an explanation of it on my last message. Go away and read it as you are just making a complete fool of yourself!

 G


----------



## CDBacklash

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_You nearly got your English right but I think you meant to end your last sentence with "but they are far from right"?!_

 

.
 NC
 hydrogenaudioforums. go there, im sure they'll enjoy being lectured by you
 You have not stated fact. You have tried to pass off 24bit audio as a mere gimick that only allows an extra dynamic range when it has the potential to be so much more. Please do a test (double blind if you like) with a test tone and tell me you do not hear a difference.


----------



## Clutz

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_What does join date have to do with anything? (other than nothing)._

 

If anyone is going to be a troll, it isn't going to be someone who has a fairly long posting history, it's going to be some newb trying to stir the pot. 

  Quote:


 I am also aware that I said something about his opinion first, because I was pretty disgusted by how blatantly he was trying to force his opinion (not to mention itd be amazing if he could preempt my post) on people on an undoubtedly grey area (at this point in time). 
 

It's not a grey area - it's simple physics. Just because it's a grey area in your mind, doesn't mean it's a grey area in reality. Get over yourself and your own opinion. You're 21, not god.

  Quote:


 Since then the thread has degraded into tit-for-tat pseudo-flaming which is why I am now ejecting myself. I have said what I wanted to said and am going to leave it at that. 
 

Which *you* started. And since this is the second time you've claimed you were going to end the argument, I'm not holding my breathe. 

  Quote:


 Please keep your personal attacks away from me. 
 

Oh, that's the pot calling the kettle black.


----------



## CDBacklash

Quote:


  Originally Posted by *Clutz* /img/forum/go_quote.gif 
_It's not a grey area - it's simple physics. Just because it's a grey area in your mind, doesn't mean it's a grey area in reality. Get over yourself and your own opinion. You're 21, not god._

 

If it is simple physics then why do people not want to believe supersonic frequencies affecting the sound of audible frequencies? You see it all the time with frequencies interfering with eachother. So what makes it so improbable that frequencies we cannot hear will not affect frequencies we can. Theres a large gap between a live sound and a digital sound. pure 16 bit signal is far noisier than pure 24bit. That will affect hte overall sound as well - whether you can pick it or not, it IS. It's a bit strange to preemptively dismiss something because it seems irrelevant now. The logical thing to do is move forward, which the OP seems to not want to do.
 The grey area is whether or not the difference is audible to the human ear- which i believe i stated quite clearly MANY times.
 There is nothing "physics" about perception unless you want to get into quantum realities which has nothing to do with this.
 Some members here seem to be beyond reason when they want to say that new technologies are futile. Okay, Final Eject.


----------



## manaox2

We got it. Let us make our own decisions. You don't necessarily need to win others, and you won't be able to do if all you do is repeatedly tell us to trust our own ears.


----------



## Clutz

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_Okay, Final Eject._

 

This is the third time you've claimed to be leaving the discussion. I still don't believe you. 

 All of the things you are accusing others of doing, you are in fact doing yourself. 

 Again, for emphasis: You are the pot calling the kettle black.


----------



## gregorio

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_NC
 hydrogenaudioforums. go there, im sure they'll enjoy being lectured by you
 You have not stated fact. You have tried to pass off 24bit audio as a mere gimick that only allows an extra dynamic range when it has the potential to be so much more. Please do a test (double blind if you like) with a test tone and tell me you do not hear a difference._

 

I never said it or implied it was a gimmick, on the contrary it is as I have stated previously, incredibly useful (but not as a playback format), I've used 24bit professionally since you were about 3 years old! Go and read the damn thread!

 24bit does not have the potential for anything other than dynamic range. Please give me the name of your lecturer and the college where you studied.

 Hydrogen audio Forums. Is Bob Katz or Nika Aldrich still on there, two of the world's leading authorities on digital audio? It was Nika who taught me this stuff (when you were about 9 years old) and if you spout your mis-information to Bob Katz (the most respected mastering engineer on the planet) and call him a liar or imply he doesn't know what he is talking about, you will quickly find yourself way more insulted than you could possibly imagine!

  Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_Some members here seem to be beyond reason when they want to say that new technologies are futile._

 

Oh dear, you can't even get this right. Higher than 16bit technology is nearly 20 years old. In terms of digital technology, 20 years is not a new technology but an old technology. It's obviously quite new to you though, judging by your understanding of it!

 The one true thing you have said is that 16bit is way noisier than 24bit. What you failed to mention is that it's 48dB noiser (without the use of noise shaped dither) and that the 48dB of noise starts at -96dB (or -120dB perceptually with noise shaped dither). You tell me who is going to notice noise 60dB below the quietest signal in the most dynamic commercial recording. In fact noise lower than -120dB is roughly the same level of noise caused by the electrons colliding inside a single 3k resistor. If you can do a DBT where you can hear the noise of a single resistor and differentiate it from all the other resistors and circuitry in your signal chain, I will personally write to the Guiness Book of Records on your behalf. Where did I get this figure of 120dB (150dB in some frequency bands) perceptual dynamic range from 16bit? I'm quoting from Bob Katz, now go and argue with him!!

 G


----------



## Aleatoris

Quote:


  Originally Posted by *Clutz* /img/forum/go_quote.gif 
_you were going to end the argument, I'm not holding my breathe. _

 

Dude! it's breath! your english sucks, go away...
 Of course, I'm joking, so please don't hurt me.


----------



## ThePredator

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_If it is simple physics then why do people not want to believe supersonic frequencies affecting the sound of audible frequencies? You see it all the time with frequencies interfering with eachother. So what makes it so improbable that frequencies we cannot hear will not affect frequencies we can. Theres a large gap between a live sound and a digital sound. pure 16 bit signal is far noisier than pure 24bit. That will affect hte overall sound as well - whether you can pick it or not, it IS. It's a bit strange to preemptively dismiss something because it seems irrelevant now. The logical thing to do is move forward, which the OP seems to not want to do.
 The grey area is whether or not the difference is audible to the human ear- which i believe i stated quite clearly MANY times.
 There is nothing "physics" about perception unless you want to get into quantum realities which has nothing to do with this.
 Some members here seem to be beyond reason when they want to say that new technologies are futile. Okay, Final Eject._

 

Huh? You can't hear a frequency higher than the audible range just because it is superimposed on a lower frequency.


----------



## mbd2884

I love gregorios posts. Helps me focus on listening to the music rather than worry about the myths some audiophiles perpetuate.


----------



## elrod-tom

Folks we're going to have to try a little harder to find a more civil tone (be it in 16 bit or 24 bit word length).


----------



## Aleatoris

Quote:


  Originally Posted by *elrod-tom* /img/forum/go_quote.gif 
_Folks we're going to have to try a little harder to find a more civil tone (be it in 16 bit or 24 bit word length)._

 

groaaaan.


----------



## elrod-tom

Quote:


  Originally Posted by *Aleatoris* /img/forum/go_quote.gif 
_groaaaan._

 

I know...couldn't resist. 
	

	
	
		
		

		
		
	


	




 In all seriousness, I don't like some of the namecalling and such that I'm seeing here. If y'all want to keep discussing this, keep it civil. 

 Thanks.


----------



## mbd2884

Hello elrod-tom, what's you current headphone listening setup? Not your bedside or work rig, the main one, in case you have multiple setups. Curious what someone who is clearly focused on the music uses, thanks.


----------



## elrod-tom

At the moment, I use a Wadia 830 and balanced markl-modded Denon D5000's right off the back balanced outs, or same with a Fisher 500C amp (the headphone jack taps the main amp section). I've had all sorts of stuff, from a Headroom Airhead through a Headroom Balanced Desktop (which is a hell of a bargain btw). I just keep playing around with stuff to find the best value.

 OT - sorry...


----------



## mbd2884

Thanks, very cool.


----------



## Pio2001

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_AFAIK, there has never been a DBT between 16bit and 24bit (under controlled conditions) using the same sample frequency, where anyone has been able to tell the difference with any more accuracy than would be expected from chance._

 

In 44.1/16 vs 24/96 tests, the audibility of the quantisation noise have been reported, but at unusual listening levels. The fact is mentionned in Meyer and Moran's paper : Double-blind test of SACD and DVD-A vs. Redbook 16/44 in JAES Septembe - Hydrogenaudio Forums

 I have myself reproduced the result with closed headphones, with a high definition recording that I myself converted to 44.1/16, and at a listening level that I would evaluate around 90 dB RMS / 112 dB for the highest peak. But I did not listen to music at all. Just to the noise during the beginning of the fade-in. 
 The conversion was done with Voxengo R8brain. Playback with Foobar2000 v0.8 (the ABX module equalizes the Replaygain levels). Score 10/10.


----------



## wnmnkh

I had not bothered to read all of previous posts, but I am sure that this topic has been discussed to death in Hydrogen forum.

Frequently Asked Questions - Hydrogenaudio Forums


----------



## wnmnkh

Some quote from Hydrogen Forum.

 Lyx
  Quote:


 We dont need it. It's just virtual useless number-games to give people the incentive to buy new equipment and then re-buy all our music. There are some *technical* arguments for using 48khz instead of 44khz.... but the actual benefit for normal endusers is zero. 
 

Garf
  Quote:


 Well, the advantage of DVDA and SACD is exactly that they are multichannel. And have DRM, which is obviously a big advantage to *some* people. 
 

These two are the conclusions of the topic IMO.


----------



## gregorio

Quote:


  Originally Posted by *Pio2001* /img/forum/go_quote.gif 
_In 44.1/16 vs 24/96 tests, the audibility of the quantisation noise have been reported, but at unusual listening levels. The fact is mentionned in Meyer and Moran's paper : Double-blind test of SACD and DVD-A vs. Redbook 16/44 in JAES Septembe - Hydrogenaudio Forums

 I have myself reproduced the result with closed headphones, with a high definition recording that I myself converted to 44.1/16, and at a listening level that I would evaluate around 90 dB RMS / 112 dB for the highest peak. But I did not listen to music at all. Just to the noise during the beginning of the fade-in. 
 The conversion was done with Voxengo R8brain. Playback with Foobar2000 v0.8 (the ABX module equalizes the Replaygain levels). Score 10/10._

 

AFAIK, the Voxengo R8brain does not apply noise-shaped dither, just standard distribution (wide band) dither. So if you turn your amp right up, you should hear the dither noise louder than on the 24bit version, although you shouldn't hear it a normal listening levels. Using a noise-shaped dither should make the dither noise much less dectable, very close, if not identical to the noise on the 24bit original, even at very high listening levels.

 G


----------



## Pio2001

Voxengo uses gaussian dither with slight noise shaping. Strong noise shaping would have been less, or not audible, I must admit.


----------



## leeperry

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_you'd be lying if you said you can perfectly recreate bits that should have been there from nothing (because what is lost will vary). Keep your opinions all you want_

 

agreed, whatever recordings I did when I was working at a studio, my own masters or some 24/96 recording floating around(Depeche Mode "Violator"/Marvin Gaye "The Marvin Gaye Collection" in lossless 24/96 from the SACD, compressed losslessly from MLP w/ a Lynx2 soundcard).....16/44.1 will NEVER sound as good as 24/96, I don't care what that Nyquist guy said 
	

	
	
		
		

		
		
	


	




 it's always funny to see very smart ppl explaining you that 16/44.1 & 24/96 sound absolutely identical 
	

	
	
		
		

		
		
	


	




 just like reverb trails on properly decoded HDCD sound more natural than the same track converted to 16/44.1

 Rupert Neve has always called the "16/44.1" foolish, and I'd guess that he's sorta aware of the tricks of the trade 
	

	
	
		
		

		
		
	


	




 from If 44.1 digital is imperfect, are audio engineers doomed t.. :

  Quote:


 First, the statement that digital doesn't deliver as competely as analog doesn't necessarily have to do with frequency response, it has to do with the emotional reation in the listener. Whether one delivers hypersonic content that the other doesn't or adds a pleaseing distortion and any other technical difference is irrelevant to the point I'm trying to make. 

 Here's the relevant section: 

*Fletcher*: There has been some measure of debate about bandwidth including frequencies above 20kHz, can we hear them, do they make a difference, etc. 

*Rupert Neve*: OK, Fletch, pin your ears back...back in 1977, just after I had sold the company, George Martin called me to say that Air Studios had taken delivery of a Neve Console which did not seem to be giving satisfaction to Geoff Emmerick. In fact, he said that Geoff is unhappy.... engineers from the company, bear in mind that at this point I was not primarily involved, had visited the studio and reported that nothing was wrong. They said that the customer is mad and that the problem will go away if we ignore it long enough. 

 Well I visited the studio and after careful listening with Geoff, I agreed with him that three panels on this 48 panel console sounded slightly different. We discovered that there was a 3dB peak at 54kHz Geoff's golden ears had perceived that there was a difference. 
 We found that 3 transformers had been incorrectly wired and it was a matter of minutes to correct this. After which Geoff was happy. And I mean that he relaxed and there was a big smile on his face. 

 As you can imagine a lot of theories were put forward, but even today I couldn't tell you how an experienced listener can perceive frequencies of the normal range of hearing. 
 And following on from this, I was visiting Japan and was invited to the laboratories of Professor Oohashi He had discovered that *when filters were applied to an audio signal cutting off frequencies of 20 kHz, the brain started to emit electric signals which can be measured and quantified 

 These signals were at the frequencies and of the pattern which are associated with frustration and anger.* Clearly we discussed this at some length and I also would forward the idea that any frequencies which were not part of the original music, such as quantisizing noise produced by compact discs and other digital sources, also produced similar brain waves.


----------



## CDBacklash

Undithered 16bit sound has actually been ABXed many times on HAF with test tones, and 16bit dithered and 24bit has also been abxed a few times on HAF with test tones.
 Telling the difference with music is a different story.


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_agreed, whatever recordings I did when I was working at a studio, my own masters or some 24/96 recording floating around(Depeche Mode "Violator"/Marvin Gaye "The Marvin Gaye Collection" in lossless 24/96 from the SACD, compressed losslessly from MLP w/ a Lynx2 soundcard).....16/44.1 will NEVER sound as good as 24/96, I don't care what that Nyquist guy said ... it's always funny to see very smart ppl explaining you that 16/44.1 & 24/96 sound absolutely identical._

 

I never said that 16/44.1 & 24/96 always sound identical. This thread was origianlly about 16 vs 24bit. At any sensible hearing level, they cannot be told apart. Under some test conditions is possible to hear the difference but you need very high levels, which precludes listening to material at a normal level as it would be way too loud. It is possible that 96kFs/s could make a difference as mentioned earlier but no one has successfully been able to identify a difference in DBT.

 The work of Professor Oohashi has caused no end of trouble. He produced an infamous paper with DBT proof that frequencies higher than 20kHz could be heard through measuring brain wave activity. This paper has been quoted and used many times. However, it was shown that his methodology was flawed. Using the correct methodology Professor Oohashi's experiments have been repeated many times in various countries but never have the results been repeatable.

 Rupert Neve is one of the most respected manufacturers of studio equipment in the business but his reputation was for analogue products, more than digital. I've heard the story you posted about him in Air Studios but I don't know it's accuracy nor what exactly was heard or what was tested. As I mentioned before, there is some anecdotal evidence for greater than 20k perception.

 Of course perception is very different to hearing. Anyone could percieve 200dB of 30kHz signal, they would percieve a great deal of pain or death!!! Unless we can actually hear beyond 20k though is it really relevant? The question is; how much program material is present at 20kHz+, how much of it can we record and how much can be heard above the noise floor. The answer in my opinion is none and testing would bare this out. Again though, with respect, we got into a discussion with sample rates of 192kFs/s, which is even further outside what is plausible.

 You might think "it's always funny to see very smart ppl explaining you that 16/44.1 & 24/96 sound absolutely identical" but then thousands of smart people professionals and consumers have taken place in number of DBT with higher sample rates in controlled conditions and there has never been a significant result (that wasn't flawed). So I would say "it's funny to see very smart people saying that 16/44.1 sounds worse than 24/96" when all the considerable proof which exists contradicts this statement.

 G


----------



## CDBacklash

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Under some test conditions is possible to hear the difference_

 

Thanks for clearing this up


----------



## HeadLover

I have a question if you may.

 In today digital area, we have no problems of storage, so putting a media in a "studio quality" like LINN has on their site (24/96 or even 24/192) isn't a problem.

 I mean, now days we have HD of even up to 2-4 tera byte, I guess that this number will be more like 16 Tera in some years from now.

 So even if a 70-80 minutes of music take up to 4GB of data, isn't a problem any more (FLAC with 24/96 of 70 minutes is something like 900MB)

 So, why not using 24/96 as the new standard ?!
 I mean, it can only bring good things to us, even if the different is less than 0.0001%, still why not ?

 I am sure that if now days someone will have the ability to plan his on RED CD, it will be like 24/96 or even more, why?
 Because we can


----------



## ILikeMusic

Quote:


 I mean, it can only bring good things to us, even if the different is less than 0.0001%, still why not ? 
 

Would it still be a good idea at 0.000001% difference or are you drawing a line?


----------



## ThePredator

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I have a question if you may.

 In today digital area, we have no problems of storage, so putting a media in a "studio quality" like LINN has on their site (24/96 or even 24/192) isn't a problem.

 I mean, now days we have HD of even up to 2-4 tera byte, I guess that this number will be more like 16 Tera in some years from now.

 So even if a 70-80 minutes of music take up to 4GB of data, isn't a problem any more (FLAC with 24/96 of 70 minutes is something like 900MB)

 So, why not using 24/96 as the new standard ?!
 I mean, it can only bring good things to us, even if the different is less than 0.0001%, still why not ?

 I am sure that if now days someone will have the ability to plan his on RED CD, it will be like 24/96 or even more, why?
 Because we can 
	

	
	
		
		

		
		
	


	


_

 


 Same reason we will never store video as raw data, there is no point. Not only are you increasing data requirements, you are increasing the time taken to move the data (across the internet, optical to magnetic, magnetic to magnetic, etc) and you are also increasing the processing time to play it back. It introduces even more complexity because most people don't have DACs that can do 24/96, much less 24/192.


----------



## gregorio

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_Thanks for clearing this up 
	

	
	
		
		

		
		
	


	


_

 

Yes, but the test conditions are quite strict otherwise a difference will not be detectable. Very low amplitude test tones need to be used rather than music and noise-shaped dither should not be used. Careful editing is also required otherwise editing clicks could cause damage (to equipment and/or ears).

  Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I have a question if you may.

 In today digital area, we have no problems of storage, so putting a media in a "studio quality" like LINN has on their site (24/96 or even 24/192) isn't a problem.

 I mean, now days we have HD of even up to 2-4 tera byte, I guess that this number will be more like 16 Tera in some years from now.

 So even if a 70-80 minutes of music take up to 4GB of data, isn't a problem any more (FLAC with 24/96 of 70 minutes is something like 900MB)

 So, why not using 24/96 as the new standard ?!
 I mean, it can only bring good things to us, even if the different is less than 0.0001%, still why not ?

 I am sure that if now days someone will have the ability to plan his on RED CD, it will be like 24/96 or even more, why?
 Because we can 
	

	
	
		
		

		
		
	


	


_

 

CD will never have hi-rez because the data rate is beyond the capabilities of the media. The only possible way around this would be quite strong data compression.

 The difference between 16bit and 24bit, if done properly, is 0%. So if you want to completely waste an extra third of your storage space for no reason whatsoever, it's your choice. The problem isn't only the storage space, it's the whole concept that many consumers think that 24bit makes a difference (however small) when it does not. Sooner or later, if it's not already happening, companies are going to start charging more for hi-rez than they do for 16bit, on the basis that 24bit is better quality and worth more than 16bit. The rip-off will be self re-inforcing as it's much easier to sell the idea that more data is better, than it is to sell the idea that it makes no difference. At that point, the truth of digital audio and hi-rez will be even worse off than $2,000 power cables because the majority of people will believe it makes a difference.

 It makes little difference to me, I've used 24bit for recording and production for well over a decade, it's not like I'm going to have to change the way I work to fullfill the demand.

 If consumers want to get the very highest digital audio quality format possible under any circumstances, currently (in theory) that would be 16bit 88.2kFs/s. But the difference between this and standard CD format has never been reliably differentiated by listeners. Instead, I would much rather see a consumer demand for high quality recording, production and mastering. If audiophiles spent as much time learning and understanding what makes a good recording, rather than trying to prove something exists beyond the science behind cables (for example), the quality of the music you listen to would improve dramatically. 

 There appears to me to be a great deal of snake oil in the audiophile world. Why spend so much time, effort and money on areas of digital audio which are going to make either no difference or differences which border on the impossible to detect, when you could be concentrating your efforts on areas which make the most dramatic of all differences. This to me seems like a much more logical avenue to persue because it would lead to an obvious quality improvement which everyone could benefit from rather than some miniscule difference which is undetectable by human beings.

 It's not like it would even be difficult, just start making note of the producer, studio and possibly engineer of those tracks which sound good quality to you. Then buy more music created by those people and/or that studio. Then you will create a demand for higher quality productions which will drive the industry to create better quality. It's no use blaming the record companies, the formats or the engineers, it's you the consumer who drive the market. If the demand is for low quality mp3s, why would a company spend extra money to make a higher quality product for less profit? So effectively, the marketplace seems to want the same old rubbish only in higher data formats?! Free downloads and digital file exchange is having a serious impact on the industry. It's simple economics, if there is less income available there is going to be less investment in the product. Make a real difference to audio quality, don't argue about ridiculous $1,000 power cables or 24/192 format digital, argue about the quality of the recording and demand and pay for better!

 Sorry for the rant but this message really needs to be driven home to the audiophile community, if you really are interested in high quality audio.

 G


----------



## HeadLover

Say, I have heard many times, even by some "Gurus" that doing music about the "Loudness War", and they say that when we all be moving to 24/96 it won't be a problem cause there will be much more "space" for the dynamic, so even if the sound it more compress, it still be good.

 What do you think about it? and that claim ?


----------



## Aleatoris

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Say, I have heard many times, even by some "Gurus" that doing music about the "Loudness War", and they say that when we all be moving to 24/96 it won't be a problem cause there will be much more "space" for the dynamic, so even if the sound it more compress, it still be good.

 What do you think about it? and that claim ?_

 

I think the whole point of the G-man's post was directed at things like the "loudness war". Why would a recording company artificially "louden" a record anyway? There as absolutely no point to it, as if I wanted loud music, I'd crank my volume knob. If there was no "loudening" we would not need to switch in the first place!


----------



## HeadLover

I am with you here !
 I hate this all LOUDNESS things, but like I have said, many claim that by going to 24/96 it will have less problems and no clipping and so on, is it true or what ?!

 And why all the "hype" over 24/96 if it doesn't matter ? and does it matter when it come to transfer a Vnyil to a PC music file ? (like FLAC)


----------



## Aleatoris

Yes, it would probably reduce clipping of (the bass frequencies of) "loudened" records. But again, why louden in the first place? 24/96 in this case would be a solution to a problem that shouldn't exist in the first place.

 There is a lot of hype over things that don't really matter
 eg. fast cars. I don't really care about fast cars, as I can't go over 110km/h on a highway. =/
 Okay, that was a bad example.


----------



## Clutz

fwiw, I don't know how going to 24/96 versus 16/44.1 is going to make any difference with respect to loudness. Loudness is compressing the dynamic range of the record so that it sounds good over the radio, iirc. I don't see how going to a format that has greater dynamic range is going to make any difference to this whatsoever. I mean, if they're already compressing the dynamic range below the theoretical maximum dynamic range of a CD already, what possible benefit can there be?


----------



## HeadLover

I am with you here dude, I HATE when my music it to loud from the record, I think now days almost every thing is bad at music when it comes to final production.

 But still, I just wonder if 24/96 will make it better/


----------



## Aleatoris

clutz, I don't know what you're referring to when you say compress. Maybe my terminology is bad.

Loudness war - Wikipedia, the free encyclopedia

 seems to me like the waveforms are just boosted.
 EDIT: just re-read wikipedia, you're right, I'm confused.


----------



## CDBacklash

clutz: I guess in an ideal world, people would feel less of a need to destroy their music to get them sounding loud because there is the potential for it to be louder already.
 Likely it wouldnt make much a difference because the source material we receive would still be crap.


----------



## scompton

Quote:


  Originally Posted by *CDBacklash* /img/forum/go_quote.gif 
_clutz: I guess in an ideal world, people would feel less of a need to destroy their music to get them sounding loud because there is the potential for it to be louder already.
 Likely it wouldnt make much a difference because the source material we receive would still be crap._

 

Why would anyone feel less of a need to compress dynamic range when they already compress it to 1/10th what redbook can handle?


----------



## Rempert

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Say, I have heard many times, even by some "Gurus" that doing music about the "Loudness War", and they say that when we all be moving to 24/96 it won't be a problem cause there will be much more "space" for the dynamic, so even if the sound it more compress, it still be good.

 What do you think about it? and that claim ?_

 

24/96 won't help with a "loud" production because that extra headroom will be an insane amount lower than the average volume of the music. The album would be just as compressed, any buzzy bass notes would still be there (much louder than the noise floor of the CD already), etc. These things aren't caused by the limitations of 44/16 audio. They are conscious choices by an engineer trying to please a client.

 The thing that really gets me is the loud remasters! It's kind of hard to believe it is the vision of the artist when the artist released the same album 30 years early and it was 80% softer with louder snare drums and less bass! But how is one supposed to decide which artist to listen to based on the quality of the production? It gets to the point where you just aren't buying music anymore, but the kiddies are still buying their itunes.


----------



## leeperry

http://patches.sonic.com/pdf/white-p..._dvd_audio.pdf











  Quote:


  Originally Posted by *scompton* /img/forum/go_quote.gif 
_Why would anyone feel less of a need to compress dynamic range when they already compress it to 1/10th what redbook can handle?_

 

coz each time you drop 6dB, you lose 1bit ?
 very low music would be 8bit 
	

	
	
		
		

		
		
	


	



 of course that's a worst case scenario, and no excuse for the loudness war.
  Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_why all the "hype" over 24/96 if it doesn't matter ? and does it matter when it come to transfer a Vnyil to a PC music file ? (like FLAC)_

 

the lower the resolution/sampling freq, the more audible the crackles...at least from my tests, don't shoot


----------



## scompton

I understand what you're saying, but my point was, if they're doing it with redbook, they'd still do it even if they have more potential than redbook.

 Edit: The charts you posted are an argument for 96 vs 44 not 24 vs 16 if I've read this thread correctly.


----------



## leeperry

all they say is that an higher fidelity can be achieved by increasing either the sampling freq, the resolution or both.

 this is a white paper on DVD-A from Sonic, not exactly your average marketing clowns...but what do they know..


----------



## s1rrah

Quote:


  Originally Posted by *mark_h* /img/forum/go_quote.gif 
_Very interesting thanks. I use R2R DACs or Vinyl so guess I'm safe...phew!_

 

Second that.

 R2R sounds right to my ear.

 Honestly, I tried to hang with the OP and some of his more vocal fellows-in-thread ... but they lost me very quickly.

 Bottom line and from a simpleton?

 A digital recording is PRODUCED (point of origin and what not) with only so much data inherent to the recording ...

 Any affectations to that orignal recording, oversampling, upsampling, etc. is just effect ... it's just an attempt to adjust or manipulate the originally recorded data ... whether that sounds better or not to one's ear is the same as whether the use of an EQ improves the sound to one's ear or not ...

 Myself? 

 I prefer to listen at the same bit depth and sample rate that the recording was made at. 

 Therefore, I use a R2R DAC design for my digital translation (MHDT Labs Havana) ... 

 Compared to listening through my soundcards DAC (Prelude) or my CD Player's DAC (Sony) ... the Havana is much more relaxed and natural and analogue sounding.

 ...


----------



## leeperry

Quote:


  Originally Posted by *s1rrah* /img/forum/go_quote.gif 
_oversampling, upsampling, etc.
 [...]
 I prefer to listen at the same bit depth and sample rate that the recording was made at._

 

agreed, from my experience they all add distortion and make the sound brighter 
	

	
	
		
		

		
		
	


	



 I fix my phones resonances w/ some EQ though, but in 32float and then dither down to 24...but still in 44.1Khz

 sinc upsampling is terribly bright...I don't quite understand why they enforce it in this EQ plugin :
Refined Audiometrics Laboratory


----------



## leeperry

Grover Washington-Winelight (DVD-Audio) (1980)

  Quote:


 This is the first 192/24 track I have had an opportunity to listen to, and it is gorgeous. Contrary to my expectations, all that extra sampling rate and bandwidth doesn't really make the audio sound any "brighter" (which kind of makes sense because my ears can't hear any higher than 16 kHz or so anyway!). It does however, make the sound seem "smoother" and the transients much better defined. Smooth and crisp - it's like having your cake and eating it! 
 

this disc is kinda cheesy, but "just the two of us" has never sounded better 
	

	
	
		
		

		
			





 like the man says, it sounds so natural & so smooth.....a far cry from the CDDA I can tell you that..

 I'm becoming so addicted to HD audio 
	

	
	
		
		

		
		
	


	




 (Depeche Mode's Violator in 24/96 5.1 lossless is simply out of this world), it makes CDDA sound agressive and mp3-like 
	

	
	
		
		

		
		
	


	




 for what it's worth I did an audiogram like 1 month ago, and the doc said my audition was as good as it gets


----------



## gregorio

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Say, I have heard many times, even by some "Gurus" that doing music about the "Loudness War", and they say that when we all be moving to 24/96 it won't be a problem cause there will be much more "space" for the dynamic, so even if the sound it more compress, it still be good.

 What do you think about it? and that claim ?_

 

I'm afraid you've got this the wrong way around. The loudness war is caused by over-compression or limiting. BTW, "compression" as an audio term is not related to data compression but to the reduction of the dynamic range. Limiting is just an extreme version of compression. Compression works by reducing the peaks in the waveform, allowing the overall level to be raised. In some genres of music the dynamic range is reduced to just a few decibels. As bit depth only encodes dynamic range, the more compressed, the fewer digital bits are required. So, if you've got a piece of music with quite a wide dynamic range, let's say 36dB, 6 bits are required to encode it. This means that on a CD you would be getting 6bits worth of music and 10bits worth of noise. With 24bit you are still only going to get 6bits of music but now you have 18bits of noise. In other words, using hi-rez 24/96 will not make the slightest difference whatsoever.

 As I mentioned in an earlier post, compression is an invaluable tool during production. It is the over use of compression which is the problem. This loudness war is not related to the format. In fact, if digital had not been invented and we all still used vinyl, we would likely still be the same position with the loudness war as we are now.

  Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_all they say is that an higher fidelity can be achieved by increasing either the sampling freq, the resolution or both._

 

This statement is true, up to a point. 44.1kFs/s is definitely better than 22kFs/s and 16bit is definitely better than 8bit. But once we get to 16bit 44.1kFs/s, we have reached the limits of analogue equipment to reproduce it and the limits of a human being to hear it. If you see anyone (or any company) claiming that 24/96 is definitely an improvement in quality or fidelity, that should set your alarm bells ringing that the company either does not understand how digital audio works or is deliberately trying to mislead you. As I mentioned before, there is no reliable proof that anyone can tell the difference between 44.1/16 and 24/96, anywhere near normal listening levels. So, at best, companies are making claims which they cannot prove.

 You have to realise, for audio equipment manufacturers and retailers, 24/96 provides the best marketing opportunity for years. It is not easy to continue to sell the same old specification equipment. Products that have been on the market for a while eventually get discounted and the profit margins are lower. It's much better and easier if you've got a new standard and therefore a whole new range of equipment you can sell at a higher price with a bigger profit margin. The fact that this new standard does not provide the slightest improvement is not really important. The retailers just want to sell more units for higher prices and if they can entice existing costomers (to the new standard) as well as new ones, how much better can it get? This cycle has pretty much been the trend over the whole history of consumer audio equipment. In some areas of consumer audio, escalating claims and prices has been going on for so long that the claims and cost of some equipment has completely lost touch with reality.

 I only really see this problem getting worse. Digital Audio Technology has already reached (and exceeded) the limits of human perception, so all that is left for the manufacturers and retailers is to continue developing products which exceed these limits by ever more ridiculous and superflous amounts. 

 G


----------



## mark_h

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_This loudness war is not related to the format._

 

???


----------



## gregorio

Compression was invented in the 1940's and was part of the basic equipment requirement for any studio by the 1960's. There are a few classic models of compressors from the 60's and 70's which are still in use today. Digital technology through the 80's and 90's saw the development of ever more powerful compressors. Today's digital compressors and limiters allow for a huge amount of compression, while the development of analogue compressors has stagnated. This loudness war is not a new thing, when I first got into recording in '92, the discussion among studio professionals regarding over compression was already in full swing, even though the majority of recordings were still mixed in analogue.

 The only way to get extra loundness is with the use of compression. More bits just allows for a larger dynamic range and nothing else. As compression reduces the dynamic range, so it makes absolutely no difference whether you're using 16bit, 24bit or 500bits. All you are going to get is more and more empty bits or bits containing only noise.

 G


----------



## leeperry

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_there is no reliable proof that anyone can tell the difference between 44.1/16 and 24/96, anywhere near normal listening levels._

 

well, it's like those so-called DBT cable tests, they are usually run on speakers in non-acoustically controlled rooms.

 I believe/know that you can hear far more details from hifi headphones(cd3k/dt770 600Ω/you name it) than from any speakers....you'll be able to enjoy the DVD-A improvement over CDDA by a long shot.


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_well, it's like those so-called DBT cable tests, they are usually run on speakers in non-acoustically controlled rooms.

 I believe/know that you can hear far more details from hifi headphones(cd3k/dt770 600Ω/you name it) than from any speakers....you'll be able to enjoy the CDDA/DVD-A improvement by a long shot._

 

There have been many DBTs, in studios, in laboratories and on consumer equipment. No reliable results have been obtained at normal hearing levels where either consumers, professionals or scientists could tell the difference.

 I think you may have mis-understood. Yes, you can of course hear more detail with a good set of cans than with speakers. However, the difference between 16 and 24bit is many orders of magnitude beyond what either your cans or speakers can reproduce, what you can hear or indeed, any recording on the market.

 As far as I can tell, the DT770 have a range of 96dB. This is not even enough to hear the full potential resolution of CD, let alone 24bit (144dB). I'm not questioning you are hearing an improvement with your DVD-A recordings, I'm just informing you that the improvement is not related to the higher bit depth. It is some improvement in mastering or production, not the increase in bit depth.

 G


----------



## nick_charles

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_well, it's like those so-called DBT cable tests, they are usually run on speakers in non-acoustically controlled rooms.

 I believe/know that you can hear far more details from hifi headphones(cd3k/dt770 600Ω/you name it) than from any speakers....you'll be able to enjoy the DVD-A improvement over CDDA by a long shot._

 

Maybe, but until I see it in a peer reviewed paper or at least in a well documented strict protocol with a skipload of test subects I will retain the right to remain dubious...call me old fashioned.


----------



## leeperry

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I'm not questioning you are hearing an improvement with your DVD-A recordings, I'm just informing you that the improvement is not related to the higher bit depth._

 

as you previously stated, it prolly has more to do w/ the increased sampling frequency...192KHz sounds clearer than 44.1 to my ears


----------



## HeadLover

Someone told me that Vinyl can't have the same "loudness war" as a cd, mean you can put sound very loud on it
 Is it true?
 Is the dynamic of a Vinyl is better than a CD or what? and if you compare a Vinyl, is it like 16/44 or 16/48 or 24/96 or what ??

 And why do people use 24/96 when transferring music from a Vinyl to a FLAC or what ever lossless audio format ?

 And btw, isn't 24/96 is better? I mean some brands claim that CD can't get better than like 90+ SNR (92 or something), while their amp and so on, can get even up to 110SNR, so we need a better format like 24/96


----------



## gregorio

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Someone told me that Vinyl can't have the same "loudness war" as a cd, mean you can put sound very loud on it
 Is it true?
 Is the dynamic of a Vinyl is better than a CD or what? and if you compare a Vinyl, is it like 16/44 or 16/48 or 24/96 or what ??

 And why do people use 24/96 when transferring music from a Vinyl to a FLAC or what ever lossless audio format ?

 And btw, isn't 24/96 is better? I mean some brands claim that CD can't get better than like 90+ SNR (92 or something), while their amp and so on, can get even up to 110SNR, so we need a better format like 24/96_

 

The reason 24bit is used when recording from vinyl is because of the additional headroom. However, the resultant 24bit recording could be converted to 16bit with no loss of quality, as of course you don't need headroom on playback.

 The theoretical dynamic range (SNR) of CD is 96dB, perceptually using noise-shaped dither this figure is increased to 120dB. Vinyl has a maximum dynamic range of 80dB. Also, 96dB (let alone 120dB) is way more dynamic range than any recording ever released (analogue or digital). This 120dB dynamic range is well beyond the capabilities of vinyl, probably at the limit of the best playback equipment and well beyond safe listening levels. The closest thing to vinyl would be 16bit 44.1kFs/s but the sound signatures of analogue and digital are so different that any direct comparison is very difficult.

 G


----------



## scompton

From what I've read, vintage vinyl was compressed to enable tighter spacing of the grooves. I also know of one anecdote of sudden dynamic changes causing the tone arm to jump. When the Telarc digital LP of the 1812 Overture came out, one of my coworkers bought it and he said that every time he played it the tone arm leapt out the track when the cannons went off.


----------



## leeperry

I tried converting my 24/96 vynil encodes to CDDA and the crackles became much more audible....it seemed pretty clear to me that the audio resolution downscale worsened the SQ 
	

	
	
		
		

		
		
	


	




 but again, it's prolly due to the double sampling frequency..


----------



## manaox2

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Someone told me that Vinyl can't have the same "loudness war" as a cd, mean you can put sound very loud on it
 Is it true?
 Is the dynamic of a Vinyl is better than a CD or what? and if you compare a Vinyl, is it like 16/44 or 16/48 or 24/96 or what ??_

 

This is just for clarification, I expect many people already understand the difference between analog and digital playback. Vinyl is analog on playback, meaning that your not taking ones and zeros to sample and that there are no bits being read, therefore you do not need a DAC. Because it does not use sampling, the sine wave comes naturally from the electrical difference created by the depth of the stylus needle instead of being reconstructed from bit sampling. I'm no expert, so that may be oversimplifying it. You would need an ADC to record what comes off of it and it would be a digital format before its recorded onto the vinyl, that is is where the 24/96 comes into play for the headroom as I understand it.


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_I tried converting my 24/96 vynil encodes to CDDA and the crackles became much more audible....it seemed pretty clear to me that the audio resolution downscale worsened the SQ 
	

	
	
		
		

		
		
	


	




 but again, it's prolly due to the double sampling frequency.._

 

It can't really be the higher sample rate either. The only option is that there is a fault or some very poor coding going on in the software of what ever you used to do the conversion.

 G


----------



## leeperry

well I was using Wavelab at that time, and I tried all the noiseshaping/resamplers available...now you tell me that this app is junk, but I believe it's the best Windows editor


----------



## Arjisme

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_well I was using Wavelab at that time, and I tried all the noiseshaping/resamplers available...now you tell me that this app is junk, but I believe it's the best Windows editor 
	

	
	
		
		

		
			



_

 

Hmm, I don't have Wavelab, but maybe someone else does? If so, maybe they would be willing to replicate what you did if you uploaded one of your files? It should be very straightforward to reconcile your assertion that quality decreased when you converted to 16 bit with gregorio's explanations that quality will not decrease (practically speaking) when doing that. Especially as you have provided a clear explanation of what the quality degradation sounds like.

 Can you post the exact steps you followed to convert to 16 bit? If you don't recall, can you repeat the process again and get the same result?


----------



## leeperry

well, it sounded like quantization....like if you downsample a CDDA to 12/13bit.

 sure, I still have the 24/96 untouched .APE files around, but do we really care that much...that's the true question 
	

	
	
		
		

		
		
	


	




 whatever rocks your boat you know, I love to rip vynil to 24/96 and listen to 5.1 24/96 MLP w/ a binaural stereo downmixing matrix....shoot me now


----------



## Arjisme

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_sure, I still have the 24/96 untouched .APE files around, but do we really care that much...that's the true question 
	

	
	
		
		

		
		
	


	


_

 

I think in this thread, we do. Gregorio has gone to great lengths to explain how 16 & 24 bit encoding impacts what we hear at the listening end. You've posted an experience that presumably refutes it. I'd say it's worth investigating.


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_well I was using Wavelab at that time, and I tried all the noiseshaping/resamplers available...now you tell me that this app is junk, but I believe it's the best Windows editor 
	

	
	
		
		

		
		
	


	


_

 

There are a number of better editors in Windows; Sonic Solutions, Nuendo and of course ProTools (plus one or two others). However, what you have described as happening, I would not have expected from Wavelab as I don't believe it's junk. There are so many reasons why this could have happened, from a dodgy installation, to some sort of incompatibility, a particularly jittery ADC process or it's not inconceivable that there's a bug in the program, although I'd have thought this would have been identified and fixed pretty quickly.

 G


----------



## endless402

hmm didnt read all of this thread before so this may have been covered:


 in the end, the music sounds the best in the format in which it was recorded in

 aka take a listen to kent poon's free tracks. 24/96 sounds better than the 16/44.1 version most likely due to the conversion process


----------



## gregorio

Quote:


  Originally Posted by *endless402* /img/forum/go_quote.gif 
_in the end, the music sounds the best in the format in which it was recorded in ... take a listen to kent poon's free tracks. 24/96 sounds better than the 16/44.1 version most likely due to the conversion process_

 

In some respects these two statements encapsulate the problem. The first statement has been true for the vast majority of the history of recording. Even until the early 1990s the recording format was different to the consumer format. For years consumers were buying vinyl but the recording format was usually 2" tape, which was never practical for the consumer. Around the middle to late 90s most studios started switching over to 24bit as the recording format.

 Now, for just about the first time, consumers have direct access to the professional recording format. Unfortunately, it's too late! It could have made a difference 15 years ago and earlier but today, the technology has developed beyond the ability of anyone to hear the difference or of any analogue equipment to reproduce it. We are well past the point where higher specification means better sound. As time goes on improvements will of course be made but the improvements will be in the processing and processes rather than the underlying digital specification.

 So your second statement of something sounding better and my point above about the current specifications being beyond what can make something sound better.

 RE the conversion causing the problem: To be honest, with professional conversion there should not be any noticable artefacts. In other words, it's unlikely to be the conversion process itself causing a perceived lower quality and more likely to be the person doing it. If you can, have a look at the peak levels on the 16bit version. If the peak level is higher than -3dBFS, it's possible that the reconstruction filter or any EQ added by the consumer could cause clipping. Secondly, it's not at all uncommon for inexperienced producers and engineers to maximise signal levels during mixing. Add just a tiny bit more limiting when converting to 16bit and distortion can so easily occur. This is even more likely to be missed if the focus is on a 24bit release. There are in fact quite a number of ways that two different versions of the same master can vary. None of these potential problems are directly related to the digital audio format though.

 G


----------



## endless402

why dont u go listen to it yourself. he's a pretty famous sound engineer

Design w Sound » AJP3 - Free Hi-Res Samples

 might also be the fact that my dacmagic upsamples 96 to 192 much more cleanly than 44.1 to 192 since it's a perfect 2:1


----------



## Acix

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_1. As far as I'm concerned, I am willing to admit that it may be possible to perceive some sonic content above 20kHz, although there is only anecdotal evidence for it. However, if there is anything to be heard it is definitely lower than 40kHz, simply because there are no studio mics that can pick up anything above this point. A 40kHz signal can be encoded using a sample rate of 96kFs/s. There is however absolutely no advantage to a sample rate of 192kHz as there simply is nothing between 48kHz and 96kHz which can be recorded. If someone thinks they hear a difference with 192kFs/s, it cannot be anything to do with the recording, because no studio mics can record above 40kHz, so the only thing which can be in these higher frequencies on a recording is noise.

 2. Sorry but you are way off the mark with your second paragraph. I have been recording and mixing exclusively in higher than 16bit since the end of 1992, not long after the technology was first available (Yamaha DMR8 + DRU8). I switched over from 20bit to 24bit in about 1995, when multi-channel 24bit converters first became available (DigiDesign 888). So, there can't be that many engineers who have a longer practical experience than me in working with >16bit. Also, I've used higher resolution recording technology not just for music recording and production but also quite extensively for film and TV sound too. My understanding of the theory side of HR has come from a fair bit of research over the years and particular thanks need to go to Nika Aldrich who gave many hours of his time online to help iron out many of my misunderstandings.

 In fact, there is not much in the digital audio chain that I haven't thoroughly tested. Take dither for example, to start with only TDPF (Triangular Probability Density Functions) were available but I've used extensively Sony Super-Bitmapping, UV22, POWr, Waves L2 and DigiDesign. I've gone through or tested countless mics, mic-pres, cables, ADCs, DACs and speakers. Since the early '90s I must have spent around $500,000 on equipment and acoustics. I've also done work in many of this country's (UK) top studios and dubbing theatres as well as my own of course.

 G_

 



 It's really simple, if you're not able to hear information between 20 and 22kHz, so your approach to sound if questionable on the whole topic. Now, as I already mentioned, proper studio equipment is needed to detect sound above these levels. Speakers, headphones and the whole signal chain needs to be able to support 24/96. This is just the beginning to work in higher resolution. I've followed you throughout the thread and it seems to me that you might a have a misunderstanding in your concept of HR sound. It's not even about if you can hear above 22kHz. Let's take your case as an example: You don't hear between 20 and 22kHz. You're still going to benefit from the headroom of high resolution that allows more room for peak level... before converting to 44.1. 
 And this is actually the benefit of recording and working in high resolution, as I said before.


----------



## Eric M

Sorry if this was asked before... I know there's no difference noticeable to humans between 16bit and 24bit, but what about frequencies.

 Does 16/96 sound better than 16/48?


----------



## JaZZ

Quote:


  Originally Posted by *Eric M* /img/forum/go_quote.gif 
_Sorry if this was asked before... I know there's no difference noticeable to humans between 16bit and 24bit..._

 

That's still not written in stone (despite the thread title), although personally I tend to give this theory some credit. 


  Quote:


 _...but what about frequencies?

 Does 16/96 sound better than 16/48?_ 
 

A clear yes from me and my ears.
.


----------



## Acix

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_There are a number of better editors in Windows; Sonic Solutions, Nuendo and of course ProTools (plus one or two others). However, what you have described as happening, I would not have expected from Wavelab as I don't believe it's junk. There are so many reasons why this could have happened, from a dodgy installation, to some sort of incompatibility, a particularly jittery ADC process or it's not inconceivable that there's a bug in the program, although I'd have thought this would have been identified and fixed pretty quickly.

 G_

 

LOL...Wavelab Nuendo and Cubase are from the same company, Steinberg.


----------



## Clutz

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_LOL...Wavelab Nuendo and Cubase are from the same company, Steinberg._

 

How does that in anyway contradict what he wrote?


----------



## Clutz

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_ Speakers, headphones and the whole signal chain needs to be able to support 24/96._

 

That doesn't make any sense whatsoever.


----------



## Acix

Quote:


  Originally Posted by *Clutz* /img/forum/go_quote.gif 
_How does that in anyway contradict what he wrote?_

 

Does Wavelab sound better than Nuendo?


----------



## Clutz

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Does Wavelab sound better than Nuendo?_

 

How would I know, I'm not a sound engineer- BUT, he didn't claim that he'd expect them to sound any different. You're making an inference that he never implied. But even so, so what? Since they're different products offered by the same company it stands to reason that they differ from each other in some way. You can by a Corvette or a Solstice- they're both made by GM, but they're very different products. 

 Your readiness to make ridiculous arguments causes me to question *everything* you write.


----------



## gregorio

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_It's really simple, if you're not able to hear information between 20 and 22kHz, so your approach to sound if questionable on the whole topic._

 

So in fact, the only people who are not questionable are babies and small children? And you've ruled out pretty much every audio professional in the world.

  Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Now, as I already mentioned, proper studio equipment is needed to detect sound above these levels. Speakers, headphones and the whole signal chain needs to be able to support 24/96. This is just the beginning to work in higher resolution._

 

And as I have already mentioned, there is no equipment on the market which can resolve 24bits, not speakers, headphones, ADCs or DACs. So what are you saying, that no one should be using hi-res audio for recording?

  Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_I've followed you throughout the thread and it seems to me that you might a have a misunderstanding in your concept of HR sound. It's not even about if you can hear above 22kHz. Let's take your case as an example: You don't hear between 20 and 22kHz. You're still going to benefit from the headroom of high resolution that allows more room for peak level... before converting to 44.1. 
 And this is actually the benefit of recording and working in high resolution, as I said before._

 

It's you who mis-understand, the sampling rate has nothing to do with headroom. Headroom is a function of amplitude and the range of amplitudes is determined by bit depth. If you had actually read this thread you would have seen on a number of my posts, including the OP, where I stated the benefits of 24bit for recording.

 G


----------



## Aleatoris

Hey G, I think the thread's gotten to the point where people are going:

 "Well, I didn't read the rest of the thread because it's too long, but IMO, you're a big fat liar"


----------



## gregorio

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Does Wavelab sound better than Nuendo?_

 

That question is impossible to answer. Both Wavelab and Nuendo are just software packages. The required hardware (ADC, DAC) is independant from the program. Also, much of the actual processing is done by plugin processors, so the sound quality varies according to which plugins are installed/used. More importantly is the user of the software.

 All these packages are just tools. It's a bit like asking which make of chisel will make a better chair. Ultimately, the chisel is irrelevant, it's the cabinet maker who makes the difference.

 G


----------



## Clutz

Quote:


  Originally Posted by *Aleatoris* /img/forum/go_quote.gif 
_Hey G, I think the thread's gotten to the point where people are going:

 "Well, I didn't read the rest of the thread, because it's too long, but IMO, you're a big fat liar"_

 

I think that's a little bit more generous to some of the detractors than is fair- people were calling him "a big fat liar" from the start. I am very impressed with gregorio's composure through this.


----------



## manaox2

Well, I read through the entire thread. So far my conclusion is that there is no reason not to use 24/192 if you have it and think it sounds best, so I don't need to worry whether its 16 bit or 24 bit and really just worry about stuff like how my digital filters react on the DAC and how it sounds. No need to even worry about specifics when it takes testing to know for sure.

 For example, The WM8740 OPUS DAC by TwistedPearAudio is agreed to be very good at 24/96 instead of 24/192 with filter #2 selected. The same DAC with the WM8741 chip on the board is agreed by many to be better with 24/192 then 24/96. So, in conclusion, its the equipment that matters more then the bit rate of the digital input for me as long as its 16/44.1 at least. Potential and performance may not always match when it comes down to engineering. With the spread of upsampling in todays audio gears, upsampling or oversampling does seem to possibly hurt in some situations, but it doesn't really hurt to have the option does it?


----------



## gregorio

Quote:


  Originally Posted by *manaox2* /img/forum/go_quote.gif 
_Well, I read through the entire thread. So far my conclusion is that there is no reason not to use 24/192 if you have it and think it sounds best, so I don't need to worry whether its 16 bit or 24 bit and really just worry about stuff like how my digital filters react on the DAC and how it sounds. No need to even worry about specifics when it takes testing to know for sure.

 For example, The WM8740 OPUS DAC by TwistedPearAudio is agreed to be very good at 24/96 instead of 24/192 with filter #2 selected. The same DAC with the WM8741 chip on the board is agreed by many to be better with 24/192 then 24/96. So, in conclusion, its the equipment that matters more then the bit rate of the digital input for me as long as its 16/44.1 at least. Potential and performance may not always match when it comes down to engineering. With the spread of upsampling in todays audio gears, upsampling or oversampling does seem to possibly hurt in some situations, but it doesn't really hurt to have the option does it?_

 

The above is good advice. 24bit cannot sound better than 16bit but the sample rate does vary depending on the filters in the DAC as you mentioned in your message above. Of course, 24bit handled properly should not sound worse than 16bit but not better either. As a general rule, a sample rate of 192kFs/s should sound worse than a sample rate of 96kFs/s, as there is nothing else to capture and it's likely to cause additional errors as the system is required to transfer and process twice the amount of data in the same time. However, it may be that a DAC manufacturer may decide to put more time and effort into the filters at 192k and have very poor filters at 96kFs/s. It is possible that under these conditions 192k could sound better. Properly implimented though, there should be no noticeable improvement between 44.1k and 96 or 192. This is the theory, in practice as I have mentioned, there have been no conclusive DBT where anyone could reliably tell the difference.

 Ideally I would reccommend 16bit 88.2kFs/s but I realise that few if any tracks are available to consumers in this format. The other potential disadvantage of 24/192 (beyond the processing requirement) is it's storage requirement, which is about 6 times greater than the same duration of 16bit 44.1k.

 G


----------



## leeperry

Quote:


  Originally Posted by *manaox2* /img/forum/go_quote.gif 
_upsampling or oversampling does seem to possibly hurt_

 

yes, resampling creates additional THD/aliasing and makes the sound brighter...whether you like your sound brighter/more distorted(more analog-like? 
	

	
	
		
		

		
			





) is a matter of taste.

 anyway, I've run a few tests...24/96 doesn't seem to sound better than 16/44.1 because of the increased bit depth, but more because of the double sampling freq


----------



## Acix

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_So in fact, the only people who are not questionable are babies and small children? And you've ruled out pretty much every audio professional in the world.
 G_

 

Since I know you're probably not a small child or a baby, I'm going to treat you as an adult...so it would be delightful if you could answer the question directly.

 Can you hear above 20 kHz, or not? 
	

	
	
		
		

		
		
	


	




 BTW: The Q10 and the Renaissance REQ are up to 21.357 HZ


----------



## clasam

Quote:


  Originally Posted by *Aleatoris* /img/forum/go_quote.gif 
_I think the whole point of the G-man's post was directed at things like the "loudness war". Why would a recording company artificially "louden" a record anyway? There as absolutely no point to it, as if I wanted loud music, I'd crank my volume knob. If there was no "loudening" we would not need to switch in the first place!_

 

I'm not sure if this was addressed, since I haven't bothered to read through all 14 pages...

 From the reading I've done on Loudness...the human ear/brain perceives louder signals as sounding better (as long as the sound produced isn't cacophonous, that is). 

 Therefore, starting with Oasis' What's the story/Morning Glory album, mastering engineers were told to keep uping the volume to 1)take advantage of this psychoacoustic effect 2)keep up with other records that were doing the same.


----------



## Clutz

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Can you hear above 20 kHz, or not? 
	

	
	
		
		

		
		
	


	


_

 

I think it's really funny that you go around questioning other peoples knowledge of high end audio when you didn't understand that LOSSLESS audio codecs were able to compress audio files without any loss of data- e.g. they are as good as the original .WAV files.

 Second, using a 
	

	
	
		
		

		
		
	


	




 in the above context is really obnoxious.


----------



## Acix

Quote:


  Originally Posted by *Clutz* /img/forum/go_quote.gif 
_using a 
	

	
	
		
		

		
		
	


	




 in the above context is really obnoxious._

 

Just let the man answer the question.


----------



## gregorio

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Since I know you're probably not a small child or a baby, I'm going to treat you as an adult...so it would be delightful if you could answer the question directly.

 Can you hear above 20 kHz, or not? 
	

	
	
		
		

		
		
	


	




 BTW: The Q10 and the Renaissance REQ are up to 21.357 HZ_

 

No, I can't, because I'm not a small child or a baby!

 I don't understand your point, virtually no adults can hear above 20kHz. I run this test with my students every year and the highest anyone has reliably heard was 19.5kHz, most are around 17kHz. My hearing, which is very good for my age, went up to 16kHz the last time I checked. I don't know of any audio professionals who can hear beyond 20kHz, just like I don't know of any equipment which can reproduce 24bits of dynamic range.

 Providing you are not a child, you cannot hear beyond 20kHz either. I take it you do have a basic understanding of how the ear works and that it's not just a case of your ears stopping at a certain frequency but that there is a gradual roll-off from about 12kHz depending on age?

 G


----------



## manaox2

I do not like the trend I see of people attacking another person's hearing when they can no longer argue using logic. He's not even posting his opinions really on what he hears, its based on mathematical principle and theory. An argument against his hearing is completely irrelevant, worse then saying Beethoven shouldn't have composed music because he suffered hearing loss.

 FTR, I can not hear much if anything over 17Khz at age 23.


----------



## Arjisme

Quote:


  Originally Posted by *manaox2* /img/forum/go_quote.gif 
_I do not like the trend I see of people attacking another person's hearing when they can no longer argue using logic. He's not even posting his opinions really on what he hears, its based on mathematical principle and theory. An argument against his hearing is completely irrelevant, worse then saying Beethoven shouldn't have composed music because he suffered hearing loss._

 

I have to agree. It amounts to an ad hominem attack against his logic, which is a well known fallacy.


----------



## digger945

Quote:


  Originally Posted by *manaox2* /img/forum/go_quote.gif 
_FTR, I can not hear much if anything over 17Khz at age 23._

 

I have a hard time hearing 14kHz on good days at age 42.


----------



## saintalfonzo

I just went and tested my hearing. I'm very surprised to say I can hear right up to 22.5 Khz ( as high as the tests I tried went ) although I have some minor hearing loss in my right ear, which I never realized until I started using bluetooth with my cell phone and switched ears. I'll be 30 this august, and I've played guitar for 15 years as well as attending over 100 concerts. I used to listen to headphones at an unhealthy volume but now I enjoy music at safer levels with the acception of some indoor concerts/shows.


----------



## Arjisme

Quote:


  Originally Posted by *saintalfonzo* /img/forum/go_quote.gif 
_I just went and tested my hearing. I'm very surprised to say I can hear right up to 22.5 Khz._

 

That's interesting. Did you test your hearing yourself or did you have an audiologist test you?


----------



## saintalfonzo

Quote:


  Originally Posted by *Arjisme* /img/forum/go_quote.gif 
_That's interesting. Did you test your hearing yourself or did you have an audiologist test you?_

 

I took 5 different tests online. I don't really think it's necessary to pay an audiologist to find out what frequencies I can hear, although doing a complete examination with an audiologist would of course yield more in-depth results. I'm not at all sensitive to such high frequencies, but I can hear the tone.


----------



## Ham Sandwich

Quote:


  Originally Posted by *saintalfonzo* /img/forum/go_quote.gif 
_I took 5 different tests online. I don't really think it's necessary to pay an audiologist to find out what frequencies I can hear, although doing a complete examination with an audiologist would of course yield more in-depth results. I'm not at all sensitive to such high frequencies, but I can hear the tone._

 

At what sampling rate did you play the tones at? You're probably hearing artifacts and not the actual tone.


----------



## Arjisme

There could be other variables at play too. I'm not an expert at hearing tests, but it was my understanding that there is a procedure to follow to set the volume of the tones. And, yeah, there could be artifacts that are being heard.

 Not trying to say you can't hear up to 22.5K, but it is so unlikely given your age and the loud volumes you used to (and still do at times) listen at that it's reasonable to explore whether there is some other explanation for your conclusion you can really hear tones at that frequency.


----------



## ILikeMusic

Quote:


  Originally Posted by *saintalfonzo* /img/forum/go_quote.gif 
_I took 5 different tests online._

 

If your audio card is subject to aliasing (as many/most are) then you could try the test 50 times and the result would be the same. In any event it is unlikely in the extreme that you can hear 22.5 kHz... I would try using some different playback hardware to see if you get the same results.


----------



## saintalfonzo

I'm not using my audio card, I'm using spdif out and the rig in my sig. I would say the tones I could hear were obvious until 21Khz, and I had to jack the volume beyond there. It shouldn't really matter what volume a frequency is played as you can hear it or you can't, but I'm not doubting other things could be affecting the result. I know I can hear better than average at as I've never met a dog whistle I couldn't hear, and I'm often annoyed by high freq given off by electronics ect when other people are unaware. My little sister has those annoying "mosquito" ringtones that she uses at school so teachers can't hear her phone ring, and I can hear those also. Another semi-unrelated thing I've noticed is that I'm always aware of what song/artist is playing in the background in public places when other people I'm with aren't even aware there is music playing (such as over a PA in a store/mall) at all. And no, I'm not schizophrenic
	

	
	
		
		

		
			





. Anyway, if one of you guys wants to send me all the rules I need to abide by to play the frequency game I'll give it a try.


----------



## gregorio

Quote:


  Originally Posted by *saintalfonzo* /img/forum/go_quote.gif 
_I'm not using my audio card, I'm using spdif out and the rig in my sig. I would say the tones I could hear were obvious until 21Khz, and I had to jack the volume beyond there. It shouldn't really matter what volume a frequency is played as you can hear it or you can't, but I'm not doubting other things could be affecting the result._

 

I'm not going to say that it is impossible for you to be able to hear 22kHz at thirty years of age but at a guess I would say that if you really can hear 22kHz you are a member of a tiny group of people on the planet.

 BTW, it will make a difference the volume that the frequency is played back at. Human hearing is not linear with a sudden cut off at a particular frequency. Human hearing rolls-off gradually from around 12kHz, meaning the higher the frequency beyond about 12kHz, the quieter it will appear. The same is true for low frequencies below about 200Hz. Therefore, the louder the volume, the more likely you are to hear higher frequencies. An Audiologist will run tests at fixed levels to avoid this issue.

 A word of warning though, a colleague of mine was running a frequency test with a class of degree students. He was certain that some of the teenagers would be able to hear 18kHz so he kept gradually turning up the volume. Eventually one of the students was able to perceive it, when he saw smoke coming out of the tweeters! On closer inspection, the tweeters on the Blue Sky monitors had completely melted!!

 G


----------



## Jensen

Gregorio, I know this thread is focused on bit depth, and on that subject when I've listened to the same recording at 16 vs 24 bit depth I could hear no difference and I tend to agree with the presentation of your arguments.

 However, that same recording (which is 24/96) that I got off of hdtracks.net does sound remarkably better than anything I've ever heard off of a redbook CD before. I'm sure it mostly has to do with the fact that it is just a very well done mastering. Now I've been reading through the thread and I'd like to hear your thoughts a bit more defined on the effects of higher sample rates and their ability to affect the quality of the playback/recording.

 It just seems to me, from a purely laymen point of view, that if you take an analog sine wave and sample it 96 thousand times a second vs 44 thousand times a second the end result is going to be a smoother, more accurate, digital reprsentation of that sine wave. The thing that struck me most about these high quality recordings was how smooth everything sounded. Especially things like the minute vibrations and decay of a cymbal shimmering after a drum stick hit it. The sound tapered off in an extremely realistic way that I've never heard in a recording before. Also, the overtones of stringed instruments in classical music seemed to sound much more pronounced and accurate. 

 These types of things seem like they might benefit from having higher sample rates, reproducing the analog wave more accurately. So, bit depth aside, are high quality recordings worth it for the sample rate increase alone? Or are all these DVD-A/SACD/DSD recordings just well mastered music that could have been put on a CD and no one would be the wiser (assuming stereo, I know multichannel stuff wouldn't fit)?


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## manaox2

Smoother waves, but doesn't resampling it add distortion and maybe even phase differences? The resampling schemes alone vary so greatly and noticeably sound different to me.


----------



## gregorio

Quote:


  Originally Posted by *Jensen* /img/forum/go_quote.gif 
_...However, that same recording (which is 24/96) that I got off of hdtracks.net does sound remarkably better than anything I've ever heard off of a redbook CD before. 

 It just seems to me, from a purely laymen point of view, that if you take an analog sine wave and sample it 96 thousand times a second vs 44 thousand times a second the end result is going to be a smoother, more accurate, digital reprsentation of that sine wave. The thing that struck me most about these high quality recordings was how smooth everything sounded. Especially things like the minute vibrations and decay of a cymbal shimmering after a drum stick hit it. The sound tapered off in an extremely realistic way that I've never heard in a recording before. Also, the overtones of stringed instruments in classical music seemed to sound much more pronounced and accurate._

 

As far as digital audio theory is concerned there is a problem with the first statement in second paragraph I've quoted. If you look at a graphical representation of the digital data in a digital audio software package, waveforms recorded with higher sample rates (and bit depths) it will appear to look smoother and more like an analogue waveform on the screen. This is useful when editing digital audio but is rather mis-leading unless you fully appreciate that you are looking at a graphical representation of the digital audio data rather than what will actually come out of a DAC once that digital audio data is converted back to analogue. The "rule" of digital audio theory is that providing the sample rate is at least double the highest audio frequency then the waveform can be reproduced perfectly. This means that recording say a 12kHz sine wave at 96kFs/s or even 192kFs/s is not going to reproduce the 12kHz sine wave any more perfectly than the already perfect 44.1kFs/s recording.

 I still haven't had chance to recalibrate my work system to actually analyse the linked 24/96 track. However, I don't doubt that those of you who heard a difference actually did hear a difference. I too have heard differences between some recordings released in both 24/96 and CD format. Unfortunately there are many potential reasons for this.

 The first thing I would say is that if you can actually hear more pronounced overtones and smoothness to cymbal decays (for example) then by definition, what you are hearing is within the audible frequency spectrum and therefore just as capable of perfect reproduction at 44.1kFs/s as at 96kFs/s. However, there maybe other factors at play, here is an example. Let's say that a particular cymbal is very exposed in a track and that it has a sonic characteristic which includes an obvious peak at around 16kHz. During the final phase of mastering or mix down noise-shaped dither is applied and it just so happens that the algorithm used redistributes the dither noise centred around 16kHz. It's entirely possible that this dither noise could to an extent mask the lovely cymbal decay and be easily missed by the mastering engineer, especially if it is played back on audiophile headphones which often enhance the higher frequencies. Another possibility is that the reconstruction filters in a particular DAC could be causing more noticable artefacts at 44.1kFs/s than at 96kFs/s. Although this effect is very small and most likely to be noticed with cheaper convertion processors. Mostly though I would expect differences between versions to be a direct result of some mastering process.

 It should also be noted that I have noticed that it's not uncommon for 16bit versions of commercially released hi-res recordings to be either deliberately of lower quality or end up being of lower quality because no one puts any real effort into making it sound as good. If you think about it, a record company (and consumers) wouldn't be too happy if they sold a recording at a premium because it's in a Hi-rez format if the 16bit version had the same or better sound quality. The whole industry, from record labels to consumer equipment manufacturers is poised to take advantage of the hi-res formats and the equipment capable of handling it. The whole marketing edifice would fall flat on it's face if consumers actually knew that hi-res was no better than good old CD.

 I might have a little time tomorrow to download the file linked in an earlier post and have a listen for myself. I might even have a go at creating a 16bit version of my own for comparison by head-fiers.

 G


----------



## Jensen

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_The "rule" of digital audio theory is that providing the sample rate is at least double the highest audio frequency then the waveform can be reproduced perfectly. This means that recording say a 12kHz sine wave at 96kFs/s or even 192kFs/s is not going to reproduce the 12kHz sine wave any more perfectly than the already perfect 44.1kFs/s recording.
_

 

Just as an FYI, the 24/96 tracks that I downloaded and am referring to were from hdtracks.net, which is offering a free "sample download" of some of the stuff they are selling. I have not listened to the linked music earlier in the thread.

 I should also note that I wasn't comparing these to 16/44 versions of the same music, of which I don't have access to. I was just remarking at how amazing the reproduced audio sounded in general.

 Now, as to the part of your post that I quoted. I do understand what you're saying regarding the sample rate having to be double the highest frequency for a reproduction of the sound to be possible, but I'm still not seeing how that is related to my question about the QUALITY of the reproduced sound. I'm sure it is and I hope you'll help me understand.

 In my mind I picture an analog sine wave on a graph. The X axis of the graph represents time. If I'm trying to digitally do a "connect the dots" game on that sine wave, it would obviously seem beneficial to have more dots. The more dots I have the smoother my wave form is going to be. This here is the crux of my question. Frequency response aside, which I understand a 44Fs/s sample rate allows me to go up to 22khz in audio, wouldn't the smoother wave form of a "hi res" recording still be a more accurate reproduction of the sound than the less smooth (44) version?


----------



## Publius

Quote:


  Originally Posted by *Jensen* /img/forum/go_quote.gif 
_Now, as to the part of your post that I quoted. I do understand what you're saying regarding the sample rate having to be double the highest frequency for a reproduction of the sound to be possible, but I'm still not seeing how that is related to my question about the QUALITY of the reproduced sound. I'm sure it is and I hope you'll help me understand.

 In my mind I picture an analog sine wave on a graph. The X axis of the graph represents time. If I'm trying to digitally do a "connect the dots" game on that sine wave, it would obviously seem beneficial to have more dots. The more dots I have the smoother my wave form is going to be. This here is the crux of my question. Frequency response aside, which I understand a 44Fs/s sample rate allows me to go up to 22khz in audio, wouldn't the smoother wave form of a "hi res" recording still be a more accurate reproduction of the sound than the less smooth (44) version?_

 

Your error is that in your mind you're "connecting the dots" with straight lines. The correct lines are bandlimited interpolations, which are how real oversampling DACs work. When you apply bandlimited interpolation between the points, you will find that adding more points by increasing the sample rate results in no additional quality gain. The new points will fall exactly on the line that you originally drew. Therefore, no additional accuracy.

 This link has been dug up thousands of times before, but it looks like you haven't seen this before, so:

http://www.lavryengineering.com/docu...ing_Theory.pdf


----------



## Acix

gregorio, do you hear the differences between 32 bit and 16 bit?


----------



## deaconblues

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_gregorio, do you hear the differences between 32 bit and 16 bit?_

 

/facepalm

 If it's already extremely difficult to tell the difference between the extremely low noise floors of 16 bit and 24 bit, why would the even lower noise floor of 32 bit be any easier to discern?


----------



## gregorio

Quote:


  Originally Posted by *Jensen* /img/forum/go_quote.gif 
_Now, as to the part of your post that I quoted. I do understand what you're saying regarding the sample rate having to be double the highest frequency for a reproduction of the sound to be possible, but I'm still not seeing how that is related to my question about the QUALITY of the reproduced sound. I'm sure it is and I hope you'll help me understand._

 

You've slightly misunderstood. If the sample rate is at least double the audio frequency, the theory states that the waveform can be reproduced perfectly. The important word here is "perfectly", not as you have stated above, "possible". The term "Perfectly" is used here in it's literal, superlative sense and means in this context a completely linear output, IE., with no distortions or artefacts of any kind and containing 100% of the original waveform and ALL it's detail. Using higher sample rates cannot make the reproduced waveform any more detailed or higher quality or more perfect.

 If you have a quick read of my last post again, perhaps it will make more sense now you know what I mean by "perfectly".

 What Publius said in his reply to you was correct.

  Quote:


  Originally Posted by *manaox2* /img/forum/go_quote.gif 
_Smoother waves, but doesn't resampling it add distortion and maybe even phase differences? The resampling schemes alone vary so greatly and noticeably sound different to me._

 

It's possible that resampling may cause distortions but generally speaking I wouldn't expect there to be anything too noticeable. Phase differences are very unlikely to be introduced. To hear the effects of phase differences usually requires differences in the milliseconds range (1/1000 sec or more). A DAC should easily be sample accurate at 44.1kFs/s. The duration of a sample at this rate is 1/44,100 sec, far more accurate than the few milliseconds required. If you can hear some phase issues when oversampling there is something fairly seriously wrong with the DAC.

  Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_gregorio, do you hear the differences between 32 bit and 16 bit?_

 

You would need to be a bit more specific. If you mean during playback of a mixed master at normal hearing levels, then no I hear no differences. If you mean while mixing a channel of audio with a poor SNR and substantially boosting, say by 60dB or more (1,000+ times), so it will balance when other channels are added to the mix, then yes I have heard a difference. This is of course the whole point of higher than 16bit for recording. Be careful not to get hung up on 32bit. There are some difficulties when using 32bit due to the way an audio software program has to deal with exactly where the decimal point is. In other words a direct comparison between 32bit and 16bit may or may not provide the expected additional 16bits of dynamic range. This is in theory of course, in practice, as previously mentioned, the practicalities of electronic circuits make the full dynamic range unobtainable anyway. If you are working professionally, 24bit in practice probably provides slightly more dynamic range (through a lower noise floor) than a 32bit system. 

 G


----------



## Jensen

So basically, 16/44 is fine for music and there is no reason to go any higher (for playback).


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## gregorio

Quote:


  Originally Posted by *Jensen* /img/forum/go_quote.gif 
_So basically, 16/44 is fine for music and there is no reason to go any higher (for playback)._

 

The answer to that is a qualified yes! By this I mean that releases in 24/96 sometimes sound better than at 16/44.1. Not because there is any real advantage to the format but simply because more effort is often put into the mixing and mastering process. To get round this what you could do is own a 16/44.1 DAC, download 24/96 recordings and convert them to 16/44.1 for playback.

 G


----------



## ILikeMusic

I'd like to thank everyone for the excellent and informative technical content of this thread. I'm sure we're learning a lot and appreciate your time in posting.

 Since we have some experts here I have a quick question that I've always wondered about... since the source audio is filtered to remove any high-frequency content above the Nyquist limit before the AD converter during the recording process, why are anti-aliasing filters required during playback? Are there artifacts that are generated by the DA playback conversion itself, even if the high frequencies are properly filtered during the recording process?


----------



## Acix

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_You would need to be a bit more specific. If you mean during playback of a mixed master at normal hearing levels, then no I hear no differences. If you mean while mixing a channel of audio with a poor SNR and substantially boosting, say by 60dB or more (1,000+ times), so it will balance when other channels are added to the mix, then yes I have heard a difference. This is of course the whole point of higher than 16bit for recording. Be careful not to get hung up on 32bit. There are some difficulties when using 32bit due to the way an audio software program has to deal with exactly where the decimal point is. In other words a direct comparison between 32bit and 16bit may or may not provide the expected additional 16bits of dynamic range. This is in theory of course, in practice, as previously mentioned, the practicalities of electronic circuits make the full dynamic range unobtainable anyway. If you are working professionally, 24bit in practice probably provides slightly more dynamic range (through a lower noise floor) than a 32bit system. 

 G_

 



 I use 32 bit for mixing editing and mastering. The 32 bit keeps the original noise floor down and allows me to apply heavy processing to the sound without add any more noise. I mean, with the floating point the SNR peaks doesn't change. I can go louder, and keep the data intact when I export to 16 bit.


----------



## wavoman

Y'all might want to read this article in International Audio Review:

The Importance of Digital Filtering

 before concluding hi res digital files offer no improvement over redbook, and that the differences we hear result only from more careful mastering.

 Sure, in theory, as perfectly explained in the Lavery paper cited by Publius, there is no information in the musical waveform above 22.05 that we care about, and sampling the waveform at double that -- redbook 44.1 -- will lead to 100% perfect re-construction of the original waveform ala Nyquist. Right as rain.

 But only with paper and pencil, using the correct sinx/x function. Try doing it in firmware, with hard real time constraints, while the CD spins or the USB cable delivers a bitstream. You can't. The digital filters in use in today's equipment can only approximate the exact reconstruction interpolation. Some do a better job than others, which is why DAC algorithms make a difference, and are proprietary.

 Some smart people figured out that doing a trivial interpolation on the bitstream itself -- adding extra 1's and 0's (oversampling) and altering the D/A digital filters accordingly actually can lead to better re-construction. Clever -- fool the DAC into thinking it has more points, and the algorithms perform better. Here's an analogy (my own -- sorry if it is not perfect): you are translating from English to French in real time for an audience. Suddenly the English speaker says every word twice ... you know what? you make fewer errors in translation, and you drop the dupes before you speak French (but you got two bites at the cherry, and translate better). This all makes sense to me.

 Now take this further -- back at the recording studio, let's sample at a higher frequency. Give the DAC some real (not made up) additional meat to chew on. It does a better job.

 I buy it. Read the citation provided above. With exact math the higher resolution sampling does nothing. But in the real world it helps digital filter designers produce a more correct interpolate.

 The proof is in the listening. I hear it. Could be placebo, who knows? But since you can download Hi Res tracks now, and store them on your music server right along with your EAC perfect bitstream copies of your CDs, why not? I am even making 88.1's out of SACDs and 96's out of DVD-A's so that everything is on the server.

 Storage is so cheap. DACs can handle it. I say go for it.


----------



## darklegion

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_To get round this what you could do is own a 16/44.1 DAC, download 24/96 recordings and convert them to 16/44.1 for playback.

 G_

 

Is there anything to look for when choosing software that can perform this conversion? I.e certain algorithms that are not as good as others, and may cause quality loss? Any recommendations of open source software that can do this, preferably with linux support?


----------



## JaZZ

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_I am even making 88.1's out of SACDs and 96's out of DVD-A's so that everything is on the server._

 

I guess you do this by «recording» from the analogue output of your multiformat player (?) through your soundcard in the case of SACDs. But how do you get your player to output hi-rez PCM signals? Or how do you rip DVD-As?
.


----------



## wavoman

Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_I guess you do this by «recording» from the analogue output of your multiformat player (?) through your soundcard in the case of SACDs. But how do you get your player to output hi-rez PCM signals? Or how do you rip DVD-As?_

 

No! Of course not. D-to-A would be crazy. It's all digital ... you need a DSD-to-LPCM 88.2 downsampler, and a DVD-A decoder that emits 96 LPCM.

 Several sources for this -- either it is built in to your CDP (Sony discourages it, and won't license the DSD logo to players that include it, but so what) or you mod it. Wadia CDP's do the SACD trick but not DVD-A.

 You can mod many Pioneer, Denon, or Marantz DVD players (always cheap on eBay) using the Vanity board from Audio Praise (or have them, or a modder, do it for you):

Audiopraise Vanity :: High Resolution Digital Output for CD/SACD/DVD-Audio

 Or mod an Oppo, or buy an Oppo with this mod done already, from Gary at Custom Home Theatre (he is doing the famous Shawn Fogg mod, and Shawn recommends him!)

http://www.custom-ht.com/

 Then I capture the LPCM with an MAudio 192 board (I use the version modded by Drew at moon-audio, a sponsor here and a very helpful guy).

 This all goes on my brand new totally silent "cappuccino" PC from Unicomp:

cappuccinopc.com : we design, integrate and manufacture small form factor PCs 

 which had driver conflicts at the NY meet so I couldn't demo it, but is all better now and very musical. This PC is half the price of Hush or Stealth and with its ram drive and fanless design makes no noise whatseover. Built in ethernet and WiFi makes connecting to your big 2 TB server in the basement a snap -- 32 Gigs locally for foo bar and the OS and whatnot, while the music is on the server outside the listening area.

 All of this I learned from forums here -- I love this community!


----------



## gregorio

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_before concluding hi res digital files offer no improvement over redbook, and that the differences we hear result only from more careful mastering.

 Sure, in theory, as perfectly explained in the Lavery paper cited by Publius, there is no information in the musical waveform above 22.05 that we care about, and sampling the waveform at double that -- redbook 44.1 -- will lead to 100% perfect re-construction of the original waveform ala Nyquist. Right as rain.

 But only with paper and pencil, using the correct sinx/x function. Try doing it in firmware, with hard real time constraints, while the CD spins or the USB cable delivers a bitstream. You can't. The digital filters in use in today's equipment can only approximate the exact reconstruction interpolation. Some do a better job than others, which is why DAC algorithms make a difference, and are proprietary.

 Some smart people figured out that doing a trivial interpolation on the bitstream itself -- adding extra 1's and 0's (oversampling) and altering the D/A digital filters accordingly actually can lead to better re-construction. Clever -- fool the DAC into thinking it has more points, and the algorithms perform better. Here's an analogy (my own -- sorry if it is not perfect): you are translating from English to French in real time for an audience. Suddenly the English speaker says every word twice ... you know what? you make fewer errors in translation, and you drop the dupes before you speak French (but you got two bites at the cherry, and translate better). This all makes sense to me._

 

For an even more accurate understanding of digital filters have a read of this article: http://www.users.qwest.net/~volt42/c...ng/Filters.pdf

 As I mentioned earlier in this thread, there are various methods in use at the moment for reconstructing the digital audio. Oversampling works well for some DAC manufacturers but better implimentation of 44.1kFs/s filters works better for others. I'm not sure if I agree with your analogy, for it to be accurate, every word would not only need to be repeated twice but would also need to be repeated twice as fast. If anything, this additional overhead is likely to cause more errors rather than fewer. I'm not saying that oversampling DACs are worse in practice, I'm just saying that there is no such thing as perfection in practice.

 I have to say that the reconstruction technology in modern good quality DACs has probably already reached the point where the analogue circuitry in the DAC after the conversion has more effect on sound quality than the actual conversion process itself. If this statement isn't true at this instant, it's getting closer to being true all the time. In other words, what we are talking about here, with regard to sample rate is going to be inaudible to the vast majority of consumers, what happens in the mixing and mastering is going to have a far more obvious impact on sound quality. I personally think there is more to hi-res audio than pure placebo but I also believe that the vast majority of the difference between hi-res and CD can be accounted for by the mixing and mastering process rather than by the differences between the digital formats.

 I should point out that I do not have an indepth understanding of the practical application of anti-alias and anti-image filters. With this in mind, it is my opinion that at this point in time, an 88.2k or 96k sample frequency may sound marginally better on some DACs. For this reason I believe that the best digital audio format at this point in time for the consumer is 16bit 88.2kFs/s. Strange that no one seems to make this format available to consumers.

  Quote:


  Originally Posted by *darklegion* /img/forum/go_quote.gif 
_Is there anything to look for when choosing software that can perform this conversion? I.e certain algorithms that are not as good as others, and may cause quality loss? Any recommendations of open source software that can do this, preferably with linux support?_

 

To be honest I'm not very familiar with consumer software for this process. There is an organisation which makes a dither called Pow-R. You might want to do a search and see which software includes this dither. Pow-R is good quality dither and is administered by committee, so it might be that there is source code available or at least a list of software which includes it.

 G


----------



## wavoman

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_For an even more accurate understanding of digital filters have a read of this article: http://www.users.qwest.net/~volt42/c...ng/Filters.pdf_

 

Excellent article, thanks for the cite and site.

  Quote:


 Oversampling works well for some DAC manufacturers but better implimentation of 44.1kFs/s filters works better for others. 
 

Well put, I agree.

  Quote:


 I'm not sure if I agree with your analogy, for it to be accurate, every word would not only need to be repeated twice but would also need to be repeated twice as fast. If anything, this additional overhead is likely to cause more errors rather than fewer. I'm not saying that oversampling DACs are worse in practice 
 

Point well taken, I was trying only to be illustrative. 


  Quote:


 I'm just saying that there is no such thing as perfection in practice. 
 

My point exactly! We agree again.

  Quote:


 I have to say that the reconstruction technology in modern good quality DACs has probably already reached the point where the analogue circuitry in the DAC after the conversion has more effect on sound quality than the actual conversion process itself. 
 

Again, well put, important, and we are in complete agreement.

  Quote:


 I personally think there is more to hi-res audio than pure placebo but I also believe that the vast majority of the difference between hi-res and CD can be accounted for by the mixing and mastering process rather than by the differences between the digital formats. 
 

And again, x2.

  Quote:


 For this reason I believe that the best digital audio format at this point in time for the consumer is 16bit 88.2kFs/s. Strange that no one seems to make this format available to consumers. 
 

My Wadia CDP outputs 88.2 LPCM from SACDs, and I capture it.


  Quote:


 To be honest I'm not very familiar with consumer software for this process. There is an organisation which makes a dither called Pow-R. 
 

I think your advice here to downsample to 44.1 is not the best -- the poster should buy a DAC that handles 88.2 and 96, like DACMagic or X-DAC or something at reasonable cost.

 But in general our views of digital audio bit rates are identical.


----------



## manaox2

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_I think your advice here to downsample to 44.1 is not the best -- the poster should buy a DAC that handles 88.2 and 96, like DACMagic or X-DAC or something at reasonable cost.

 But in general our views of digital audio bit rates are identical._

 

My OPUS has to use a TPA metronome module to upsample 88.2 because it can't be used natively. No problem there, although dithering down doesn't sound like a good solution. Its rare to find that necessary though in my experience.


----------



## Acix

Quote:


  Originally Posted by *manaox2* /img/forum/go_quote.gif 
_My OPUS has to use a TPA metronome module to upsample 88.2 because it can't be used natively. No problem there, although dithering down doesn't sound like a good solution. Its rare to find that necessary though in my experience._

 

Maybe the Apogee Mini-DAC can be a good solution for dithering problem.
Apogee Electronics > Products > Mini-DAC.


----------



## manaox2

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Maybe the Apogee Mini-DAC can be a good solution for dithering problem.
Apogee Electronics > Products > Mini-DAC._

 

Doesn't have a digital out that I can see for someone that wants to use their current DAC, I'm sure that their are professional inline hardware resamplers somewhere.


----------



## leeperry

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_Y'all might want to read this article in International Audio Review:

The Importance of Digital Filtering

 before concluding hi res digital files offer no improvement over redbook, and that the differences we hear result only from more careful mastering.

 Sure, in theory, as perfectly explained in the Lavery paper cited by Publius, there is no information in the musical waveform above 22.05 that we care about, and sampling the waveform at double that -- redbook 44.1 -- will lead to 100% perfect re-construction of the original waveform ala Nyquist. Right as rain.

 But only with paper and pencil, using the correct sinx/x function. Try doing it in firmware, with hard real time constraints, while the CD spins or the USB cable delivers a bitstream. You can't. The digital filters in use in today's equipment can only approximate the exact reconstruction interpolation. Some do a better job than others, which is why DAC algorithms make a difference, and are proprietary._

 

very interesting link, thanks!

 so what is it that we can do from the PC-side? resample everything to 24/96? last time I tried it made the sound overly brighter(higher THD/saturation/aliasing?) 
	

	
	
		
		

		
		
	


	




 my soundcard has a pretty old AKM DAC, so that means that newer AKM IC's would recreate the original waveform more accurately from 16/44.1 
	

	
	
		
		

		
		
	


	




 more here : http://www.iar-80.com/page21.html

 from page 2 of the same link :

  Quote:


 A conventional CD has only 16 bits, and a Sony DSD master tape has only 8 effective bits of resolution. But IAR research showed long ago that the human ear/brain can hear finer than 20 bits of resolution on music. Since the human ear/brain can discern, apprise, and appreciate the true musical waveform to an accuracy of 20 bit resolution, it follows that any representation of that same music waveform with cruder resolution, e.g. only 16 bit quantization, will only crudely approximate the true and audibly discernible amplitude value of the music waveform for each sample point, and will be somewhat erroneous at each sample point. 
 

 Quote:


 Is there a way to do this? Yes. Increase the sampling rate! If the averaging algorithm has twice as many sample points to average for improving a given audio frequency, then it can do at least twice as good a job, at that frequency, of reducing various digital errors and improving the accuracy of the music waveform. High power averaging algorithms can do even better than twice as well, depending on the curves engineered into the algorithm. If we double the sampling rate, we double the number of sample points per cycle at every audio frequency.


----------



## leeperry

Page Title

  Quote:


 It required the much more aggressive 9th order noise shaping to energize the Purcell's capabilities, and furnish a significant sonic improvement over 16/44 CD. 
 

there's no such thing on PC, right?


----------



## nick_charles

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_
 A conventional CD has only 16 bits, and a Sony DSD master tape has only 8 effective bits of resolution. But IAR research showed long ago that the human ear/brain can hear finer than 20 bits of resolution on music. _

 


 Hmm, I have trawled the AES library and I cannot find this particular piece of research (or any pointers to it) which would surely be quoted by every high res source manufacturer and would be an easy journal publication if it were a well controlled study.

 I have to say I am somewhat skeptrical, but I would be happy to be proved wrong...


----------



## leeperry

actually all that stuff sounds like commercial bs for a $12K DAC 
	

	
	
		
		

		
			





 it uses "9th order noise shaping to energize the Purcell's capabilities" ya know


----------



## Pio2001

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_very interesting link, thanks!_

 

What is it about, exactly ? I read through the first paragraphs, and got so bored that I didn't go on. It all looks like a big introduction to a text that never begins.


----------



## leeperry

Quote:


  Originally Posted by *Pio2001* /img/forum/go_quote.gif 
_What is it about, exactly ? I read through the first paragraphs, and got so bored that I didn't go on. It all looks like a big introduction to a text that never begins._

 

well it's about how crappy usual DAC's are at converting 16/44.1 data into waveforms, and how good that $12K DAC is, especially w/ its own matched amp.... that costs another measly $50K 
	

	
	
		
		

		
		
	


	




 I have to agree that some CD's sounds really low-resolution, but some others sound almost as good as 24/96...so it's all in the dithering/production quality/remastering/source material I'd guess 
	

	
	
		
		

		
		
	


	




 Ozone4 has many different type of dithering algorithms(I put it through on all my movies), but I can't really hear much difference...so the hell w/ it


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_so what is it that we can do from the PC-side? resample everything to 24/96? last time I tried it made the sound overly brighter(higher THD/saturation/aliasing?) 
	

	
	
		
		

		
		
	


	




 my soundcard has a pretty old AKM DAC, so that means that newer AKM IC's would recreate the original waveform more accurately from 16/44.1 
	

	
	
		
		

		
		
	


	




 more here : Page Title

 from page 2 of the same link :_

 

Strange article you linked there. To start with it sounded like a commercial for the very expensive DAC but then in the last paragraph before the second section the article essentially trashes the DAC, stating that it was only the 9th order shaping which made any difference. My guess it that the article is probably nearly 10 years old. There is no doubt that some of it is BS, for instance, being able to hear a 20bit dynamic range on music. It's BS because almost no one would by a music product with the 120dB dynamic range of 20bit and secondly 120dB dynamic range is already attainable on a CD.

 In some respects this thread doesn't help head-fiers because there is not much concrete advice. 24bit over 16bit offers no advantages but sampling rates higher than CD may, under certain circumstances, offer marginal improvement. The difficulty is that I can't tell you which DAC or probably more importantly, which DAC in which system is going to provide that possible improvement. Upsampling a CD (as opposed to native 24/96) is likely to be even more marginal, as of course the frequencies which have been removed by the anti-alias filter during the original recording or mastering can not be put back in by resampling using a higher sampling rate. Once they are gone, they are gone for good! So, the only possible improvement is not in the content of the digital audio but purely in bypassing the DACs 44.1kFs/s filters. However, I don't believe the vast majority of listeners would be able to hear the difference and science cannot provide prove that anyone can hear a difference under controlled conditions.

 The best advice I can give is to try out both formats on your DAC and stick to whichever one you feel sounds the best. Hopefully though, after reading this thread you will have more of an open mind towards the potential quality 16/44.1, as I hope we have shown that potentially CD is the highest quality we will ever need and the chances are that any differences you notice will be from the mixing/mastering process. This information might help you make more of an informed decision rather than just going with 24/94 because of marketing or the fallacy that more data is necessarily better.

 G


----------



## wavoman

Quote:


 In some respects this thread doesn't help head-fiers because there is not much concrete advice 
 

I am willing to summarize into real advice! Here goes -- and I will even venture beyond the current topic just because I have the mike. I expect to be flamed (but don't really care!).

 (1) Buy the best DAC you can, one that can handle 88.2 and 96. Audition carefully, the DAC will play a major role in SQ. I would use an outboard DAC, not one on a sound card. I don't believe you can create low-noise analog signals inside a computer. Consider a modder to improve the off-the-shelf DAC unit. 

 (2) If you buy (download) 24/96 files, leave them, do not downsample to 44.1

 (3) When you rip CDs, rip them (with EAC or the MAC equivalent) and keep them at 16/44.1 Do not artificially oversample them or up-convert them on the PC. Pump the bitstream out at 44.1 (USB or S/PDIF from a sound card) to your DAC, and let its interpolation algorithm do what is does (including its own internal up-converison step if it has one).

 (4) Rip DVD-A's at 96 and rip/down-convert SACDs from DSD to LPCM 88.2. Leave them this way, do not downsample further to 44.1. Do this digitally, not with analog conversion. Sonys, Pioneers, Marantzs, Denons, and Oppos can all be modded to do this. Use a good PC sound card to capture the LPCM. Furthermore, always use the PC as the source, not the original CD (some SACDs might be an exception, but I have yet to find one). 

 (5) Come out of the DAC balanced, into a balanced HP amp, into a balanced pair of HPs, the best you can afford. Carefully consider impedance of your phones and amp, and audition a lot before you choose tubes v solid state, dynamic vs stat. Avoid closed phones unless you need isolation or containment, high-end Denons or Ultrasones being the possible exception (again, audition).

 (6) As much as possible, buy used (here and AudioGon; always use a credit card or PayPal). Consider sending used gear back to the factory for a tune-up if this service is offered, or a modder.

 (7) Don't pay more than the Blue Jeans Cable price for any cable. With no moving parts in use during playback, and no sound waves from speakers, it is hard to see why you need to spend any money on isolation products (feet or shelves).

 (8) Enjoy!


----------



## manaox2

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_I am willing to summarize into real advice! Here goes -- and I will even venture beyond the current topic just because I have the mike. I expect to be flamed (but don't really care!)._

 

Great advice! Thanks


----------



## gregorio

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_I am willing to summarize into real advice! Here goes -- and I will even venture beyond the current topic just because I have the mike. I expect to be flamed (but don't really care!).

 (1) Buy the best DAC you can, one that can handle 88.2 and 96. Audition carefully, the DAC will play a major role in SQ. I would use an outboard DAC, not one on a sound card. I don't believe you can create low-noise analog signals inside a computer. Consider a modder to improve the off-the-shelf DAC unit. 

 (2) If you buy (download) 24/96 files, leave them, do not downsample to 44.1

 (3) When you rip CDs, rip them (with EAC or the MAC equivalent) and keep them at 16/44.1 Do not artificially oversample them or up-convert them on the PC. Pump the bitstream out at 44.1 (USB or S/PDIF from a sound card) to your DAC, and let its interpolation algorithm do what is does (including its own internal up-converison step if it has one).

 (4) Rip DVD-A's at 96 and rip/down-convert SACDs from DSD to LPCM 88.2. Leave them this way, do not downsample further to 44.1. Do this digitally, not with analog conversion. Sonys, Pioneers, Marantzs, Denons, and Oppos can all be modded to do this. Use a good PC sound card to capture the LPCM. Furthermore, always use the PC as the source, not the original CD (some SACDs might be an exception, but I have yet to find one). 

 (5) Come out of the DAC balanced, into a balanced HP amp, into a balanced pair of HPs, the best you can afford. Carefully consider impedance of your phones and amp, and audition a lot before you choose tubes v solid state, dynamic vs stat. Avoid closed phones unless you need isolation or containment, high-end Denons or Ultrasones being the possible exception (again, audition).

 (6) As much as possible, buy used (here and AudioGon; always use a credit card or PayPal). Consider sending used gear back to the factory for a tune-up if this service is offered, or a modder.

 (7) Don't pay more than the Blue Jeans Cable price for any cable. With no moving parts in use during playback, and no sound waves from speakers, it is hard to see why you need to spend any money on isolation products (feet or shelves).

 (8) Enjoy!_

 

Generally good advice:

 1) I'm not sure that the DAC will play a major role in SQ. Headphones (or speakers + environment), the amp and of course the origianl recording quality are more important to sound quality than the DAC. The transport (if you have one) is probably the only other bit of the kit in the chain less important than the DAC as far as SQ is concerned.

 2. This is true for some systems but not all.

 3. Agreed.

 4. True, although again, down sampling to 16/44.1 will probably not affect SQ for most systems.

 5. Balanced will make more of a difference if there is high interference or cable runs longer than a few meters. If neither of these two situations apply, the user is extremely unlikely to hear any difference between balanced and unbalanced.

 6. You can make up your own mind about buying new or used.

 7. Agreed.

 G


----------



## mbd2884

In response to wavoman the best advice I've heard from this thread.

 16/44.1 is more than adaquate for PLAYBACK and spending more for the 24/96 is pointless.

 If you are working with 24/96, may be advisable or even 32 from the STUDIO. 

 Playback and Studio work are two different things. Myself as a consumer buying an expensive DAC just for the 24/96 doesn't seem to make much sense unless my hearing spontaneously evolves to beyond human ability.


----------



## dex85

pardon my ignorance, but shouldn't this make 'vinyl sounds so much better than cd' myth explode as well?(if there is any myth to it) otherwise what makes vinyl supposedly sound better than ripped cd if its not 24/94?


----------



## gregorio

Quote:


  Originally Posted by *dex85* /img/forum/go_quote.gif 
_pardon my ignorance, but shouldn't this make 'vinyl sounds so much better than cd' myth explode as well?(if there is any myth to it) otherwise what makes vinyl supposedly sound better than ripped cd if its not 24/94?_

 

Depends on what you mean by "better". If you mean a more accurate reproduction of what is recorded from the microphone then yes, CD is far more accurate than vinyl. However, if you mean what makes vinyl sound better subjectively to some listeners, then it's actually the inaccuracies. Things like tape saturation during recording/mixing, inaccuracies and compression artifacts in the higher frequencies. In fact there are a range of distortions and inaccuracies in analogue/vinyl which many people feel gives a richer, warmer sound than digital.

  Quote:


  Originally Posted by *mbd2884* /img/forum/go_quote.gif 
_16/44.1 is more than adaquate for PLAYBACK and spending more for the 24/96 is pointless.

 If you are working with 24/96, may be advisable or even 32 from the STUDIO._

 

As a general rule I would tend to agree with your first statement. Although, there can be a minute difference which may be detected by some audiophiles.

 Your second point; 32bit provides no benefit over 24bit for studio work and in some cases 24bit is superior to 32bit. 32bit only exists to allow standard computer CPUs to process the digital audio data using a relatively cheap software packages. More professional computer based audio systems, such as ProTools HD, contain their own DSP chips and process the audio at 24bit.

 G


----------



## dex85

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Depends on what you mean by "better". If you mean a more accurate reproduction of what is recorded from the microphone then yes, CD is far more accurate than vinyl. However, if you mean what makes vinyl sound better subjectively to some listeners, then it's actually the inaccuracies. Things like tape saturation during recording/mixing, inaccuracies and compression artifacts in the higher frequencies. In fact there are a range of distortions and inaccuracies in analogue/vinyl which many people feel gives a richer, warmer sound than digital._

 

heh, you are a walking audio encyclopedia. i've never understood what was the fuss about vinyl, even after i auditioned some not-so-cheap turntable. so i'm glad you made it clear for me. lately i've been on the verge of buying a turntable (i was getting tired of my friends picking on me for sticking with 'underachieving' digital), so you actually saved me some money


----------



## manaox2

Quote:


  Originally Posted by *dex85* /img/forum/go_quote.gif 
_heh, you are a walking audio encyclopedia. i've never understood what was the fuss about vinyl, even after i auditioned some not-so-cheap turntable. so i'm glad you made it clear for me. lately i've been on the verge of buying a turntable (i was getting tired of my friends picking on me for sticking with 'underachieving' digital), so you actually saved me some money 
	

	
	
		
		

		
		
	


	


_

 

Not completely sticking up for Vinyl, but the mastering on many records seem of higher quality, similar to the argument that SACD and DVD-A may often have better mastering. I'm sticking with needle drops for playback though, don't see the point in getting two mediocre rigs when I can get one higher end setup.


----------



## gregorio

Quote:


  Originally Posted by *dex85* /img/forum/go_quote.gif 
_heh, you are a walking audio encyclopedia. i've never understood what was the fuss about vinyl, even after i auditioned some not-so-cheap turntable. so i'm glad you made it clear for me. lately i've been on the verge of buying a turntable (i was getting tired of my friends picking on me for sticking with 'underachieving' digital), so you actually saved me some money 
	

	
	
		
		

		
		
	


	


_

 

I have met people with a far greater knowledge of audio than me but I've picked up a fair amount in the 20+ years I've been in the industry.

 What I said about vinyl sounded much more negative than I meant it to. Everything I said was true but not all artifacts or inaccuracies are a bad thing. Just because digital is more accurate does not necessarily make it sound better than analogue. For many years DSP manufacturers have been trying to make analogue emulation processors, to try and give digital audio that warmer analogue sound. Nothing in my opinion has been too successful so far though. I personally prefer the sound of very high quality digital but it's a tough call and I certainly wouldn't knock anyone who felt analogue sounded better.

 G


----------



## ILikeMusic

One can't eliminate the real-world realities inherent in the analog/digital debate though. In a studio environment high-end analog and digital may sound equivalent but it's very difficult to carry that through in distribution. Analog quality degrades with each step, beginning with the original pressing. Then add ripping the vinyl copy to tape or lossless, etc., and quality degrades further (and with each additional playing) whereas in the digial domain quality of the nth copy and playing will be essentially identical to the original release. That's a big advantage to overcome in the real world.

 .


----------



## dex85

^^fair enough, im not even close to competent to make any calls about vinyl vs digital. i've just auditioned one turntable (i think it was Rega P2+some Nad phono preamp+some onkyo receiver+Mordaunt Short Alumni speakers). it sounded way too warm and sweet for my taste. maybe it has to do with me being more of an analytical listener. long story short, i wasn't impressed.

 also good point about going analog and digital route at the same time. that's a blackhole for money, im sticking with digital.


----------



## leeperry

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_For many years DSP manufacturers have been trying to make analogue emulation processors, to try and give digital audio that warmer analogue sound. Nothing in my opinion has been too successful so far though. I personally prefer the sound of very high quality digital but it's a tough call and I certainly wouldn't knock anyone who felt analogue sounded better._

 

I've tried a lot of plugins that are supposed to give a "tube" colored sound....and simply end up outputting distorted agressive sound.
 it prolly works on guitar tracks in Cubase, but from what I've been told a tube amp actually makes the sound more "mellow", more laid back...less agressive than solid state.

 I have to admit that I very much loved OzoneMP on my cd3k coz this thing had unbearable trebles to begin with(well I didn't a have matching amp so it was poorly driven)....but getting "tube" sound on PC is mostly bs I think.

 like that SSL EQ, the sound all the engineers in the world crave for...yeah right 
	

	
	
		
		

		
			









  Quote:


 The unique sound of Solid State Logic’s 4000 Series analogue mixing consoles is sought after worldwide. Engineers of pop and rock music, broadcast transmissions and television post-production value the SSL 4000’s flexible dynamics chain as much as the trademark SSL “punchy” sound. Waves and SSL engineers have worked together for over a year to recreate the sound characteristics of the classic SSL 4000 Series E and G Series consoles. Now, those who “mix in the box” can achieve the sound they thought they’d lost when they moved to the digital world.


----------



## Acix

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_32bit provides no benefit over 24bit for studio work and in some cases 24bit is superior to 32bit. 32bit only exists to allow standard computer CPUs to process the digital audio data using a relatively cheap software packages. More professional computer based audio systems, such as ProTools HD, contain their own DSP chips and process the audio at 24bit.

 G_

 


 No, Gregorio. 32 bit provides a lot of benefits compared to 16 and 24 bit. 32 bit is way more flexible than 24, and as I already explained to you before, 32 bit keeps the original noise floor down and allows the application of heavy processing to the sound without adding any additional noise. With the floating point the SNR peaks don't change. You can go louder and keep the data intact when exporting to 16 bit.

 I don't understand why you insist that there are no benefits to mixing, editing and mastering in 32 bit. It looks like you either don't have any real experience with it, or you just want to satisfy your argument in your own thread. I've worked in 32 bit for the last 5 years and every musician who has worked in 32 bit notices the benefits right away. Also 24 bit is going to sound much better than 16 bit. As a mastering engineer, I get files in 16/96, 16/48 and 24/96 and I can tell you that everything is better than 16/44. This was proven to me when I got my first Sony DAT machine back in '93 with a 16/48. But yes, unfortunately, the better audio converters are going to be more expensive.

 Here is a link that elaborates on bit depth more clearly than I have.

Bit Depth - Audacity Wiki


----------



## DistortingJack

Acix, where the hell did you get that chart? To my eyes it looks like complete bollocks. Not only do microphones vary WILDLY in dynamic range, but digital does NOT have headroom. You may choose to call that way the difference between 0dBFS and the "safe recording level", but this varies wildly on the type of music and instrument recorded. Also, different types of dithering change the noise floor and thus the dynamic range, and the size of the speaker changing the equation completely, I mean no.
 All bollocks. Sorry.


----------



## Acix

Quote:


  Originally Posted by *DistortingJack* /img/forum/go_quote.gif 
_Acix, where the hell did you get that chart? To my eyes it looks like complete bollocks. Not only do microphones vary WILDLY in dynamic range, but digital does NOT have headroom. You may choose to call that way the difference between 0dBFS and the "safe recording level", but this varies wildly on the type of music and instrument recorded. Also, different types of dithering change the noise floor and thus the dynamic range, and the size of the speaker changing the equation completely, I mean no.
 All bollocks. Sorry._

 

LOL...if it was up to you and gregorio, we'd probably still be living under the belief that the world is flat, and in 16 bit.

 Here's the link to the chart: Headroom - Wikipedia, the free encyclopedia


----------



## Clutz

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_LOL...if it was up to you and gregorio, we'd probably still be living under the belief that the world is flat, and in 16 bit.

 Here's the link to the chart: Headroom - Wikipedia, the free encyclopedia_

 

What I find most ironic about your attitude is that you didn't understand the difference between a lossy and a lossless codec- yet you claim to be some expert on recording and mastering. I'm sorry, but you lost all credibility when you didn't understand what a lossless codec meant.


----------



## DistortingJack

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_LOL...if it was up to you and gregorio, we'd probably still be living under the belief that the world is flat, and in 16 bit.

 Here's the link to the chart: Headroom - Wikipedia, the free encyclopedia_

 

I make music, and I know my numbers. The chart is technically right, because almost every single level has been chosen arbitrarily. "typical" in this chart means whatever the guy decided to prove his claim.
 It doesn't take the signal to noise ratios of different live sound systems. Is it using powered speakers? Is it using noisy mics in a noisy stage to record a quiet acoustic guitar? Is it using a condenser microphone on a snare drum? Is leakage considered noise or not? Is it just the inherent noise floor of the system, then what system? Digital condenser microphones have a real noise floor 130 dB below clipping point. 100dB SPL but is it percussive music? Because it's RMS and thus percussive music can blow up speakers at the same volumes as highly compressed non-percussive music. So they use a −18 dBFS alignment level, why does it show almost 25 dB of headroom in 24bit? Is the compact disc dithered? What dither did it use? Good dither can bring down the noise floor by several dB. And in the end, speakers. So my laptop speakers can output 105 dB? But the big JBL line array speakers can output more than that. 200 speakers can output a lot more than that and it has been known to happen in large festivals. How far away are you from the speakers when are you doing the measurement at? Amplitude decreases very, very quickly in open air. Or maybe it would be in a tiny club? So, are people chatting noise? 

 Instead of putting up fancy diagrams which mean absolutely nothing to prove your points, you should lose the attitude and start reading up science books, not press releases for semi-experimental prosumer equipment. No, your ADAMS are not the best speakers in the world (they are very nice but they are not flat at the very high frequencies you claim to clearly hear). 
 By the way, I record to 24-bit 96 kHz and I mix with 32bit floating point. The reason why is because after every digital process, there is a very steep low-pass filter, which does introduce some artifacts if the plugin is not perfectly designed, and introduces latency so programmers compromise on it. So, even if one 96 to 44.1 can't be heard (as in can't, have you voted on the poll by the way?), constant filtering does detract from sound quality, by introducing pre-echoes and the like, but on frequencies BELOW 22 kHz, not above. 
 Also it's nice to pitch something down a whole octave and have musical information above 10k.
 Actually, I do not agree with Gregorio in 24 being better than 32 bit. Fixed point can be better than floating point at low resolutions (8-bit) because floating point changes the noise level constantly and it can get very annoying. 32-bit floating point is 24-bit resolution + 8 bit volume information, which is nice because you can stop worrying about having faders that are too low, in the mixer and during processing, since floating point keeps the resolution at very low levels. Convenience in a DAW is always appreciated, but not necessary.
 The world is not flat, and in the music production world, higher resolutions are handy, but in the consumer format, with real music, as long as your DA converter is good enough, the maximum resolution that can be heard is 16bit 44.1kHz. Period.


----------



## Ham Sandwich

He he he, the Google ad I see at the top of the page for this thread is for HDtracks. "You can hear the difference" "Click here for a FREE 96/24 album" "Hear what your headphones really sound like"


----------



## wnmnkh

Acix, there is some very important question left unanswered so far.

 I perfectly understand 24bit/32bit are good for 'recording', but we are talking about end users.

 Why we need MOAR headrooms and 96+ SNR for the end user playback? I mean if I want to hear the benefits I need to play the songs very, very loudly (more than enough to cause permanent hearing damage) and this is assumed that I have top-class speakers/headphones/amps/dacs to match the spec of 24bit recordings.


----------



## manaox2

Quote:


  Originally Posted by *Ham Sandwich* /img/forum/go_quote.gif 
_He he he, the Google ad I see at the top of the page for this thread is for HDtracks. "You can hear the difference" "Click here for a FREE 96/24 album" "Hear what your headphones really sound like"_

 

Speaking of, I have three albums from HDTracks. There free one isn't that impressive, the greatest audiophile vocal and ultimate demo vol. 2 albums are really good IMO. I don't think it has to do with the bitrate, but the mastering is pretty spectacular and about the closest thing I've seen to SACD you can buy and download in digital. Their selection could be a little bigger though.</shill>


----------



## wavoman

Quote:


  Originally Posted by *DistortingJack* /img/forum/go_quote.gif 
_in the consumer format, with real music, as long as your DA converter is good enough, the maximum resolution that can be heard is 16bit 44.1kHz. Period._

 

Let's explore this. Let's first assume that there is no music above 22.05. Then, if you could build a perfect boxcar digital filter, you could recover the original waveform perfectly from 44.1 (Nyquist). So far so good.

 But you can't build such a filter, real-world digital filters are only approximate, so if we had more sampling points the reconstruction would be better.

 If the recording engineer has created a 96 bitstream, why not sell it to us unwashed consumers? We can easily download it, store it, play it back thru our DACs. Logic says the end result has to be a better reconstruction of the original recorderd waveform, no? (Not simply because it is more points, but because Nyquist does not apply exactly in the real world).

 Selling us a higher bitstream is precisely what DVD-A does, and I'm all for it. Most DVD-A's present a 96 LPCM bitstream (some 192), easy to capture and playback from a computer+DAC. SACDs are tougher, the equipment we can buy downconverts to 88.2 or 176.4 LPCM which is not the original DSD bitstream. But still it is easy to store and playback, so why not?

 Maybe your argument is: *we can't hear the difference between a 96 and a 44.1 reconstruction*, even if the 96 is the more accurate. That might be true, and some group-blind experiments claim they have proven this, but I think the experiments were poor. 

 I believe the right at-home experiment is to take a DVD-A, rip it at 96 and rip it again at 44.1. A modded Oppo can do this. Then listen on your own rig and compare. Some of us tried this with a pair of FLAC files that a member was circulating around here a year ago, but I couldn't get them to play properly, and I didn't see any follow ups ... when I get my modded CDP I plan to do this myself.


----------



## gregorio

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_No, Gregorio. 32 bit provides a lot of benefits compared to 16 and 24 bit. 32 bit is way more flexible than 24, and as I already explained to you before, 32 bit keeps the original noise floor down and allows the application of heavy processing to the sound without adding any additional noise. With the floating point the SNR peaks don't change. You can go louder and keep the data intact when exporting to 16 bit.

 I don't understand why you insist that there are no benefits to mixing, editing and mastering in 32 bit. It looks like you either don't have any real experience with it, or you just want to satisfy your argument in your own thread. I've worked in 32 bit for the last 5 years and every musician who has worked in 32 bit notices the benefits right away. Also 24 bit is going to sound much better than 16 bit. As a mastering engineer, I get files in 16/96, 16/48 and 24/96 and I can tell you that everything is better than 16/44. This was proven to me when I got my first Sony DAT machine back in '93 with a 16/48. But yes, unfortunately, the better audio converters are going to be more expensive._

 

I think you mis-understand. There is no direct comparison between 24bit and 32bit because 24bit is integer and 32bit is floating point. There is absolutely no noise floor benefit of 32bit over 24bit, in fact the opposite can be true. The point you seem to be missing is that your 32bit value is not a number (integer) but a float. The whole point being that the decimal point is free to "float". At some stage this floating point data has to be converted to fixed integer (every time you play it back for example). The software then has to decide where that decimal point is going to be fixed. In other words, 32bit does not have a fixed dynamic range like 16bit or 24bit so under some conditions 32bit may provide poorer dynamic range than fixed 24bit. The second point to consider is the additional noise incurred during effects processing, most 24bit systems apply the processing algorithms using double precision (IE. 48bit) maths and the better systems then dither the result back to the 24bit buss.

 I do not believe your claim that all the musicians you've worked with for 5 years can tell the difference between 32bit and 24bit. You've completely made that up!! Firstly how can they hear the difference when no DAC on the planet can fully resolve even 24bit? Please tell me the DAC you use and it's dynamic range.

 I think Acix, that despite this thread you still do not understand how digital audio works. This is a serious concern as you claim to not only be a professional in the field but a mastering engineer, the most skilled audio engineering role in the business. The article you posted a link to deals with consumer and pro-sumer quality, for instance it keeps referring to the abilities of the "Audacity" card. What I am talking about is the high end of the professional market, not the pro-sumer market.

 Lastly, if 32bit software is so superior to 24bit why do the vast majority of world class studios use 24bit fixed integer systems? You think maybe they don't know about it yet or can't afford a few hundred dollars for Cubase or Logic? Why are all the best ADCs and DACs fixed integer? If you are a professional Acix and you are going to quote the numbers then you had better be sure you have a fairly good idea of how the numbers work!

 For others reading these last few posts, we are talking about the recording and mixing of music rather than the playback of a distribution mix. The conversation now has no direct relevance to the quality or performance of your playback system.

 G


----------



## gregorio

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_But you can't build such a filter, real-world digital filters are only approximate, so if we had more sampling points the reconstruction would be better.

 If the recording engineer has created a 96 bitstream, why not sell it to us unwashed consumers? We can easily download it, store it, play it back thru our DACs. Logic says the end result has to be a better reconstruction of the original recorderd waveform, no? (Not simply because it is more points, but because Nyquist does not apply exactly in the real world)._

 

More sampling points do not by themselves make the reconstruction better. I realise that it's not possible to make a perfect digital anti-image filter but that's not really an issue as we can get very close to perfect. Taken in context the digital filters have no real impact. When I say "taken in context" I mean that there is no piece of equipment in the whole audio chain (except a digital transport) which is perfect. Certainly not the microphones or mic pre-amps, not the ADC or mixing desk or any of the mixing, production or mastering processes, not the consumer's DAC, amp and certainly not their speakers or headphones. In this context, the slight lack of perfection in the anti-imaging filters is virtually irrelevant.

  Quote:


  Originally Posted by *wnmnkh* /img/forum/go_quote.gif 
_Acix, there is some very important question left unanswered so far.

 I perfectly understand 24bit/32bit are good for 'recording', but we are talking about end users.

 Why we need MOAR headrooms and 96+ SNR for the end user playback? I mean if I want to hear the benefits I need to play the songs very, very loudly (more than enough to cause permanent hearing damage) and this is assumed that I have top-class speakers/headphones/amps/dacs to match the spec of 24bit recordings._

 

What you say is completely true! We have gotten off the track a little and gone into the professional realms of formats and techniques employed during the creation of the music. Of course how we create the music has a direct impact on the sound when it is played back by a consumer but what we're talking about with Acix is not directly relevant for the consumer.

 G


----------



## uofmtiger

Quote:


 (4) Rip DVD-A's at 96 and rip/down-convert SACDs from DSD to LPCM 88.2. Leave them this way, do not downsample further to 44.1. Do this digitally, not with analog conversion. Sonys, Pioneers, Marantzs, Denons, and Oppos can all be modded to do this. Use a good PC sound card to capture the LPCM. Furthermore, always use the PC as the source, not the original CD (some SACDs might be an exception, but I have yet to find one). 
 

Just have some questions related to recording vinyl to a computer. My computer is a Sony with DSD recording built in. It also has several different choices for wav formats. I have been recording vinyl to DSD for archiving and then also outputting to a 24/192 wav file. The main reason is that my PS Audio DLIII DAC samples to 24/96 or 24/192. Would it make a bit of difference to output them to 16/44 or 16/88.2 and just allow my DAC to upsample them?


----------



## leeperry

so, to go back to the idea that all DAC's are not equal when it comes to reconstructing the original waveform from a 16/44.1 audio file....what do you guys think of that "VLSC" Onkyo thingie : 
Translated version of http://www.e-onkyo.com/goods/detail.asp?cgds_id=SEU33GXVB&ictg_no=34

TECHNOLOGY | ONKYO Asia and Oceania Website

http://www.intl.onkyo.com/technology/glossary/vlsc.html










 I might take the plunge on one of these


----------



## wavoman

Quote:


  Originally Posted by *uofmtiger* /img/forum/go_quote.gif 
_...I have been recording vinyl to DSD for archiving and then also outputting to a 24/192 wav file. The main reason is that my PS Audio DLIII DAC samples to 24/96 or 24/192. Would it make a bit of difference to output them to 16/44 or 16/88.2 and just allow my DAC to upsample them?_

 

Different folks here will give you different answers, but I would not change what you are doing. It seems perfect. Any DAC upsampling to 96 or 192 will not reproduce the same result as your direct DSD downsample to that rate. Do only one re-sample, not two!

 Your DAC supports 24/192, you can get your Sony to store DSD and downsample to 24/192, you are done man and have a perfect set-up. OK, you could also downsample the DSD to 96 and feed that to the DAC (no further upsampling) ... that will in all likelihood sound the same.

 There's more: does your DAC handle 88.2 with no further re-sampling? You are unlikely to hear a difference between 88.2 and 96, and 88.2 I believe is an easier downsample for the DSD downsamplng algorithm (this is what my Wadia does). 

 Not done yet -- many DACS re-sample _everything _to some common (high) bit rate. Check yours, and downsample the DSD to that. Now you probably have achieved the least processing, which is the goal.


----------



## wavoman

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_More sampling points do not by themselves make the reconstruction better._

 

Actually, for many algorithms, they do. I agree that in theory you don't need to go past the Nyquist bound, but not so in practice. I also agree that some algorithms might not use the extra points, but I believe some do (it is hard to know, the algorithms are not well documented).

  Quote:


 In this context, the slight lack of perfection in the anti-imaging filters is virtually irrelevant. 
 

You kinda have to prove that. It is not self-evident. In fact, just the opposite. I personally think it's probably true, but it remains unproven.

 I believe the reconstruction will be better with more points using the type of algorithms actually built in to most DACS today (from the little I can glean from manufacturer's disclosures). I also believe -- but can't prove -- the the better reconstruction is not audible.

 The AES papers don't prove it to me. So for now, I rip my SACDs at 88.2 and DVD-A's at 96, since there is no loss to doing this (except disk space, which is today essentially free), and pick a DAC that does well with these (no further arbitrary upsampling, except that built-in to their reconstruction algorithm). I do plan to try a direct rip to 44.1 and a careful compare someday, but that is not an easy experiment for me to set up. And I don't really care, since there is no loss in doing what I am doing, and there might be a gain.


----------



## gregorio

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_Actually, for many algorithms, they do. I agree that in theory you don't need to go past the Nyquist bound, but not so in practice. I also agree that some algorithms might not use the extra points, but I believe some do (it is hard to know, the algorithms are not well documented).

 You kinda have to prove that. It is not self-evident. In fact, just the opposite. I personally think it's probably true, but it remains unproven.

 I believe the reconstruction will be better with more points using the type of algorithms actually built in to most DACS today (from the little I can glean from manufacturer's disclosures). I also believe -- but can't prove -- the the better reconstruction is not audible._

 

The only proof I have is some DBT I took part in a few years ago, under studio conditions. There are also a number of DBTs carried out under laboratory conditions which have been published, where no one could tell the difference between various sample rates.

 I can't prove or disprove that reconstruction is more accurate with more sampling points because we are going beyond the precision of studio microphones and equipment to measure the tiny effect of the anti-imaging filters, you'd need laboratory equipment to do that. This fact alone would tend to imply the statement is false or at least be pretty much irrelevant to recording studios. I personally record and mix music at 24/88.2 and TV/Film audio at 24/48, which I believe covers all the bases at the moment.

 G


----------



## gregorio

Quote:


  Originally Posted by *uofmtiger* /img/forum/go_quote.gif 
_Just have some questions related to recording vinyl to a computer. My computer is a Sony with DSD recording built in. It also has several different choices for wav formats. I have been recording vinyl to DSD for archiving and then also outputting to a 24/192 wav file. The main reason is that my PS Audio DLIII DAC samples to 24/96 or 24/192. Would it make a bit of difference to output them to 16/44 or 16/88.2 and just allow my DAC to upsample them?_

 

There probably won't be much of a difference. Certainly use 16bit and then provided your DAC handles native 88.2, I would use that too. If your DAC only outputs 24/192 then you may as well stick to that unless storage space is at a premium.

 G


----------



## Acix

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I think you mis-understand. There is no direct comparison between 24bit and 32bit because 24bit is integer and 32bit is floating point. There is absolutely no noise floor benefit of 32bit over 24bit, in fact the opposite can be true. The point you seem to be missing is that your 32bit value is not a number (integer) but a float. The whole point being that the decimal point is free to "float". At some stage this floating point data has to be converted to fixed integer (every time you play it back for example). The software then has to decide where that decimal point is going to be fixed. In other words, 32bit does not have a fixed dynamic range like 16bit or 24bit so under some conditions 32bit may provide poorer dynamic range than fixed 24bit. The second point to consider is the additional noise incurred during effects processing, most 24bit systems apply the processing algorithms using double precision (IE. 48bit) maths and the better systems then dither the result back to the 24bit buss.

 I do not believe your claim that all the musicians you've worked with for 5 years can tell the difference between 32bit and 24bit. You've completely made that up!! Firstly how can they hear the difference when no DAC on the planet can fully resolve even 24bit? Please tell me the DAC you use and it's dynamic range.

 I think Acix, that despite this thread you still do not understand how digital audio works. This is a serious concern as you claim to not only be a professional in the field but a mastering engineer, the most skilled audio engineering role in the business. The article you posted a link to deals with consumer and pro-sumer quality, for instance it keeps referring to the abilities of the "Audacity" card. What I am talking about is the high end of the professional market, not the pro-sumer market.

 Lastly, if 32bit software is so superior to 24bit why do the vast majority of world class studios use 24bit fixed integer systems? You think maybe they don't know about it yet or can't afford a few hundred dollars for Cubase or Logic? Why are all the best ADCs and DACs fixed integer? If you are a professional Acix and you are going to quote the numbers then you had better be sure you have a fairly good idea of how the numbers work!

 For others reading these last few posts, we are talking about the recording and mixing of music rather than the playback of a distribution mix. The conversation now has no direct relevance to the quality or performance of your playback system.

 G_

 



 Hehehe... well...even post-production dudes like you can mix and edit in 32.

 I can see that you don't have any experience working in 32bit floating point, but is doesn't make me or others wrong. You can try working in 32bit floating point, experience it for yourself it and see what I'm talking about. It's not wise to talk about things that you don't have your own experience with.

 The vast majority of world class studios use 32bit is floating point as well. In many studios you will find Nuendo, Wavelab, Cubase and Logic. They probably find benefits of 32 floating point, too. Sorry gregorio, 32 floating point is not theoretical, it's been practical for years.


----------



## gregorio

Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Hehehe... well...even post-production dudes like you can mix and edit in 32.

 I can see that you don't have any experience working in 32bit floating point, but is doesn't make me or others wrong. You can try working in 32bit floating point, experience it for yourself it and see what I'm talking about. It's not wise to talk about things that you don't have your own experience with.

 The vast majority of world class studios use 32bit is floating point as well. In many studios you will find Nuendo, Wavelab, Cubase and Logic. They probably find benefits of 32 floating point, too. Sorry gregorio, 32 floating point is not theoretical, it's been practical for years._

 

You are mistaken at every level, I've used 32bit extensively and yes, you will find some studios who use Nuendo, Logic, etc., these programs tend to dominate the budget studio market. The higher class commercial studios are largely dominated by 24bit systems. As far as audio-post is concerned, no one would try to mix a film in Cubase or Nuendo or any other 32bit system, the market is completely dominated by 24bit systems such as ProTools. If you want to fool yourself or be defensive and believe 32bit float is better then feel free. But you're not going to convince me because I know both in practical application and in theory that 32bit is far from superior to 24bit.

 G


----------



## Gamerphile

Not really following the thread but just an input for thought:

 Had a 32bit float / 96KHz original master recording I got from a friend who actually did the album recording event with a pianist. Resampled it to 24 and 16 bit versions still with 96KHz sampling. System set to 96KHz operation and 16 or 24 bit operation - can't do 32 bit float in the DAC/HW part only on the player.

 My brief conclusion on comparision was that with the original recording you had to turn the volume up so loud to hear the voices negotiating at the at start and all the drama that I'd say something like 18-24 dB was needed. That left the CD bit quality 16 bit unsigned int version to have way to high noise floor where things where lost due to noise and related distortions. Now this was a typical classical recording compared to my collection - these are always 3-4 click's or 18-24 dB lower than a lot of the pop and rock disks today...
 This was not present in 24 bit version.
 So I really think that headroom is needed...


----------



## gregorio

Quote:


  Originally Posted by *Gamerphile* /img/forum/go_quote.gif 
_Not really following the thread but just an input for thought:

 Had a 32bit float / 96KHz original master recording I got from a friend who actually did the album recording event with a pianist. Resampled it to 24 and 16 bit versions still with 96KHz sampling. System set to 96KHz operation and 16 or 24 bit operation - can't do 32 bit float in the DAC/HW part only on the player.

 My brief conclusion on comparision was that with the original recording you had to turn the volume up so loud to hear the voices negotiating at the at start and all the drama that I'd say something like 18-24 dB was needed. That left the CD bit quality 16 bit unsigned int version to have way to high noise floor where things where lost due to noise and related distortions. Now this was a typical classical recording compared to my collection - these are always 3-4 click's or 18-24 dB lower than a lot of the pop and rock disks today...
 This was not present in 24 bit version.
 So I really think that headroom is needed..._

 

As has been mentioned. The headroom allowed by 24bit is essential during recording but when mastered down to CD, no headroom is necessary and then 16bit is more than adequate. If you heard considerably more noise on the 16bit version than the 24bit version, it hasn't been mastered properly (or at all).

 G


----------



## Ham Sandwich

As a software guy, the thought of using 32-bit floats for better accuracy over 24-bit integer math is not obvious. Floating point math is messy. There are whole sections of senior and graduate level numerical analysis math courses devoted to the problems of floating point math in computers. Floating point and accuracy are not words that automatically go together.

 Integer math is always what it is. Predictable. Not so much funny business. Just gotta be careful with overflow and that is easy to deal with.

 Just because floating point has 32 bits doesn't automatically mean it is going to be better for accuracy compared to integer math.


----------



## DistortingJack

Quote:


  Originally Posted by *Gamerphile* /img/forum/go_quote.gif 
_[...]with the original recording you had to turn the volume up so loud to hear the voices negotiating at the at start and all the drama that I'd say something like 18-24 dB was needed. That left the CD bit quality 16 bit unsigned int version to have way to high noise floor where things where lost due to noise and related distortions. Now this was a typical classical recording compared to my collection - these are always 3-4 click's or 18-24 dB lower than a lot of the pop and rock disks today...
 This was not present in 24 bit version.
 So I really think that headroom is needed..._

 

1. 24 dB gain is more than the difference in level from the quietest CD ever released to the loudest ever (arguably Metallica's Death Magnetic). A piano recording should be about 8-10 dB lower than a rock album, at most.
 2. If things are "distorted" in the 16 bit version, then you used a **** dither. The noise floor of a dithered 16 bit file is about -120, definitely under -100dB. If someone allowed useful noises to be at that level, then it was a really bad recording session.
 3. You added an absolutely massive amount of gain. That's the kind of "editing" we're discussing in this thread! Someone who's not messing around adding huge amounts of gain to their recordings should NOT be able to hear any difference in noise floor. So, take the stupidly quiet file, add the necessary gain, and THEN dither down. Christ.

 So, to finish up: Record at 96kHz, 24 bit, mix with a 32bit or more floating point DAW (Gregorio, sorry mate, Pro Tools uses a 48 bit fixed-point mixer) and create a 24 bit, 96kHz premaster. 
 Take to a mastering engineer for careful last processing and level adjustment. On most music there is also some degree of dynamic range compression;
 Mastering engineer then takes the resulting 96kHz, 24 bit file he just made, dithers it to 16 bit, and resamples to 44.1kHz. This file is identical sounding to the previous one, except it fits on CD's, on any playback equipmant and at any comfortable level, because it's using all the 16 bit dynamic range and is not stupidly quiet like your recording.
 That's all, folks.


----------



## spanimal

I have read this post from the beginning and I am not impressed. I like my gear, it sounds really nice with good recordings on CD. I will fire off some shots now.

 I love fishing. The sights and sound by a quiet river plays a large part. Firstly Blu-Ray with all its higher definition is not even close to the visual stimulus when comparing, it is woefully inadequate. You know what is coming now...

 The sounds of the gentle waves lapping against rocks. The chirp of the birds and the buzzing of insects - I am no sound engineer, but challenge me when I say I have a high-fidelity reference for which no modern equipment can even come close. I have never been a spectator to a rock concert - though I do believe a any studio recording will obliterate a live band when it comes to high fidelity. I used to be a drummer in my high school band and have played in concerts with thousands of spectators - so I don't really know what the audience hears - though I have an opinion.

 How then can one claim that our current methodologies of audio reproduction are sufficient when I am declaring that all our technologies and techniques are in fact extraordinarily primitve at recreating the perfect sound reproduction. Is this not a factor common to all of us with interests in audio. Every advancement, whether in digital mastering or new electronic topologies, must be embraced at all expense. Only if to hasten the process for which technologies can trickle down to an affordable level as soon as possible - so we can all experience someday before our short time here expires.

 Professional racing car drivers race in formula one cars. As an end consumer I still dream of experiencing the thrill of driving these professional devices even though they far surpass the requirements f daily transportation. Should the technology arise for 2000 bit audio with 50 million ghz sampling rates become feasible though extraordinarily expensive - I beleive I would lust after such a device that hopefully brings me closer to reality... or I can just go fishing.


----------



## Publius

Quote:


  Originally Posted by *spanimal* /img/forum/go_quote.gif 
_Professional racing car drivers race in formula one cars. As an end consumer I still dream of experiencing the thrill of driving these professional devices even though they far surpass the requirements f daily transportation. Should the technology arise for 2000 bit audio with 50 million ghz sampling rates become feasible though extraordinarily expensive - I beleive I would lust after such a device that hopefully brings me closer to reality... or I can just go fishing.
	

	
	
		
		

		
		
	


	


_

 

I fully agree that existing sound reproduction technologies are inadequate for truly realistic reproduction, but...

 The whole bitdepth thing is a red herring. People have a natural tendency to look at the easiest numbers to manipulate and think they are the road to salvation. Just like Intel perpetuated the megahertz myth with Pentium 4, there's this myth that increasing the bitdepth and sampling rate brings one "closer to reality". But your own experience with Blu-Ray undercuts this idea that any kind of improvement is available, even for extravagant increases in bandwidth. Moreover, some engineers (such as Dan Lavry) seriously believe that increasing bit depth and sample rate to insane levels will lead to _inferior_ performance. 

 The theory of acoustics - and most especially, the notion of a sound field, and position sensing, HRTFs, etc - is entirely capable of explaining _what is happening_ when you listen to a river, or a rock concert, or a drum set, and why that sounds so much different than when the same environments are played back on speakers. It's an extremely hard problem, but there isn't anything _fundamentally_ unknown about it. What is unknown is the exact nature of the solution. 

 Whatever the solution is, it won't be easily encapsulated into two clean little numbers like bitdepth and sample rate. It's more likely to be a complex mixture of all sorts of things, including high-channel surround sound, room treatment, high power, etc. But it will never happen if there's never a market for it. This focus on high res is sucking interest and $$$ away from solutions that actually have a chance at making audio more realistic.


----------



## gregorio

Quote:


  Originally Posted by *spanimal* /img/forum/go_quote.gif 
_How then can one claim that our current methodologies of audio reproduction are sufficient when I am declaring that all our technologies and techniques are in fact extraordinarily primitve at recreating the perfect sound reproduction. Is this not a factor common to all of us with interests in audio. Every advancement, whether in digital mastering or new electronic topologies, must be embraced at all expense. Only if to hasten the process for which technologies can trickle down to an affordable level as soon as possible - so we can all experience someday before our short time here expires.

 Professional racing car drivers race in formula one cars. As an end consumer I still dream of experiencing the thrill of driving these professional devices even though they far surpass the requirements f daily transportation. Should the technology arise for 2000 bit audio with 50 million ghz sampling rates become feasible though extraordinarily expensive - I beleive I would lust after such a device that hopefully brings me closer to reality... or I can just go fishing.
	

	
	
		
		

		
		
	


	


_

 

I would agree there are many areas of audio which are a weak link and could be improved. Microphones, amps, speakers, headphones to name but four. ADCs could be improved as well but the digital audio formats themselves are already way beyond what is both necessary and possible. For some of the digital formats available today the most important improvement required would be a complete re-design of the human ear!

 The analogy of the formular one car is inadequate as a human being can drive an F1 car whereas a human being can't hear the full dynamic range which can be encoded in 16bit. To carry your analogy through though, you want to try taking the kids to school or to go shopping in an F1 car?

 2000bits will be ideal if you want to record the actual volume of a nuclear explosion at it's origin, I'm sure you'll be very pleased, at least for a few nanoseconds until you are vapourised by the sound!! You'd need speakers the size of Manhattan though! 180dB will kill but you want 6000dB, not very bright. In answer to your last sentence, it wouldn't bring you "closer to reality" but it would bring you right up close to your maker!

 Although you said you read the thread, I don't think you have or at least not understood it. I created this thread to show that we have already reached and exceeded the point where higher numbers are better, formats such as 24/192 exist simply as marketing gimicks to con people like you who erroneously believe bigger numbers are automatically better. These bigger numbers will not get you closer to reality but further away from it.

 G


----------



## thathertz

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I would agree there are many areas of audio which are a weak link and could be improved. Microphones, amps, speakers, headphones to name but four. ADCs could be improved as well but the digital audio formats themselves are already way beyond what is both necessary and possible. For some of the digital formats available today the most important improvement required would be a complete re-design of the human ear!

 The analogy of the formular one car is inadequate as a human being can drive an F1 car whereas a human being can't hear the full dynamic range which can be encoded in 16bit. To carry your analogy through though, you want to try taking the kids to school or to go shopping in an F1 car?

 2000bits will be ideal if you want to record the actual volume of a nuclear explosion at it's origin, I'm sure you'll be very pleased, at least for a few nanoseconds until you are vapourised by the sound!! You'd need speakers the size of Manhattan though! 180dB will kill but you want 6000dB, not very bright. In answer to your last sentence, it wouldn't bring you "closer to reality" but it would bring you right up close to your maker!

 Although you said you read the thread, I don't think you have or at least not understood it. I created this thread to show that we have already reached and exceeded the point where higher numbers are better, formats such as 24/192 exist simply as marketing gimicks to con people like you who erroneously believe bigger numbers are automatically better. These bigger numbers will not get you closer to reality but further away from it.

 G_

 

So the greatest improvements we'll hear in the future will not come from higher bit-depths and sampling frequencies but from the other end of the line: the speakers and the listening environment. 

 Or perhaps one day we'll be offered mods to our ears? 
	

	
	
		
		

		
		
	


	




 I can see the threads now: 

*Post here if SinglePower still have your ears.*

 Many thanks to all involved in this thread so far, it's been very interesting indeed.


----------



## uofmtiger

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_There probably won't be much of a difference. Certainly use 16bit and then provided your DAC handles native 88.2, I would use that too. If your DAC only outputs 24/192 then you may as well stick to that unless storage space is at a premium.

 G_

 

 Quote:


 There's more: does your DAC handle 88.2 with no further re-sampling? 
 

It can also handle 24/96 (it has two settings..24/96 and 24/192). I think I am going to go with 24/96 since it should suffice and save a little space.

 Thanks for the responses.


----------



## spanimal

Thanks for the mature reponses to my arguments. Gregorio, you have proven to me that you are of the few few sound engineers that understand the concept of audio fidelity and applaud you for your efforts in everyway. Without getting too technical it is hence safe for me to presume that indeed 44.1/16 is sufficient for existing technologies. My experiences have suggested to me that very well mastered CDs equals my 2 hi res discs, at the very least. In light of these positive responses, I must recind the aggressive nature of my previous post and hope that discussions like these do not get interfered with by immaturity.

 Nevertheless I still think it would be kinda cool to go grocery shopping in a formula one car and have a system able to playback a nuclear explosion. This would be the pinnacle of awesomeness!


----------



## Kees

Quote:


  Originally Posted by *spanimal* /img/forum/go_quote.gif 
_Thanks for the mature reponses to my arguments. Gregorio, you have proven to me that you are of the few few sound engineers that understand the concept of audio fidelity and applaud you for your efforts in everyway. Without getting too technical it is hence safe for me to presume that indeed 44.1/16 is sufficient for existing technologies. My experiences have suggested to me that very well mastered CDs equals my 2 hi res discs, at the very least. In light of these positive responses, I must recind the aggressive nature of my previous post and hope that discussions like these do not get interfered with by immaturity.

 Nevertheless I still think it would be kinda cool to go grocery shopping in a formula one car and have a system able to playback a nuclear explosion. This would be the pinnacle of awesomeness!_

 

Without the fall-out please. Thanks.


----------



## spanimal

What fall out? My post was designed as bait to expose any ignoramus here. It did not, so I am quite satisfied with the intelligence of the posters in this thread and respect the opinions of the posters. There is reason for me to believe that a higher resolution format will yield sonic benefits - just as blue ray is clearly superior to regular dvd, however my listening preferences are top 40 and there just is not any hi res music in this genre. My in depth knowledge of digital music is somewhat lacking, so out of respect for the very knowledgeable people here, I reposition my stance to argue that when regular format is mastered well it is indeed very acceptable. Nevertheless, it would be cool to be able to go grocery shopping in a formula one car.


----------



## Kees

Quote:


  Originally Posted by *spanimal* /img/forum/go_quote.gif 
_What fall out? My post was designed as bait to expose any ignoramus here. It did not, so I am quite satisfied with the intelligence of the posters in this thread and respect the opinions of the posters. There is reason for me to believe that a higher resolution format will yield sonic benefits - just as blue ray is clearly superior to regular dvd, however my listening preferences are top 40 and there just is not any hi res music in this genre. My in depth knowledge of digital music is somewhat lacking, so out of respect for the very knowledgeable people here, I reposition my stance to argue that when regular format is mastered well it is indeed very acceptable. Nevertheless, it would be cool to be able to go grocery shopping in a formula one car._

 

Nuclear explosion. Fall-out. Get it?
 The theoretical knowledge on this subject in this thread is rather lacking, which makes all the "expert" explanations a bit misplaced IMO.


----------



## Arjisme

Quote:


  Originally Posted by *Kees* /img/forum/go_quote.gif 
_The theoretical knowledge on this subject in this thread is rather lacking, which makes all the "expert" explanations a bit misplaced IMO._

 

Please don't hold back. If you have something of value to add, please do so. What is this missing theoretical knowledge?


----------



## gregorio

Quote:


  Originally Posted by *thathertz* /img/forum/go_quote.gif 
_So the greatest improvements we'll hear in the future will not come from higher bit-depths and sampling frequencies but from the other end of the line: the speakers and the listening environment. 

 Or perhaps one day we'll be offered mods to our ears? 
	

	
	
		
		

		
			



_

 

I believe there are improvements to be made with just about all of the technology. Even in ADCs and DACs, there's the actual mathematical algorithms used, which include the filters, decimation and other processes but improvements in these areas are likely to be quite minimal (with regard to perceived quality) and require no changes in the digital formats themselves.

  Quote:


  Originally Posted by *spanimal* /img/forum/go_quote.gif 
_Without getting too technical it is hence safe for me to presume that indeed 44.1/16 is sufficient for existing technologies. My experiences have suggested to me that very well mastered CDs equals my 2 hi res discs, at the very least. In light of these positive responses, I must recind the aggressive nature of my previous post and hope that discussions like these do not get interfered with by immaturity._

 

I am not willing to say absolutely that you will never hear any benefit to higher sample rates. If there are audiophiles out there who believe they can hear a difference between different power cables or even fuses, then I'm certain there will be some who are convinced they can hear a difference between 16/44.1 and 24/96. In reality though the tiny differences in sample rates are incredibly unlikely to result in any real perceivable difference. Compared to these minute differences, the differences you are likely to notice between a really well recorded and mixed album and a poorly recorded and produced album are absolutely massive. So, if you are seriously interested in SQ, you need to be making note of the top studios and producers, who are the most likely to be creating the highest quality masters, rather than looking for higher rates of data.

 G


----------



## leeperry

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I believe there are improvements to be made with just about all of the technology. Even in ADCs and DACs, there's the actual mathematical algorithms used, which include the filters, decimation and other processes but improvements in these areas are likely to be quite minimal (with regard to perceived quality) and require no changes in the digital formats themselves._

 

apparently the AK4396 is where the money is, it does all kind of filtering...and everyone agrees that it's the "miracle DAC".
  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I am not willing to say absolutely that you will never hear any benefit to higher sample rates._

 

I think the real issue is that most consumer soundcards resample...so you never really get full bit-perfect audio.

 like the Asus Essence STX, that has a software based headphones output volume control, and a fixed masterclock...so everything will be resampled to this sample rate 
	

	
	
		
		

		
		
	


	




 and some soundcards are even more sneaky, and won't let you change the masterclock.






 so if your fixed masterclock is set to 96KHz(like on my M-Audio Audiophile USB), then 24/96 will be a drastic improvement over 16/44.1


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_so if your fixed masterclock is set to 96KHz(like on my M-Audio Audiophile USB), then 24/96 will be a drastic improvement over 16/44.1 
	

	
	
		
		

		
		
	


	


_

 

Only if you believe the marketing! 
	

	
	
		
		

		
		
	


	




 If it's a really good DAC then you won't hear a difference, if it's a really poor DAC then you might, possibly.

 G


----------



## leeperry

or if it's a p*ss poor SRC to match the masterclock


----------



## qusp

yes, dithering is never a good thing; no matter how well implemented. occasionally with dithering 1+1=3 i'm lucky enough to be able to set my master clock in teeny tiny increments to essentially any rate I wish. with the RME unless I choose re sampling if I change the sample rate or clock the music gets quicker/slower. great for syncing with old analogue synths and such


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_or if it's a p*ss poor SRC to match the masterclock 
	

	
	
		
		

		
		
	


	


_

 

I find this re-clocking idea to be rather strange. The only way re-clocking could have any advantage is if the masterclock in the DAC is better than the masterclock in the ADC during the original recording (A standard PLL will solve any jitter introduced during transport). So if the original recording was done on a relatively cheap (poor) quality ADC there may be some fractional improvement but if the original recording was done on a very good quality ADC then it's more than likely that the masterclock in the DAC is actually going to be less accurate and therefore, at least in theory, produce poorer quality audio. Many good studios use dedicated masterclocks to synchronise their ADCs and other digital equipment. In my studio I have a masterclock which cost over $1,000, it's very unlikely that any DAC on the market is going to have a masterclock more accurate than what is already encoded in the digital audio data stream.

  Quote:


  Originally Posted by *qusp* /img/forum/go_quote.gif 
_yes, dithering is never a good thing; no matter how well implemented. occasionally with dithering 1+1=3_

 

That's strange, I would say that whenever a quantisation or re-quantisation process occurs that dither is not only a good thing but a vital thing.

 G


----------



## spanimal

WELL... mission accomplished. I have finally exposed the ignoramus on this thread. That ignoramus is........ME!!! Fallout - right, I was thinking hard about what exactly my man Kees meant by fallout. For all my big words, man, I am kinda slow.

 I have gone back to compare the SACD and CD version of Norah Jones, I hear differences, possibly due to different circuitry, but cannot establish which version is "better". I listened again to my Metallica DVD-A and noted it sounds very nice but cannot establish that it is "better" than other CD sources.

 Guys, I have a quick off the topic question. During the course of comparing today - I have accidently discovered that Windows media player had a louder volume at max than my Logitech Medialife player (Remote control friendly). Then I remembered something about bit-perfect and listened carefully and windows media player had a significant sound quality improvent. Is windows media player bit-perfect?


----------



## leeperry

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I find this re-clocking idea to be rather strange._

 

I think you misunderstood me gregorio....you've never really used a crappy consumer soundcard 
	

	
	
		
		

		
		
	


	




 the X-Fi, the Asus Essence STX, the M-Audio consumer range....they will all FORCE you to choose a locked masterclock, as you can see here : http://www.image-load.eu/out.php/i159037_sample.png

 so if you leave it on 192KHz(it's more so it's got to be better!), then ALL your audio will be resampled...but if you leave it on 96KHz, all your 44.1 stuff will be resampled too 
	

	
	
		
		

		
		
	


	




 and who wants to change it constantly whether he listens to CDDA/DVD/DVD-A? 
	

	
	
		
		

		
		
	


	




 my M-Audio Audiophile USB sounds lousy w/ CDDA stuff because the masterclock is indeed 96Khz.

 I've just ordered a Gina soundcard w/ bit-perfect WDM drivers and automatic masterclock(like on the RME)....this thing doesn't resample AT ALL(if you force the masterclock to 96Khz and play a CDDA, it will indeed play twice too fast)

 hopefully my CDDA stuff will not sound distorted as it does on the Audiophile USB atm.

 and these are prosumer soundcards, real consumer soundcards use an arbitraty masterclock and won't even let you change it manually 
	

	
	
		
		

		
		
	


	




 most ppl listen to music through KMixer, so who cares after all


----------



## Acix

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_I think you misunderstood me gregorio....you've never really used a crappy consumer soundcard 
	

	
	
		
		

		
		
	


	




 the X-Fi, the Asus Essence STX, the M-Audio consumer range....they will all FORCE you to choose a locked masterclock, as you can see here : http://www.image-load.eu/out.php/i159037_sample.png

 so if you leave it on 192KHz(it's more so it's got to be better!), then ALL your audio will be resampled...but if you leave it on 96KHz, all your 44.1 stuff will be resampled too 
	

	
	
		
		

		
		
	


	




 and who wants to change it constantly whether he listens to CDDA/DVD/DVD-A? 
	

	
	
		
		

		
		
	


	




 my M-Audio Audiophile USB sounds lousy w/ CDDA stuff because the masterclock is indeed 96Khz.

 I've just ordered a Gina soundcard w/ bit-perfect WDM drivers and automatic masterclock(like on the RME)....this thing doesn't resample AT ALL(if you force the masterclock to 96Khz and play a CDDA, it will indeed play twice too fast)

 hopefully my CDDA stuff will not sound distorted as it does on the Audiophile USB atm.

 and these are prosumer soundcards, real consumer soundcards use an arbitraty masterclock and won't even let you change it manually 
	

	
	
		
		

		
		
	


	




 most ppl listen to music through KMixer, so who cares after all 
	

	
	
		
		

		
		
	


	


_

 

Why not the RME?


----------



## spanimal

Leeperry - very interesting observations you made with the output rates of soundcards. I had checked the incoming signal of my soundcards digital output to my receiver (I do not use its dac, but it will let me know the sampling rate of an incoming signal) and it always showed 48khz - there was no option to change it in anyway. The CD players digital output showed 44.1. I stayed well away from a computer source until I worked out my options.

 I bought the Cambridge Audio Dacmagic about a month ago so I can now take advantage of lossless files and the convenience of storage and use the PC as a serious audio source. When I first hooked up the dacmagic with its standard USB cable it indicated 48khz and I was very worried. I played lossless through Logitech software and the incoming signal indicators immediately switched to 44.1khz - I was so relieved and presumed that I was getting a perfect CD signal. BUT yesterday I found Window media player volume to be louder and presumed that if difference in volume is detected then these two programmes are giving me different bits (All available volume options at MAX - though all indicated 44.1khz. The sound of Windows media player is very much noticeable as an improvement up and over the logitech when volume is taken into consideration. Neverthless - I am getting anxious and paranoid about not getting the original CD data to the DAC. What is my course of action? Is Foobar the answer? - I will download this and install.


----------



## pompon

My XMeridian soundcard have an option "Allows application to take control of this device".

 In fact, foobar will set the clock according of the resolution of my songs.

 When you using asio, wasapi or kernel streaming, you don't have the windows mixer (resampled) nasty to care with.


----------



## qusp

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I find this re-clocking idea to be rather strange. The only way re-clocking could have any advantage is if the masterclock in the DAC is better than the masterclock in the ADC during the original recording (A standard PLL will solve any jitter introduced during transport). So if the original recording was done on a relatively cheap (poor) quality ADC there may be some fractional improvement but if the original recording was done on a very good quality ADC then it's more than likely that the masterclock in the DAC is actually going to be less accurate and therefore, at least in theory, produce poorer quality audio. Many good studios use dedicated masterclocks to synchronise their ADCs and other digital equipment. In my studio I have a masterclock which cost over $1,000, it's very unlikely that any DAC on the market is going to have a masterclock more accurate than what is already encoded in the digital audio data stream. 



*That's strange, I would say that whenever a quantisation or re-quantisation process occurs that dither is not only a good thing but a vital thing.*

 G_

 

bit perfect audio + dither = fail. I try to at all costs avoid REclocking at anything but the original sample rate, but most of the time I dont even need to do this.
 I have also used such clocks as the big ben and while the RME doesnt come close; it does a very good job. of course if the sample rate is changed you have to dither (unless its a direct multiple like 88.2, which I suppose still qualifies as dithering, but does not act upon the signal or create the artifacts that I dislike (I use this sometimes in my rig) but I try to avoid such processes.

 when quantizing unless you change the rate and resample dithering is not needed nor desired. the only times I will use dithering or reclocking are when I have multiple sources that I am mixing together in logic; in which case the RME is set as master and it does a good job of syncing outputs from plugins and various sources. but with 2 channel audio. no way; you are correct a decent recording will have a quality clock that can be recovered. and imposed on the signal


----------



## leeperry

Quote:


  Originally Posted by *pompon* /img/forum/go_quote.gif 
_My XMeridian soundcard have an option "Allows application to take control of this device".

 In fact, foobar will set the clock according of the resolution of my songs.

 When you using asio, wasapi or kernel streaming, you don't have the windows mixer (resampled) nasty to care with._

 

well I didn't pay too much attention to the X-Meridian cards, mostly because the don't have an external rack w/ a headphones output and a volume knob(software based volume control kills the dynamics too!)

 but the X-Meridian card in this Japanese SNR/THD test failed big time :
DOS/V POWER REPORT | Impress Japan
DOS/V POWER REPORT | Impress Japan
DOS/V POWER REPORT | Impress Japan
DOS/V POWER REPORT | Impress Japan
DOS/V POWER REPORT | Impress Japan
DOS/V POWER REPORT | Impress Japan
DOS/V POWER REPORT | Impress Japan
DOS/V POWER REPORT | Impress Japan

 and foobar/ASIO won't necessarily save you. my M-Audio still has a fixed masterclock of 96KHz in ASIO(actually its ASIO drivers only allow 48/88.2/96Khz), and the Asus Essence STX still forces you to choose a masterclock.

 mostly that's because it's better for end-users, if they play several streams at the same time, the soundcard drivers will all play them at the right pitch!
  Quote:


  Originally Posted by *Acix* /img/forum/go_quote.gif 
_Why not the RME? 
	

	
	
		
		

		
			



_

 

well I wanted an external rack w/ a headphones output and a knob, and the Echo can be bought for really cheap second hand 
	

	
	
		
		

		
		
	


	



  Quote:


  Originally Posted by *spanimal* /img/forum/go_quote.gif 
_I had checked the incoming signal of my soundcards digital output to my receiver
 [...]
 I am getting anxious and paranoid about not getting the original CD data to the DAC. What is my course of action? Is Foobar the answer? - I will download this and install._

 

I don't really know 
	

	
	
		
		

		
		
	


	



 most manufacturers lie about all this....what's your soundcard?
 get a real professional soundcard, and you'll get true bit-perfect stuff(Lynx/RME/Echo/and some others I'm sure)
 but to truly compare 16/44.1 and 24/96 you need truly bitperfect outputs...


----------



## DistortingJack

Quote:


  Originally Posted by *spanimal* /img/forum/go_quote.gif 
_ yesterday I found Window media player volume to be louder and presumed that if difference in volume is detected then these two programmes are giving me different bits (All available volume options at MAX - though all indicated 44.1khz. The sound of Windows media player is very much noticeable as an improvement up and over the logitech when volume is taken into consideration. Neverthless - I am getting anxious and paranoid about not getting the original CD data to the DAC. What is my course of action? Is Foobar the answer? - I will download this and install._

 

Foobar is the answer. WMP is a horrible piece of software, ad iTunes is not a lot better.
 I think your problem is the set of "enhancements" WMP has set as default. There's WOW sounds, EQ, among other ***** lurking around in the sound preferences. Take them all off.
 And install foobar.


----------



## leeperry

well I've found some bit-perfect drivers for my USB soundcard : USB 2 Audio - the low latency experience

 they replace the manufacturer's drivers, and you can actually set the masterclock to whatever you like.....and w/ 44.1(that the M-Audio drivers did NOT allow, not even in ASIO), my CDDA's sound way better 
	

	
	
		
		

		
		
	


	




 too bad you have to select 44.1/48/96 manually(like on the Asus Essence STX), that's a show stopper for me


----------



## pompon

I have X-Meridian 7.1 soundcard ... not the Auzentech X-Plosion.
 The X-Meridian use a true circuit to control the output power ... like the Xonar Essence STX. 

 No matter what numbers said ... that said nothing about "is-it good or not". Vinyl have 55 db or dynamic, channel separation is not good ... but I want listening all the album. A simple DVD player have good specs ... but after 10 min, I power off the unit and go listening TV because I am not in the music at all.


----------



## leeperry

Quote:


  Originally Posted by *DistortingJack* /img/forum/go_quote.gif 
_Foobar is the answer. WMP is a horrible piece of software_

 

funny, I personally think that Foobar is an horrible piece of software 
	

	
	
		
		

		
		
	


	




 -takes 1 or 2 seconds each time I open a file in ASIO to initialize? 
 -requires to close/open foobar if I play a file w/ a different sample rate(ASIO again?)
 -supports VST plugins, but won't remember the settings you put there...so it's pointless if you use an EQ plugin
 -VST plugin work in 16bit only, no 32FP for you 
	

	
	
		
		

		
		
	


	



 -can't choose your own coeffs for 5.1>stereo downmix

 MPC+ffdshow do all this, and even more! and even xmplay works better than foobar, it can open FLAC/APE, supports WASAPI/ASIO(w/o any opening delay!) and can force the masterclock


----------



## Ham Sandwich

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_funny, I personally think that Foobar is an horrible piece of software 
	

	
	
		
		

		
		
	


	




 -takes 1 or 2 seconds each time I open a file in ASIO to initialize? 
 -requires to close/open foobar if I play a file w/ a different sample rate(ASIO again?)_

 

My Foobar doesn't behave like that. If you're having behavior like that it's an audio hardware or audio driver issue. I'm able to play different sample rates in Foobar (or J. River Media Center) and have my M-Audio FW410 sync up a different sample rate using either ASIO or WASAPI.

 The suggestion by DistortingJack is valid. Spanimal needs to try a different media player that is able to play clean and bit-perfect. Foobar does that and is free. J. River Media Jukebox (or J. River Media Center) would also fit. But Foobar is good enough for a listening test to see if the sound is different.


----------



## leeperry

Quote:


  Originally Posted by *Ham Sandwich* /img/forum/go_quote.gif 
_My Foobar doesn't behave like that. If you're having behavior like that it's an audio hardware or audio driver issue. I'm able to play different sample rates in Foobar (or J. River Media Center) and have my M-Audio FW410 sync up a different sample rate using either ASIO or WASAPI.

 Foobar does that and is free._

 

if I click on a file in the playlist, everything's cool. but if I right click on a file and choose "open w/ foobar", I get this for 1 or 2 secs :




 none of this happens w/ xmplay, which is also freeware, supports ASIO/WASAPI and forces the soundcard masterclock to behave 
	

	
	
		
		

		
		
	


	




 plus the skins are too awesome : XMPlay Support


----------



## spanimal

The first thing I checked on Windows media player is if it had DSP's selected - no it had no dsp's even installed or selectable. I have downloaded foobar and cannot distinguish between windows player or foobar - even the volume was identical - I guess I'll just stick with foobar to be safe. On a sidenote though the input to the dacmagic is 44.1 from the computer - this dac resamples everything to 24/192 internally before out putting to analog. A lot of high end dacs do this - surely it cannot be such a bad thing (directed at leeperry).


----------



## leeperry

yeah, some ppl say the sound is better if you resample 16/44.1 to 24/192, there's even a guy w/ a lenghty PDF tutorial about that.

 IMHO, you cannot create data that's not there...you can only worsen the THD and create aliasing/distortion 
	

	
	
		
		

		
		
	


	




 honestly resampling w/ SRC/libavcodec makes the sound noticably less "accurate", more "noisy" and "brighter"


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_yeah, some ppl say the sound is better if you resample 16/44.1 to 24/192, there's even a guy w/ a lenghty PDF tutorial about that.

 IMHO, you cannot create data that's not there...you can only worsen the THD and create aliasing/distortion 
	

	
	
		
		

		
		
	


	




 honestly resampling w/ SRC/libavcodec makes the sound noticably less "accurate", more "noisy" and "brighter"_

 

It's difficult to comment on this. In theory, re-sampling from say 16/44.1 to 24/192 can do nothing except worsen the sound quality. Any quantisation errors and filtering artifacts of 16/44.1 are already part of the encoded data-stream. Increasing the sample rate cannot magically replace the frequencies filtered out or any other effects of the filter or conversion process. In practice, this up-sampling may end up sounding slightly better or slightly worse, depending on the DAC and how the reconstruction filters have been implemented.

 So if you have a DAC where you can switch sample rates, do a few tests and see what sounds better on your system. All else being equal, I would advise you to play the audio back in the format in which it was distributed but it is possible that you have a DAC where virtually all the effort has been put into the design of up-sampling and that playing it back at original sample rate doesn't sound quite as good. This scenario is unlikely but not impossible. If your DAC has fixed up-sampling, then this is proof that all the design effort has gone into this process, to the point that other options don't even exist. This obviously saves design and manufacturing costs.

 The vast majority of listeners though are not going to be able to hear the differences in any of this!! The level of differences we're talking about here are marginal at best. Mostly, it's measurable and operational differences rather than audible differences.

 G


----------



## JaZZ

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_IMHO, you cannot create data that's not there..._

 

That's not exactly true. Actually oversampling/upsampling is there to create data that aren't there anymore: the sampling points between the sampling points. Of course it's just estimated data. But apart from the most simple method (linear interpolation), what most modern DACs do -- interpolation by sinc-function algorithms -- is good enough for a reconstruction corresponding to a low-pass filter at around 22 kHz at the latest. Which is exactly what a conventional analog filter does -- or has done in earlier-generation CDPs. So the only function of oversampling/upsampling is low-pass filtering, in co-operation with a serial analog filter necessary for smoothing the residual stair steps. 

 I don't know exactly what's going on in «resampling» DACs or sound cards, resp. I imagine it's possibly (uninteger) upsampling in addition to a filter design already consisting of oversampling (= the «digital filter») and analogue filtering. Which shows how obsolete the upsampling is in this case. Unless it (alternatively) replaces a purely analogue filter system with itself and a simpler analogue filter circuit. In the worst case it's just forced sample-rate conversion. 
.


----------



## leeperry

well, is there any of these SRC "magic" algorithms on PC? ppl keep talking about them, but it's only to be seen in 4 figures external DAC's..

http://photos.imageevent.com/cics/v0...rts%20v0.3.pdf





 this is a fantastic guide, I've learned a lot from it!

 but I've heard the secret rabbit code, it's in ffdshow and Reclock....and even in "highest quality" mode it sounds really bad. it kills the soundstage/dynamics range and makes the sound bright and agressive....it's a slaughter


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_http://photos.imageevent.com/cics/v0...rts%20v0.3.pdf

 this is a fantastic guide, I've learned a lot from it!_

 

I haven't read it all but there is a lot of useful information. Unfortunately, there are also a quite a few completely off the wall statements such as "SPDIF (and other proprietary variants) is a lossy unintelligent legacy interface". SPDIF can easily be bit perfect so I don't know where the "lossy" idea comes from. And, although accurately mentioning Nyquist Theorem he then states "Clearly at 24/192 we get more detail at peaks and troughs bringing us closer to the original analogue waveform." This statement is true but largely irrelevant as it doesn't really matter how closely the digital data can be represented graphically, it's what comes out of the DAC which is important and more sampling points beyond the Nyquist limit does not necessarily mean a more accurate reproduction of the original waveform. These are just two of quite a number of inaccuracies I noticed.

  Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Actually oversampling/upsampling is there to create data that aren't there anymore: the sampling points between the sampling points._

 

Yes and no. The "sampling points between the sampling points" are interpolated exactly the same way as would be with the lower sample rate. So although there is more data, it can't really be described as new. Any artifacts of the original ADC process (including the anti-alias filter) will still be present and there will still not be any data above the original Nyquist limit.

 G


----------



## JaZZ

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Yes and no. The "sampling points between the sampling points" are interpolated exactly the same way as would be with the lower sample rate. So although there is more data, it can't really be described as new. Any artifacts of the original ADC process (including the anti-alias filter) will still be present and there will still not be any data above the original Nyquist limit._

 

Yeah, sure. 
	

	
	
		
		

		
			





  Quote:


  Originally Posted by *JaZZ* /img/forum/go_quote.gif 
_Of course it's just estimated data. But apart from the most simple method (linear interpolation), what most modern DACs do -- interpolation by sinc-function algorithms -- is good enough for a reconstruction corresponding to a low-pass filter at around 22 kHz at the latest. Which is exactly what a conventional analog filter does -- or has done in earlier-generation CDPs. So the only function of oversampling/upsampling is low-pass filtering, in co-operation with a serial analog filter necessary for smoothing the residual stair steps._

 

Thanks for your contribution! 
	

	
	
		
		

		
		
	


	



.


----------



## HeadLover

I used to think that 24/96 is better, but NO more !!

 I have like tested many many 16/44 with 24/96
 I have the DAC1 PRE that can accept USB input of up to 24/96 with BIT PERFECT (using WASAPI and foobar2000)
 And I do believe that 16/44 is sometimes even better, maybe becuase less bits need to be transfered over the USB line, so the CPU can work less and maybe even the DAC need to work less and the SQ is better.


----------



## mbd2884

I think the real answer is with the mofos who love their SACD players is that the SACD player has better laser technology and better D/As. It's not the 24/96, but the parts are better in SACD players than in redbook CD players.

 That's it, so simple. Solution? Computers to the rescue with external DAC!


----------



## Ashirgo

And we can easily assume that SACD material may be mastered in a different way (for audiophiles) and it is why people find SACD preferable to CD. But as it has been already written here, I just support that idea.


----------



## JaZZ

SACD sounds different from CD -- on the same player. Not necessarily better (particularly when based on a low-rez recording!), but there's definitely a characteristic sonic difference. And that's true for different players with which I've done the comparison. Now I'm not sure if that speaks for DSD.

 DVD-A on the other hand sounds more similar to redbook CD.

 (To my ears)
.


----------



## nick_charles

...


----------



## dex85

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_but I've heard the secret rabbit code, it's in ffdshow and Reclock....and even in "highest quality" mode it sounds really bad. it kills the soundstage/dynamics range and makes the sound bright and agressive....it's a slaughter 
	

	
	
		
		

		
		
	


	


_

 

correct me if i'm wrong but asio and others forbid os and therefore all audio codecs to interfere with song data. it goes straigh to soundcard/external DAC as raw data. so how ffdshow or any other codecs could affect your music?


----------



## spanimal

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_but to truly compare 16/44.1 and 24/96 you need truly bitperfect outputs..._

 

You have something there with your loathing of upsampling. My dacmagic is fixed at 24 192, however there is an option to roll off all the frequencies above 22khz - this would be truer to the original CD. It then focuses all its electronics on developing the data below this point instead of wasting it on artificial data created above this point. Guess what, it sounds much better, It removes what I believe is artificial "air" and "extension" in the treble creating an overall fuller and more stable sound with better pratt, with the benefit of the upsampling to reduce jitter.


----------



## leeperry

Quote:


  Originally Posted by *dex85* /img/forum/go_quote.gif 
_correct me if i'm wrong but asio and others forbid os and therefore all audio codecs to interfere with song data. it goes straigh to soundcard/external DAC as raw data. so how ffdshow or any other codecs could affect your music?_

 

you can use ffdshow in 32FP in any DirectShow player, and go through Reclock in KS 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *spanimal* /img/forum/go_quote.gif 
_You have something there with your loathing of upsampling. My dacmagic is fixed at 24 192, however there is an option to roll off all the frequencies above 22khz - this would be truer to the original CD. It then focuses all its electronics on developing the data below this point instead of wasting it on artificial data created above this point. Guess what, it sounds much better, It removes what I believe is artificial "air" and "extension" in the treble creating an overall fuller and more stable sound with better pratt, with the benefit of the upsampling to reduce jitter._

 

resampling is evil...and my soon-to-be-arrived next soundcard(Asus Essence STX) resamples anything that's <96KHz : http://www.image-load.eu/out.php/i160484_wtf.png

 it even resamples ASIO! sucks to be me I guess


----------



## dex85

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_you can use ffdshow in 32FP in any DirectShow player, and go through Reclock in KS 
	

	
	
		
		

		
		
	


	


_

 

yes, but why would you do that with music playback? i just use player with asio support and don't worry about windows and codecs.


  Quote:


 resampling is evil...and my soon-to-be-arrived next soundcard(Asus Essence STX) resamples anything that's <96KHz 
 

so if i select 44.1 kHz sample rate in drivers it will resample
 44.1 kHz songs anyway?


----------



## leeperry

Quote:


  Originally Posted by *dex85* /img/forum/go_quote.gif 
_yes, but why would you do that with music playback? i just use player with asio support and don't worry about windows and codecs._

 

coz for music, I use a VST EQ to follow this tutorial : http://www.head-fi.org/forums/f4/how...torial-413900/

 and for movies I use the Lexicon Logic7 5.1>binaural stereo downmix matrix + Ozone4 to get a crazy 3D-ish sound w/ headphones 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *dex85* /img/forum/go_quote.gif 
_so if i select 44.1 kHz sample rate in drivers it will resample 44.1 kHz songs anyway?_

 

yes, the damn thing resamples even ASIO and KS if the freq <96KHz...shame! 
	

	
	
		
		

		
		
	


	




 I'm b*tching at the Asus techsupport at this point 
	

	
	
		
		

		
		
	


	




 some ppl say perfect resampling is inaudible, and that this soundcard just sounds too darn good!

 we'll see about that, I'm already very impressed by my M-Audio Audiophile USB tbh...in foobar w/ ASIO and 32FP VST, it sounds -really- good(same THD/SNR results as the 0404USB and Wavio USB) 
	

	
	
		
		

		
		
	


	




 and it's got 3*NJM5532 on the headphones out and a pretty old AKM DAC...so the Asus will be an improvement(using LME49720), even w/ the DSP/SRC crapola on top of it


----------



## pila405

And thats why DSD encoding is considered being the best. [right?]

 Here is a link:
Direct Stream Digital - Wikipedia, the free encyclopedia


----------



## pompon

Headroom just do few samples to give an idea of the general graphs ... You can't use that to make a "correction". 

 Like .. I do 20 samples with tones in my room to try to do a "correct room" frequency graph.  You need A LOT MORE resolution. 

 If you using stereo ... the tricks is ALL EFFECT, DSP, HEADPHONE DOLBY ... O F F ...


----------



## spanimal

Quote:


  Originally Posted by *mbd2884* /img/forum/go_quote.gif 
_I think the real answer is with the mofos who love their SACD players is that the SACD player has better laser technology and better D/As. It's not the 24/96, but the parts are better in SACD players than in redbook CD players.

 That's it, so simple. Solution? Computers to the rescue with external DAC!_

 

Masturbation Generation - I shall say no more regarding this excessive masturbator. PM me for a personal meetimg to solve our differences. Come prepared.


----------



## b0dhi

If dithering a 16bit sound causes an audible change then obviously the least significant bit in 16bit sound must be audible, since dither usually only alters that bit. 

 24bit sampling, assuming it has real-world resolution beyond 16bits, will quantise the 16th bit based on actual sample data and not by noise. Since the 16th bit is audible, and since good 24bit quantisation has set the 16th-LSB by sample data whereas 16bit with dither has set the 16th LSB by noise, the two must sound different.

 It's either one or the other - either 16bit and 24bit sound the same and dither is inaudible, or dither is audible and 16bit and 24bit can sound different.


----------



## JaZZ

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_If dithering a 16bit sound causes an audible change then obviously the least significant bit in 16bit sound must be audible, since dither usually only alters that bit. 

 24bit sampling, assuming it has real-world resolution beyond 16bits, will quantise the 16th bit based on actual sample data and not by noise. Since the 16th bit is audible, and since good 24bit quantisation has set the 16th-LSB by sample data whereas 16bit with dither has set the 16th LSB by noise, the two must sound different.

 It's either one or the other - either 16bit and 24bit sound the same and dither is inaudible, or dither is audible and 16bit and 24bit can sound different._

 

Interesting thoughts!
.


----------



## gregorio

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_If dithering a 16bit sound causes an audible change then obviously the least significant bit in 16bit sound must be audible, since dither usually only alters that bit. 

 24bit sampling, assuming it has real-world resolution beyond 16bits, will quantise the 16th bit based on actual sample data and not by noise. Since the 16th bit is audible, and since good 24bit quantisation has set the 16th-LSB by sample data whereas 16bit with dither has set the 16th LSB by noise, the two must sound different.

 It's either one or the other - either 16bit and 24bit sound the same and dither is inaudible, or dither is audible and 16bit and 24bit can sound different._

 

Good question. But it's not quite that simple, you are applying analogue terms (like noise) to what is a purely mathematical process. The dithering process happens in the digital domain, we talk about dither being the addition of noise but of course in reality it's actually a set of mathematical algorithms. These algorithms cause randomisation and randomisation would be perceived as white noise and at 16bit this white noise would be down at -96dB. So far so good and your assertion would be true that there would be nothing but dither noise beyond -96dB, whereas theoretically you can encode actual signal below -96dB with 24bit. However, there are two points to bear in mind:

 1. Even an extremely dynamic recording will not have a dynamic range much in excess of 60dB, so everything below -60dB is pretty much noise anyway, so the -96dB dither noise is only going to add slightly to the noise floor which is already part of the recording.

 2. Going on from my first paragraph - The description I gave is how dither was applied when higher than 16bit recording first became available in the early '90s. However in the mid '90s noise shaping became available and has been extensively developed since then. The same process as above is carried out but as part of the mathematical process the resultant set of randomised results (dither noise) is redistributed to insensitive areas of the hearing spectrum. So with noise shaping you still have exactly the same amount of noise but now it is much less noticeable because it is concentrated say above 16kHz (the exact range of frequencies where the dither noise is moved depends on user settings in the dither software interface). This would result in quite a lot of dither noise centred in a frequency band where most people won't have any sensitivity to it, whereas the frequency band humans are most sensitive to, say 250Hz - 6kHz is almost completely without dither noise. Some experts claim that within this critical frequency band modern noise shaping results in a perceptual dynamic range with 16bit of up to 150dB (which is actually beyond 24bit resolution). So a good mastering engineer will choose a noise shaping algorithm which redistributes the vast majority of the dither noise but doesn't load up >16Khz with too much noise which may cause a slight muddiness (or lack of "space") in the very high frequencies for those few who may be sensitive to it.

 So in reality it would be more accurate to say that noise is added equivalent to the 16th bit level but (depending on the dither algorithm used) that doesn't mean that if your turn your amp up so you can hear what is going on at the 16th bit (-96dB) that you will actually hear any dither noise. Chances are that you will hear significant noise at that level but it will be the physical noise floor (as opposed to the digital noise floor) of the recording and your system. 

 In practise modern noise shaping lowers the noise floor (in the critical hearing band) to the point which can't be resolved by even a world class sound system. So in a real world situation, a correctly chosen and applied noise shaped dithered 16bit file will measure slightly differently from an original 24bit file but the differences will not be audible. This statement is true at any normal listening level, even for an extreme audiophile!

 G


----------



## Publius

Yeah b0dhi, the issue is not black and white. Nobody's disputing that situations exist where 24 bits can be ABX'd against 16 bits. That's trivial to do. But it's extremely difficult - possibly impossible - to ABX the difference involving music playback not specifically designed to exaggerate the differences. ie, playing back extremely attenuated music at extremely high volumes will easily show off the difference, but such situations simply don't seem to show up in commercially released music.


----------



## HeadLover

I still can't understand
 Apart from size of the files
 Why not using a 24bit on 96KHZ audio files ?!

 It won't be "less good", so why not ???


----------



## gregorio

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I still can't understand
 Apart from size of the files
 Why not using a 24bit on 96KHZ audio files ?!

 It won't be "less good", so why not ???_

 

No reason at all but it's a matter of principle. There is no benefit to the consumer of 24/96 but that is not the impression given in the marketing by companies who manufacture consumer equipment. Even the terminology itself changed when the consumer manufacturers got involved, we use to call it "higher than 16bit recording" it's now known as "high resolution", giving the impression that it's better than standard resolution. The result is that consumers are conditioned to pay more for a "high resolution" DAC, audio file, soundcard or transport even though they could get better SQ by buying a higher quality (but similarly priced) 16/44.1 DAC, soundcard or whatever. The whole thing has become circular with some areas of the music industry realising they can charge more for their music if it's in 24/96 and therefore mastering it to a higher quality to further convince consumers that 24/96 is better than CD!

 What should have happened is that the quality of 16/44.1 mastering should be to a higher standard and then there wouldn't be any audible difference or any demand for "Hi-Rez" but there's relatively little profit in that as consumers would not have to go out and buy a whole new set of equipment to create a 24/96 digital chain. Predictably, even some cable companies are jumping on the band wagon claiming "Hi-Rez" digital interconnects, which is even more ridiculous because a stock "standard rez" cable will pass "Hi-Rez" bit perfect, how much better than perfect do you want a cable to be?

 It makes me angry that consumers are being so completely mislead purely to relieve them of their cash. At least with the great banking scandal it was mainly down to incompetance and stupidity driven by greed, what I see in the consumer audio world is still driven by greed but it's worse because it's deliberately calculated lies (or cleverly worded implications) and deception.

 G


----------



## ILikeMusic

Quote:


 It makes me angry that consumers are being so completely mislead purely to relieve them of their cash 
 

Remember though that some people _want_ to be misled. It is apparently part of the fun.


----------



## Publius

What gregorio said... none of this would matter much if it weren't for various financial interests promoting high res to high heaven. At the very least, the dearth of audible differences with real music should give anybody thinking about shelling out more money for a high res version considerable pause.

 I find it rather comical that 2L Records has some sample high res tracks showing off their obscenely overengineered download formats - with classical music almost utterly devoid of dynamic range. Admittedly it's very well performed and recorded, but still...

 Personally I think consumers are much smarter than many people give them credit for; the virtually complete failure of DVD-A and SACD in the marketplace jives nicely with the evidence for the inaudibility of high res on psychoacoustic and double-blind grounds. If it really was that much better - on emotional grounds - there would be a stampede to buy high res.


----------



## Publius

Quote:


  Originally Posted by *ILikeMusic* /img/forum/go_quote.gif 
_Remember though that some people want to be misled. It is apparently part of the fun._

 

Nobody wants to be misled. _Everybody_ in these debates believes science is on their side, and not on their opponents' side. (I would go so far as to say that's kind of the problem in the first place, but that's for a different conversation.)

 That said, I do think a lot of people are excessively trusting of figures of authority, particular audio engineers, audio reviewers, etc. instead of being sufficiently trusting in their own listening ability, or has a high enough self-esteem to learn the scientific concepts involved to a sufficient degree of competency. I've come to expect that expert opinions are meaningless in the audio field.


----------



## b0dhi

Quote:


  Originally Posted by *Publius* /img/forum/go_quote.gif 
_Personally I think consumers are much smarter than many people give them credit for; the virtually complete failure of DVD-A and SACD in the marketplace jives nicely with the evidence for the inaudibility of high res on psychoacoustic and double-blind grounds. If it really was that much better - on emotional grounds - there would be a stampede to buy high res._

 

This is jumping to conclusions IMO. Betamax lost to VHS even though it was superior. MP3 continues to reign supreme even though there are much better (psychoacoustically) codecs out there. 

 The failure of DVD-A and SACD are much more complex than just psychoacoustics. However, I would say that the marketing hype is overdone - the differences are nowhere near as big as claimed.


  Quote:


  Originally Posted by *Publius* /img/forum/go_quote.gif 
_Yeah b0dhi, the issue is not black and white. Nobody's disputing that situations exist where 24 bits can be ABX'd against 16 bits. That's trivial to do. But it's extremely difficult - possibly impossible - to ABX the difference involving music playback not specifically designed to exaggerate the differences. ie, playing back extremely attenuated music at extremely high volumes will easily show off the difference, but such situations simply don't seem to show up in commercially released music._

 

Because we're dealing with boundary conditions in terms of hearing capability, an ABX is more reflective of how well trained the person is in ABX rather than whether there is a difference in actual music listening.

 Moreover, my point wasn't about theoretical ABXing but actual mastering practices of all good studios in the world. I'm talking about normal music listening here. Putting aside dither that's applied to the audio chain at each re-quantisation stage, the final dither is the one I'm talking about. There's no accumulation of quantisation error possible after this stage, so dither at this stage can only affect the 16th-LSB. Since the studios do this to increase perceived dynamic range, they obviously must think it's audible. So what I'm saying is either the studios are right and 16bit is audibly different from 24bit _in regular music listening_, or they're wrong and it's inaudible.


----------



## Bostonears

Quote:


  Originally Posted by *Publius* /img/forum/go_quote.gif 
_Personally I think consumers are much smarter than many people give them credit for; the virtually complete failure of DVD-A and SACD in the marketplace jives nicely with the evidence for the inaudibility of high res on psychoacoustic and double-blind grounds. If it really was that much better - on emotional grounds - there would be a stampede to buy high res._

 

Personally, I never bought into DVD-A or SACD because I wanted all my discs to be compatible with all my CD players, including the one in my car, where I listen more than anywhere else. Of course, a moving car is about the worst place to try to hear a difference between 16 bit and 24 bit sound, but that wouldn't have mattered to me as long as I could have played the same disc as I played at home.

 Eventually, the market will evolve to where all digital players support 24 bits. At that point it probably won't matter whether or not 24 bits is audibly superior to 16 bits, the content will all migrate to 24 bits anyway.


----------



## JaZZ

Quote:


  Originally Posted by *Publius* /img/forum/go_quote.gif 
_Personally I think consumers are much smarter than many people give them credit for; the virtually complete failure of DVD-A and SACD in the marketplace jives nicely with the evidence for the inaudibility of high res on psychoacoustic and double-blind grounds. If it really was that much better - on emotional grounds - there would be a stampede to buy high res._

 

There's the equivalent in the music sector: The charts speak a clear language as to which music is really worth listening to.
.


----------



## thathertz

Quote:


  Originally Posted by *Publius* /img/forum/go_quote.gif 
_Personally I think consumers are much smarter than many people give them credit for; the virtually complete failure of DVD-A and SACD in the marketplace jives nicely with the evidence for the inaudibility of high res on psychoacoustic and double-blind grounds. If it really was that much better - on emotional grounds - there would be a stampede to buy high res._

 

Not true. Most people don't particularly care about the benefits of SACD / 
 DVD-A. mp3 sounds good enough (to the masses), it's portable between 
 PC, laptop and DAP (usually Ipod) and it's CHEAP (or FREE in most cases). 

 I have tried to impress my wife, son and daughter with SACD, FLAC's, and
 my system as a whole but they really don't care that much for sound 
 quality - even though they can hear the difference and admit it sounds good. 

 Even with FLAC being more prevalent on DAP's now, the masses will still
 go for mp3 because they can get more tracks on their player.


----------



## Arjisme

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_Moreover, my point wasn't about theoretical ABXing but actual mastering practices of all good studios in the world. I'm talking about normal music listening here. Putting aside dither that's applied to the audio chain at each re-quantisation stage, *the final dither is the one I'm talking about*. There's no accumulation of quantisation error possible after this stage, so dither at this stage can only affect the 16th-LSB. Since the studios do this to increase perceived dynamic range, they obviously must think it's audible. So *what I'm saying is either the studios are right and 16bit is audibly different from 24bit in regular music listening, or they're wrong and it's inaudible.*_

 

It sounds like you are separating that final dithering process from what you are referring to as "16bit." But that step should be considered part of the resultant 16bit product, since it is the final result consumers get to buy and listen to.


----------



## gregorio

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_Moreover, my point wasn't about theoretical ABXing but actual mastering practices of all good studios in the world. I'm talking about normal music listening here. Putting aside dither that's applied to the audio chain at each re-quantisation stage, the final dither is the one I'm talking about. There's no accumulation of quantisation error possible after this stage, so dither at this stage can only affect the 16th-LSB. Since the studios do this to increase perceived dynamic range, they obviously must think it's audible. So what I'm saying is either the studios are right and 16bit is audibly different from 24bit in regular music listening, or they're wrong and it's inaudible._

 

I think I understand where the mis-understadning lies. Dither is not added to increase the dynamic range. Dither does not increase dynamic range, it decreases it, as when dithering you are effectively adding white noise to the signal. It's not easy to explain in simple terminology but I'll give it a go:

 Think of it this way, to get to 16bit from 24bit we have to remove 8bits. Now, if we just cut off (truncate) those 8bits we are going to incur errors. These errors may be correlated (related) to the signal and therefore interact with (modulate) it, producing unintended tones in the program material. Although mathematically related to the signal, these tones are unlikely to be related harmonically, resulting in clashes (odd harmonics) which may be quite audible. However, if we can make sure that the truncation errors are all decorrelated from the program material then we have a perfect signal + some white noise, as decorrelated errors are percieved as white noise. The term "Dither" is this process of decorrelating the errors, which results in additional white noise. So effectively what we have done is cure the problem of odd harmonics and exchanged it for much less objectionable white noise. 

 BTW, it's this white noise (decorrelated errors) which defines the digital noise floor of the system, in the case of 16bit the digital noise floor is -96dB. 

 Ideally of course, we would like to get rid of both the correlated errors and the decorrelated errors (white noise), so we have a perfect signal with no noise. Unfortunately, that scenario is mathematically impossible but there is a cheat, noise shaping the dither. Moving that noise into an area of the frequency spectrum where are ours are insensitive, giving the perception that it's been reduced and therefore perceptually lowering the digital noise floor.

 Hope this helps?

 G


----------



## twylight

Is there a tl:dr summary of this anywhere in this thread? Everything I have is 16/41 flac.


----------



## Aleatoris

Here you go: 

 Stick with what you got. Unless you really want to re-buy all your music (or re-rip, if your CDs are 24/96).


----------



## b0dhi

Quote:


  Originally Posted by *Arjisme* /img/forum/go_quote.gif 
_It sounds like you are separating that final dithering process from what you are referring to as "16bit." But that step should be considered part of the resultant 16bit product, since it is the final result consumers get to buy and listen to._

 

Ofcourse it's part of the product. I never implied anything different. I'm merely highlighting a corollary - _if_ the final dither makes any audible difference, then the 16th-LSB of audio makes an audible difference, and therefore 16bit and 24bit are audibly different (because the 16th-LSB will be quantised differently in each when you compare dithered 16bit with 24bit (dithered or undithered)).


----------



## gregorio

b0dhi - Have a look at my last post. In theory, there will be differences between a 24bit master and a 16bit dithered version but these differences will not be audible at normal listening levels. I'm talking purely in terms of digital theory, I'm not considering that there may be other factors which may make a difference, say variations in mastering for example.

 G


----------



## Hudson

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Going back to the above, when we add a 'bit' of data we double the number of values available and therefore halve the number of quantisation errors. If we halve the number of quantisation errors, the result (after dithering) is a perfect waveform with halve the amount of noise._

 

Gregorio, this bit from your original post confuses me a little.

 You state that increasing bit depth by one bit halves the number of quantisation errors. I don't see how you're reducing the number of errors. Do you mean, halve the value of each quantisation error by reducing the range over which the error can occur?

 Cheers, stu.


----------



## gregorio

Quote:


  Originally Posted by *Hudson* /img/forum/go_quote.gif 
_Gregorio, this bit from your original post confuses me a little.You state that increasing bit depth by one bit halves the number of quantisation errors. I don't see how you're reducing the number of errors. Do you mean, halve the value of each quantisation error by reducing the range over which the error can occur?_

 

Yes, that's exactly what I meant, I didn't word it particularly well in the OP, sorry. I suppose potentially the number of errors could be reduced but taken as a statistical average it is the amount of error which is reduced by half. To be honest I don't know the precise maths (algorithms) used, so I'm not sure if each individual error is exactly reduced by half or, taken as an average, the overall amount of error is reduced by half. My guess is that the statistical nature of quantisation error and dithering means the latter is more likely to be true.

 G


----------



## Hudson

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_Yes, that's exactly what I meant, I didn't word it particularly well in the OP, sorry. I suppose potentially the number of errors could be reduced but taken as a statistical average it is the amount of error which is reduced by half. To be honest I don't know the precise maths (algorithms) used, so I'm not sure if each individual error is exactly reduced by half or, taken as an average, the overall amount of error is reduced by half. My guess is that the statistical nature of quantisation error and dithering means the latter is more likely to be true.

 G_

 

Thanks for clearing that up. I was pretty sure that was what you meant but my brain started to look for explanations that weren't there.

 I find this subject very interesting and having done a lot of dsp related work recently I would like to look into this area a little more. Could you recommend some sources of further reading regarding dithering implementation? (preferably not wiki stuff).

 Cheers, Stu.


----------



## jcx

as I mentioned much earlier in this thread you can listen to samples of dither at exagerated low 8,12 bit resolutions to get the concept of correlated quantization error with truncation and the improvement with dither:
http://www.head-fi.org/forums/f46/24...ml#post4051105


----------



## gregorio

Quote:


  Originally Posted by *Hudson* /img/forum/go_quote.gif 
_I find this subject very interesting and having done a lot of dsp related work recently I would like to look into this area a little more. Could you recommend some sources of further reading regarding dithering implementation? (preferably not wiki stuff). Cheers, Stu._

 

To be honest, I've picked up what I know from talking to other studio professionals and in online discussions with experts on the subject in professional audio forums. If you want real techie answers about dithering a good place to start would be the AES (Audio Engineering Society). Their journals is one of the most respected peer reviewed publishers of white papers on anything to do with audio. They must have some back issues which deal with dither but you'll have to pay. I have to say the articles are VERY technical, scientific papers and it was rare that I was able to fully understand most of them.

 The other person you could try and contact on here would be Dan Lavry (Lavry Engineering) who is a sponser on Head-Fi. Dan is regarded by many as one of the leading experts in the field of digital audio. I would imagine Dan knows a great deal about dither (or at least a lot more than me) and could probably point you in the direction of some informative documents.

 In the meantime have a look at this site, it has a couple of good papers on dither (which are not too techie): Digital Audio Explained - Papers

 G


----------



## Hudson

Thanks for the info. I shall have a good read when I can. And thanks for the interesting post.


----------



## leeperry

so this thread is on page 2 now...the myth has been exploded yet? 
	

	
	
		
		

		
			





 having tried many different types of audio files, several soundcards, OS, audio renderers...I think I'll dare to say that the more crappy/old your audio system, the more 24/96 will make a difference over 16/44.1.

 as I picture it, there's simply more data to help the DAC recreating the original waveform in 24/96....w/ 16/44.1 you need much better filtering, jitter control etc etc...

 feel free to disagree


----------



## Donald North

One important point that is rarely discussed and explain to the public: 24-bit FORMAT does not automatically equate to 24-bit RESOLUTION. This is due to noise in the electronics. Surveying the available commercial A-D and D-A converters and looking at their published noise specifications, few break 17 and 18 bit in actual resolution achieved.


----------



## gregorio

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_as I picture it, there's simply more data to help the DAC recreating the original waveform in 24/96....w/ 16/44.1 you need much better filtering, jitter control etc etc..._

 

This statement sounds entirely logical and plausible and demonstrates perfectly, in my opinion, the main problem with many people's understanding of digital audio. The truth of the matter is that providing there are more than two samples per audio cycle, the waveform can be recreated perfectly. This is the basic tenet of the Nyquist Theorem upon which digital audio is based. In other words, if we take say a 18kHz waveform and sample it at 44.1kFS/s and 96kFS/s, the 18kHz waveform will be recreated just as accurately at 44.1 as it will at 96. In fact, provided the sample rate is greater than 36kFS/s there is no increase in accuracy of the reconstructed waveform whatever the sample rate. Jitter control is irrelevant and how filters are implemented are down to individual DAC manufacturers. I re-iterate what has been mentioned before in this thread, in controlled DBTs no one has been able to distinguish 16/44.1 from higher resolution. This is substantiating evidence for the underlying scientific principles of digital audio. This deals with the sample rate and the bit depth (16 or 24) has already been dealt with in the OP.

 G


----------



## leeperry

well, on my previous M-Audio Audiophile USB 24/96 was WAY better than 16/44.1..it sounded a lot more hi-fi

 on my current STX(fit w/ LME49720HA op-amps), the difference doesn't really exist anymore....and DAC manufacturers(especially AKM) have been clear that they've kept on improving the oversampling and post-filtering in a pretty amazing way in their latest DAC chips.

 so I still rest my case that w/ low quality stuff, 24/96 will have an edge over 16/44.1 due to poor DAC/op-amp filtering(44.1 being a lot less linear than 48/96)....but w/ better/newer gear the difference is not quite audible anymore 
	

	
	
		
		

		
		
	


	




 at least, that's what I'm hearing...and going through manufacturer's datasheets/statements seems to confirm my impressions. I can honestly say that I'm finally enjoying my CD collection


----------



## HeadLover

I don't know, but I still claim that 24/96 is a little better, but maybe I am just crazy


----------



## JaZZ

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I don't know, but I still claim that 24/96 is a little better, but maybe I am just crazy 
	

	
	
		
		

		
			



_

 

With my McCormack UDP-1 (which isn't crap) I hear the same. But I must concede that on my computer rig the difference between a 24/96 original and the downsampled version (16/44.1) is smaller.
.


----------



## sanderx

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_This statement sounds entirely logical and plausible and demonstrates perfectly, in my opinion, the main problem with many people's understanding of digital audio. The truth of the matter is that providing there are more than two samples per audio cycle, the waveform can be recreated perfectly. This is the basic tenet of the Nyquist Theorem upon which digital audio is based._

 

I find it disturbing when people describe the Nyquist theorem like this. The theorem really only holds if all past and future samples are available or that the signal be periodic (which then happens to provide said past and future signals by virtue of repetition). There is no real guarantee that in other cases the waveform (function) will be re-created. Integration from -infinity to infinity is integral part of the theorem. In fact, for any k point sample size you will be able to find infinitely many functions that are not precisely reproduced despite having no frequency component above fs/2. The Nyquist theorem gets abused way too much - having to deal with just past samples or a limited set of samples or quantization sets different limits than Nyquist. 

 However, its also true that DAC-s do produce fairly good reproductions of the initial waveforms up to frequency of fs/2 ... but tat is rather despite than thanks to the Nyquist theorem that really sets the limits in this case, rather than providing guarantees.


----------



## jph-22

Gregorio:

 Thanks for your overview on digital recording. I have one question, then a comment:

 1. Besides dynamic range, doesn't higher-bit capture also extend signal to noise ratio ?

 2. I think orchestras have closer to 70db of dynamic range, not 60, but who's counting ? 16 bits encodes far more than EITHER of these !! The complaints about Red Book/16 stemmed from the intermediate calculations done on the signal (the "headroom" needed in production) that siphoned-off some resolution. But this was completely solved nearly 20 years ago with 20-bit recording. After this, we finally got the full 16 bits of resolution on disc - dithering helped us even more.

 It's only in the last few years that we began to HEAR that resolution !!

 Thanks again


----------



## Donald North

Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_I re-iterate what has been mentioned before in this thread, in controlled DBTs no one has been able to distinguish 16/44.1 from higher resolution. 

 G_

 

I recently read a paper in the AES where the authors conducted listening tests between 16/44.1 and higher resolution, finding that the listeners could not hear a difference.

 At no where is it mentioned that the equipment and recordings used were measured and confirmed that they actually do contain and produce higher resolution audio! As I've mentioned earlier, many of these "hi-res" digital recorders and a-d/d-a converters don't actually deliver the goods based on their published noise and dynamic range specs. Some fail to achieve even true 16 bit performance. How can a listener compare and have the opportunity to distinguish between 16/44.1 and 24/96 if they aren't actually presented with higher resolution audio?


----------



## cerbie

Non-asbestos flame-proof suit ON. 
	

	
	
		
		

		
		
	


	




 I've now read through this whole thread...and my conclusion is this: 96kHz/24bps looks to me like a consumer format to stop at, _which I did not think before I started reading it_. Having a format that *just* reaches human limits means a dynamic and detailed recording will need perfection in its processing to live up to that limit.

 Depth/frequency: if humans can hear some differences to around -80dB (IIRC) of the main sounds by actual trials of some kinds, then the format should be several dB of "what if" headroom, before any _other_ headroom is counted in. There is much I do not understand, especially as it concerns masking/noise-shaping. I do, however, understand the basic maths for Nyquist, and dealings with practical error approximation. Also, the Nyquist theorem only fully holds for the half frequency limit with perfect no-harmonic primary tones, with perfect periods, and no other sounds being used, kind of eliminating reality, as most such things tend to. I'm not bashing digital audio--analog has even more pitfalls, unless you only listen to master tapes--but physical reality does introduce imperfections, making it as good as a theorem attempting to describe physics, not some magical panacea.

 You may be able to fool ears, but wouldn't it be better _to not have to?_ 96/24 (or 88.2/whatever, if the masters are that), but the ability to play back 96 is more common) would allow for very small errors made in the production process[1] to be below the equipment noise floor by several bits (which CD is not, anymore), much less the actual audible noise floor (which CD generally is, except in the case of IEMs, and then only sometimes), thus eliminating the need for _absolute perfection_, *making a production that is good enough by one set of trained ears more likely to be good enough by all sets of ears*.

 Storage space: with hard drive and DVD backups, we are at about a $0.25 per gigabyte, and decreasing (around $0.50/GB even with RAID 5 + RAID 5 + TY DVDs). 192/24 or higher would be double or more for basically nothing, but 88.2/16 to 96/24 seem pretty reasonable, at this point and time, and most new computers can output those bit-perfectly (just slap on a DAC).

 I'm not losing sleep at night because I might be missing something in CDDA-based stuff I listen to (my output chain is not up that task yet, much less any biological aspects), and overall production quality is still the big priority; but if 96/24 becomes reasonable for non-loudness-war-afflicted recordings that I'm interested in, then I'll bite (there are now only a handful, and all too expensive, IMO). I like the idea of a format having a bit of headroom well beyond any potential for what I can hear, and that headroom allowing for minor errors to have a much greater chance--_by orders of magnitude_--of being completely inaudible.

 1[size=x-small] - gregorio admits errors can and do happen (but tends to phrase it more like, "if it's done just right, then...")--and I'm going to assume the guys doing production work are _not_ *perfect* beings, with *perfect* equipment, and *perfect* software, which appears to be an assumption being made for 44.1/16 being good enough. Software differences alone are being blamed for pages worth of discussion.

 If that can happen, then the format itself needs to be given the headroom to allow _many such errors to propagate_, yet still remain below quality equipment's noise floor by a reasonable confidence, and below _the most strenuous recordings' noise floors_ by a reasonable confidence, as well--even if that means only recordings with cannon blasts count! Either that, or the best methods of filtering need to be applicable independent of being secrets of software/hardware companies, so that all software is identical in terms of input/output, including having as many options for processing available (extra bits and samples/sec seems a lot easier to do).[/size]

 Now, once I get a bite to eat, I'm going to go _happily listen_ to some 44.1/16 recordings of music with <30dB range, and light digital clipping, not to even get into available resolution or quality of amplification (IE: regardless of considering any possible nitpicks, *the music comes first*).


----------



## nick_charles

Quote:


  Originally Posted by *Donald North* /img/forum/go_quote.gif 
_I recently read a paper in the AES where the authors conducted listening tests between 16/44.1 and higher resolution, finding that the listeners could not hear a difference.

 At no where is it mentioned that the equipment and recordings used were measured and confirmed that they actually do contain and produce higher resolution audio! As I've mentioned earlier, many of these "hi-res" digital recorders and a-d/d-a converters don't actually deliver the goods based on their published noise and dynamic range specs. Some fail to achieve even true 16 bit performance. How can a listener compare and have the opportunity to distinguish between 16/44.1 and 24/96 if they aren't actually presented with higher resolution audio?_

 

That would be Meyer and Moran 2007. M and M did follow up their paper and described their kit which was based on 3 universal/high res players which ranged in their SNRs (at least one was measured by Hysteriaophile, one was a pretty well regarded unit), from about 108db to 118db iirc but it is on the BAS website. 

http://www.bostonaudiosociety.org/explanation.htm


 So one could argue that the machines did not exploit the capabilities of high res PCM or DSD. However, the AD/DA stage which was a standard 16 bit stage must by definition have throttled the SNR by between 9 and 19db or a factor of 8x in the most extreme case and removed all frequencies above 22049 and severely curtailed all frequencies above 20K.

 The point that is perhaps more interesting is where would you find any piece of recorded music with a dynamic range of more than 96db and what would you play it on so that you got all of that dynamic range above background noise ?


----------



## Donald North

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_That would be Meyer and Moran 2007. M and M did follow up their paper and described their kit which was based on 3 universal/high res players which ranged in their SNRs (at least one was measured by Hysteriaophile, one was a pretty well regarded unit), from about 105db to 115db iirc but it is on the BAS website. 

BAS Experiment Explanation page - Oct 2007


 So one could argue that the machines did not exploit the capabilities of high res PCM or DSD. However, the AD/DA stage which was a standard 16 bit stage must by definition have throttled the SNR by between 9 and 19db or a factor of 8x in the most extreme case and removed all frequencies above 22049 and severely curtailed all frequencies above 20K.

 The point that is perhaps more interesting is where would you find any piece of recorded music with a dynamic range of more than 96db and what would you play it on so that you got all of that dynamic range above background noise ?_

 

Thanks for this link Nick. 

 This test leaves much to be desired. For starters, I would have not used commercially available software. Instead I would have made my own recordings using state-of-the-art equipment which have published and known capabilities. 

 Better yet, I would have set up a live feed as the control. You would then have live musicians playing and you could select among direct, hi-res PCM or DSD, and 16/44.1 in the monitor room and compare. I suspect this would be very informative.


----------



## nick_charles

Quote:


  Originally Posted by *Donald North* /img/forum/go_quote.gif 
_Thanks for this link Nick. 

 This test leaves much to be desired. For starters, I would have not used commercially available software. Instead I would have made my own recordings using state-of-the-art equipment which have published and known capabilities. 

 Better yet, I would have set up a live feed as the control. You would then have live musicians playing and you could select among direct, hi-res PCM or DSD, and 16/44.1 in the monitor room and compare. I suspect this would be very informative._

 

If the question is "is an optimal High-Res system with optimally recorded/processed high res programme material audibly different from an optimal 16/44.1 system with optimally recorded/processed 16/44 material ?" then yes that would be a good approach.

 Generally though the question is framed as "is SACD/DVD-A better than CD ?" and this typically devolves into I have X on SACD/DVD-A and it is better/no better than X on CD, which may be true/false for a given instance of X. 

 Clearly SACD/DVD-A are technically superior to CD but if no SACD/DVD-A player delivers the 120db or 144db dynamic range and no recording exploits this anyway then for practical purposes this superiority is moot ?

 In any case M and M did show that the noise differences could be detected if you cranked up the volume high enough (unpleasantly or painfully loud)...


----------



## Donald North

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_If the question is "is an optimal High-Res system with optimally recorded/processed high res programme material audibly different from an optimal 16/44.1 system with optimally recorded/processed 16/44 material ?" then yes that would be a good approach._

 

This is an important question to ask and pursue, if we are trying to determine what is an audibly transparent recording & playback format/medium.

 I know from first hand experience which sounds closer to a live feed: all-analog LP or 16/44.1 CD.

  Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Generally though the question is framed as "is SACD/DVD-A better than CD ?" and this typically devolves into I have X on SACD/DVD-A and it is better/no better than X on CD, which may be true/false for a given instance of X._

 

This is a problem. Why should consumers buy a SACD/DVD-A if it hasn't been confirmed that there's actually higher resolution content on them. I personally have heard a couple SACD vs. CD comparisons and have not been impressed. 

  Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Clearly SACD/DVD-A are technically superior to CD but if no SACD/DVD-A player delivers the 120db or 144db dynamic range and no recording exploits this anyway then for practical purposes this superiority is moot ?_

 

Well, the format has the potential to be superior but it's all in execution starting at the recording studio. Even if one's playback amplifier doesn't have -120dB noise floor, it would still be interesting to hear. You won't know what you will hear until you try


----------



## leeperry

Quote:


  Originally Posted by *Donald North* /img/forum/go_quote.gif 
_I personally have heard a couple SACD vs. CD comparisons and have not been impressed._

 

try Violator from DM, the SACD is really impressive...prolly due to superior mastering, but still!


----------



## Edwood

My brain......hurts.......

 So basically it's a Catch22 scenario.

 Without pushing Higher resolution recording, there is no expectation of better quality. But the Higher resolution content does not make a noticeable difference, so in the end it's a waste of money. It's really the original content creation and engineering that's important. Garbage in, Garbage out. But without being held to a higher standard, there will be no higher quality. So the illusion of higher resolution recording content must be maintained in order to maintain final music product quality.

 Believe the lie.


----------



## nick_charles

Quote:


  Originally Posted by *Donald North* /img/forum/go_quote.gif 
_
 I know from first hand experience which sounds closer to a live feed: all-analog LP or 16/44.1 CD._

 

I have to ask...

  Quote:


 Even if one's playback amplifier doesn't have -120dB noise floor, it would still be interesting to hear. You won't know what you will hear until you try 
	

	
	
		
		

		
		
	


	



 

If the amp has a -100db noise floor then 44db of a 144db system will be lost in noise ? But there is also ambient noise which is rarely below 25 - 30db so to get 144db of dynamic range you need to have music peaking at over 160db


----------



## brexel

hi!

 first of all thanks a lot for that very interesting posting...i do however have a question:

  Quote:


  Originally Posted by *gregorio* /img/forum/go_quote.gif 
_So, 24bit does add more 'resolution' compared to 16bit but this added resolution doesn't mean higher quality, it just means we can encode a larger dynamic range._

 

so, do i interpret this sentence right when i think you say that the effective voltage difference between

 1111111111111111 and 1111111111111110 on 16 bit audio is the same as the difference between

 111111111111111111111111 and 111111111111111111111110 on 24 bit audio?


----------



## Donald North

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_I have to ask..._

 

It was the all-analog LP. See these links:
Vinyl Asylum: RE: Nobody would buy a 'perfect' turntable by Donald North
Vinyl Asylum: RE: Nobody would buy a 'perfect' turntable by Donald North
Propeller Head Plaza: RE: a technical discussion of optimum Turntable/Arm peformance and design goals by Donald North


----------



## nick_charles

Quote:


  Originally Posted by *Donald North* /img/forum/go_quote.gif 
_It was the all-analog LP._

 

So, remind me of the real world limitations of a vinyl system compared to a 16/44.1 digital system 
	

	
	
		
		

		
		
	


	




 How can something that has so many real limitations (poor snr, high distortion, limited dynamic range at high and low frequencies, physical problems such as groove noise, rumble, bearing noise, speed stability, clicks, pops and the rest be accurate in any meaningful sense of the word ?


----------



## Donald North

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_So, remind me of the real world limitations of a vinyl system compared to a 16/44.1 digital system 
	

	
	
		
		

		
			





 How can something that has so many real limitations (poor snr, high distortion, limited dynamic range at high and low frequencies, physical problems such as groove noise, rumble, bearing noise, speed stability, clicks, pops and the rest be accurate in any meaningful sense of the word ?_

 

Well, you first have to hear and witness for yourself 
	

	
	
		
		

		
		
	


	




 What makes you say it has high distortion? I don't hear any distortion with my Goldmund Studio with T-3F linear tracking tonearm and Audio-Technica OC-9MLII cartridge with microline stylus.

 As for dynamic range, have you seen this:
Clearaudio Moving Coil Cartridges

 Many of your other complaints (rumble, bearing noise, speed stability, etc) are all a function of the quality of the turntable. In the comparisons I heard at the Caltech Music Lab, we were using an AR turntable with SME Series III tonearm and Shure V-15VMR cartridge: all good quality but not necessarily state of the art. CD player was a Luxman, also good quality at the time.


----------



## DoYouRight

Well I read this whole article and still am confused about an issue that has been plaguing me for awhile. 

 How do I get 24/96 out of my laptop without optical. I am searching all these crazy usb dongles, but I would like it to be less complicated.

 I don't think 24/96 sounds better, but some Vinyl Rips in flac are 24/96 and I would like to be able to play them over a B22. So even the Buffalo32 doesnt do anything over 16/44 over USB. So I need something to give me laptop to spdif or something.

 My quesiton is, if the dac wont do that over usb or something, will it still play sound and just not be sent as 24 but only 16bit?

 I just want to hear the vinyl rips I dont think there is any more of a "better" sq.


----------



## cerbie

You need something that explicitly supports it. Generally, 32-48/16 is what you get with the common implementations (usually using PCM270x). You need something you can plug in that has drivers for your OS to get higher sampling rates out.


----------



## sanderx

Quote:


  Originally Posted by *DoYouRight* /img/forum/go_quote.gif 
_How do I get 24/96 out of my laptop without optical. I am searching all these crazy usb dongles, but I would like it to be less complicated.

 I don't think 24/96 sounds better, but some Vinyl Rips in flac are 24/96 and I would like to be able to play them over a B22. So even the Buffalo32 doesnt do anything over 16/44 over USB. So I need something to give me laptop to spdif or something._

 

Essentially you have two options - either a dac with optical spdif in or a portable dac with firewire in. Depends on what your laptop has and what is cheaper to add. While it is possible to make a usb dac that has a higher rate than 16/44.1|48kHz, these will require drivers and this seems to be a way too expensive proposition for most makers of usb dacs. Weiss DAC2, Weiss Minerva, Apogee Mini-DAC. 

 Adding a Audigy 2 ZS to your laptop is another way.


----------



## nick_charles

Quote:


  Originally Posted by *Donald North* /img/forum/go_quote.gif 
_Well, you first have to hear and witness for yourself 
	

	
	
		
		

		
		
	


	




 What makes you say it has high distortion? I don't hear any distortion with my Goldmund Studio with T-3F linear tracking tonearm and Audio-Technica OC-9MLII cartridge with microline stylus.

 As for dynamic range, have you seen this:
Clearaudio Moving Coil Cartridges

 Many of your other complaints (rumble, bearing noise, speed stability, etc) are all a function of the quality of the turntable. In the comparisons I heard at the Caltech Music Lab, we were using an AR turntable with SME Series III tonearm and Shure V-15VMR cartridge: all good quality but not necessarily state of the art. CD player was a Luxman, also good quality at the time._

 

Hmm , I must admit to some skepticism of Clearaudio's graphs, I have never seen anyone claim that record noise can get down to -100db or below. This is a good 30db less than Ben Bauer was finding and 20db below the -80db that was being touted even in the early 2000s as being physically impossible to better due to the real problem of physical irregularities in groove walls on vinyl LPs, but those German engineers are very clever so who knows...

 However,

Digital Vinyl ? !

 suggests that this is not possible...and more to the point look at their graph very carefully, at that precise frequency the gap between noise and signal is 100db however three things leap out, 1) the 300hz test signal is displaced by about 10hz that is a huge frequency inaccuracy , 2) look a few hz to the right and you see a nice big -75db noise spike which they have conveniently not measured and 3) look at the harmonic distortion is is grotesque...the 2nd harmonic is at -35db !!

 But even allowing this 4db better than CD at one specific frequency on a pristine test record at an unspecified speed on the outer edge. What is the dynamic range like at say
 5K , 10K or 15K or in the inner 1/3rd.

 But going back to the harmonic distortion , -35db for the 2nd harmonic, hmm how does this $10,000 (rrp) cartridge compare with say a $145 DVD player, well the Oppo 970H delivers nothing above -100db and the far technically inferior Sony Playstation One delivers harmonics no higher than -90db.

 Channel separation is what 45db ?

 So lets summarize to get an LP system with a dynamic range better than CD at one specific frequency, shifted about 10hz from its real value albeit with massive harmonic distortion a frequency shifted noise peak at -75db and dreadful crosstalk I need a $10,000 cartridge , does that mean I would have to upgrade my Rega ?


----------



## Donald North

What dreadful crosstalk? And how much channel separation is really needed? 25-30dB?

 When listening to loudspeakers, what's the channel separation at the listener's ears? 
	

	
	
		
		

		
		
	


	




 Inner groove distortion is caused by geometric error from pivoted tonearms. Linear tracking tonearms play the record as it was cut and don't have this problem.

 You don't need a $10k cartridge: The Audio Technica AT-OC9MLII with microline stylus is the best tracking moving coil cartridge I've encountered to date, regardless of price. I think it sounds good too.

 We can argue technicalities all we want. At the end of the day, the proof is in the sound. I highly recommend you buy Boyk's Mussorgsky LP and CD and give them a listen for yourself. I'm curious to hear your opinion.

http://www.performancerecordings.com/albums.html


----------



## DoYouRight

The thing is, I have my onboard SPDIF routed to the HDMI on my Dell XPS M1330. Also I have expresscard only so I have the creative notebook x-fi so it will optical to unit and decode the dts, BUT I have tried so much but this component is NOT compatible with Ubuntu, the creative drivers for Ubuntu SUCK SO BAD. So Im thinking I will need a 0404usb or M-Audio Fasttrack pro, will either of these support dts stuff for movies than the D/A be done via SPDIF on my dac?


----------



## nick_charles

Quote:


  Originally Posted by *Donald North* /img/forum/go_quote.gif 
_What dreadful crosstalk? And how much channel separation is really needed? 25-30dB?_

 

You are the engineer, not me, but at a point when the crosstalk gets too bad stereo turns to a mono like mush 
	

	
	
		
		

		
		
	


	





  Quote:


 When listening to loudspeakers, what's the channel separation at the listener's ears? 
	

	
	
		
		

		
		
	


	



 

Well, the stereo image from speakers is created by the intersection of two slightly different images, if the two images start the same then you do not get stereo ?

  Quote:


 Inner groove distortion is caused by geometric error from pivoted tonearms. Linear tracking tonearms play the record as it was cut and don't have this problem. 
 

I did not mention inner groove distortion, I alluded to the difficulty of getting the same dynamic range in inner grooves.


  Quote:


 We can argue technicalities all we want. At the end of the day, the proof is in the sound. I highly recommend you buy Boyk's Mussorgsky LP and CD and give them a listen for yourself. I'm curious to hear your opinion. 
 

These technicalities are surely part of your profession as an engineer. You made an assertion about which (CD or LP) was more accurate, accuracy is about the least change from the source i.e adding least and taking away least.


----------



## cerbie

Quote:


  Originally Posted by *DoYouRight* /img/forum/go_quote.gif 
_The thing is, I have my onboard SPDIF routed to the HDMI on my Dell XPS M1330. Also I have expresscard only so I have the creative notebook x-fi so it will optical to unit and decode the dts, BUT I have tried so much but this component is NOT compatible with Ubuntu, the creative drivers for Ubuntu SUCK SO BAD. So Im thinking I will need a 0404usb or M-Audio Fasttrack pro, will either of these support dts stuff for movies than the D/A be done via SPDIF on my dac?_

 

Time for a thread in *Computer Audio*, maybe? My personal opinion would be to lower your standards (get Windows), or get a desktop computer to handle the work (w/ a PCI or PCI-e card that Linux likes). Blame the legal existence of intellectual property.


----------



## jcx

As a engineer I might point out the ClearAudio graphs need a little more explaination to justify the claim they demonstrate geater than CD S/N from vinyl - the 1 Hz BW spot noise for 16 bit CD is -120 dB (what is the integration time for ClearAudio's graphs - I assume 1 Hz spot noise bandwidth when its not otherwize indicated)

 the practical noise floor of noise shaped dithered 16/44.1 is below -140 dB over our most sensitive hearing range - check out Lukin's dither noise floor graphs:

Homepage of Alexey Lukin


----------



## Donald North

Quote:


  Originally Posted by *jcx* /img/forum/go_quote.gif 
_As a engineer I might point out the ClearAudio graphs need a little more explaination to justify the claim they demonstrate geater than CD S/N from vinyl - the 1 Hz BW spot noise for 16 bit CD is -120 dB (what is the integration time for ClearAudio's graphs - I assume 1 Hz spot noise bandwidth when its not otherwize indicated)

 the practical noise floor of noise shaped dithered 16/44.1 is below -140 dB over our most sensitive hearing range - check out Lukin's dither noise floor graphs:

Homepage of Alexey Lukin_

 

I agree, I'd like to see more details how the Clearaudio measurement was performed. 

 The spectrum graphs in your link don't correlate with the values in the table. For example: No noise shaping (TPDF dither) is -132dB on the graph while -96dBrms in the table. 

 Additionally, using the formula dB dynamic range = 6.02n + 1.76, where n is the number of bits, 16-bit CD has a theoretical maximum dynamic range of 98dB.


----------



## jcx

you're right about Lukin's TPDF level looking wrong, probably need to add ~ 15 dB for 16/44.1 spot noise in dB/sqrt(Hz)?

 assuming the same scale factor the rest of the graphs still show up to ~130 dB spot noise at our peak hearing sensitivity, depending on which noise shaping used with CD - low enough that ambient room, electronic and microphone noise set the real limits in any recording of a live event

 I am deeply suspicious of the green curve in the ClearAudio graph – is it really RIAA amped cart needle on a rotating record, groove cut with live electronics driving the cutter head? Where’s the low frequency noise? Isn't the red line the classic -60dB audio standard S/N test track?

http://www.musicalsurroundings.com/clearaudio/mc.html

 EDIT: Google Language tools to the rescue

 green:tonarm angehoben == arm raised - hardly a test of Vinyl dynamic range!!


 generously crediting the red base line as flat -105 dB I get -110 dB * sqrt(20KHz) ~ 90 dB broadband dynamic range from the record groove


----------



## Hergal79

Hi Gregorio, this is my first message in the forum.
 I have a doubt about technical implementation of the Nyquist Theorem, there is a detail that make me doubt about the reconstruction. 
 The reconstruction suppose that you know the ENTIRE signal, you have to sum the signals modulated by functions of the form sin(x)/x, but when you do in real time is obvious that you can only sum the contributions of samples that are played or have been played (in the past). The point is: the samples to be reproduced in the future have always a not null contribution (because sin(x)/x decrease in absolute value away from the peak but is false that it cancels out after a moment). 
 It's impossible to know the entire signal in a system... 
 I imagine that the solution is to delay play, to add the contributions involved in a short time in the future, but I'm just assuming ...
 I'm wrong? What about the perfect reconstruction?
 Sorry for my english and for my not technical writing. I hope I made myself clear.
 Thanks for the thread!


----------



## Publius

Quote:


  Originally Posted by *jcx* /img/forum/go_quote.gif 
_green:tonarm angehoben == arm raised - hardly a test of Vinyl dynamic range!!

 generously crediting the red base line as flat -105 dB I get -110 dB * sqrt(20KHz) ~ 90 dB broadband dynamic range from the record groove_

 

I'll add another factoid here. While I'm not going to defend Clearaudio's advertising, bald-faced lying that it is, I _would_ like to defend the use of arm-raised measurements in relation to bit depth.

 My flat-eq recordings (with an AT-OC9) have a noise floor of -75dBFS, with a reference tone of 1khz 0db recording at, IIRC, -27dbFS. (even then I get pretty close to clipping with some 12"s.) However, most of this energy is from a wide +20db noise bump at 700hz, and lots of 1khz harmonic hash, both the result of my audio interface that really shouldn't be there. When I look at a spectrum plot in Audacity underneath those peaks the spot noise floor, after 73.5db of amplification, is about -64dB. Moreover, recording in 16 bits instead of 24 bits raises this noise floor by 1-2dB. This suggests a spot noise floor of something like -136dB/rtHz?

 So, at the very least, I think I can offer a reasonable argument for recording some (but not all) vinyl chains at greater than 16 bits; while the vinyl noise floor is very unlikely to go beneath the thermal noise floor of the cartridge, it is not _completely_ out of the question, so you might as well keep the quantization noise well below the cartridge noise if you can.

 That said, the noise floor on every record I've ever recorded has been far higher than that, and that is certainly not real evidence that anything greater than 16 bits is necessary for post-RIAA playback. In fact, all my post-processed vinyl recordings are encoded with LossyWAV, which at --extreme yields an average "real" bit depth of 15.2 bits


----------



## Publius

Quote:


  Originally Posted by *Hergal79* /img/forum/go_quote.gif 
_Hi Gregorio, this is my first message in the forum.
 I have a doubt about technical implementation of the Nyquist Theorem, there is a detail that make me doubt about the reconstruction. 
 The reconstruction suppose that you know the ENTIRE signal, you have to sum the signals modulated by functions of the form sin(x)/x, but when you do in real time is obvious that you can only sum the contributions of samples that are played or have been played (in the past). The point is: the samples to be reproduced in the future have always a not null contribution (because sin(x)/x decrease in absolute value away from the peak but is false that it cancels out after a moment). 
 It's impossible to know the entire signal in a system... 
 I imagine that the solution is to delay play, to add the contributions involved in a short time in the future, but I'm just assuming ...
 I'm wrong? What about the perfect reconstruction?
 Sorry for my english and for my not technical writing. I hope I made myself clear.
 Thanks for the thread!_

 

Read up on symmetric FIR filters. This is elementary digital filter design stuff. Yes, the filtering is time-delayed. Moreover, the introduction of a "finite" filter to implement the "infinite" sin(x)/x operation affects the signal in ways that are at once a) easily quantifiable and b) shown to be unimportant in modern designs. The same sort of analysis applies to other filters which are not straight-up truncations of sinc(x).

 Just because a reconstruction isn't "perfect" doesn't mean one is justified in fretting over the imperfections.


----------



## Publius

Another point on SNR. If you take the theoretical value on a 20-20khz rms basis, and compare the rms voltage of the cart with a 0db sine vs the total rms thermal noise, you get

 20*log(0.8e-3/(0.13e-9*sqrt(30*20000)))/log(10)

 = 78 dB

 This, it must be said, is 2db better than my OC9 (which for 0.4db/12 ohms yields 76db). On a spot basis that improves to 121db, but CD-DA evaluated on the same basis would yield something like 133-139db depending on how you treat the RMS values.

 If you are _particularly_ generous, and allow the use of velocity peaks extending to +15dB, then Clearaudio's numbers more or less equal out. But that's not what they're claiming..... I mean, given that (like I said before) the fact that I am recording 10khz 0db tones at -27dbFS and yet still clipping on some music does imply that records are out there which are punching 110cm/s, but much of that velocity only comes about due to tracking/tracing distortion - not because the cutting head is actually moving that fast to begin with.


----------



## xnor

Great write-up gregorio! I learned quite a few things there..


----------



## tosehee

I've learned so many things from this thread. Thank you very much, G!!

 Now, I am totally convinced that it is a total waste of space to store 24/192kFz materials. What software can I use to down sample these, so there is absolutely no audible differences between this and downsampled 16/44.1k?


----------



## leeperry

G is gone, he seems to believe that we're too dumb...maybe we are, after all 
	

	
	
		
		

		
		
	


	




 some manufacturer posted some supposedly 16/24bit files w/ a huge SQ difference, but the mods killed it?! http://www.head-fi.org/forums/showthread.php?t=440388

 it was called "Real-world comparisons of 24-bit and 16-bit music from Linn Records - free giveaway!"


----------



## tosehee

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_G is gone, he seems to believe that we're too dumb...maybe we are, after all 
	

	
	
		
		

		
		
	


	




 some manufacturer posted some supposedly 16/24bit files w/ a huge SQ difference, but the mods killed it?! http://www.head-fi.org/forums/showthread.php?t=440388

 it was called "Real-world comparisons of 24-bit and 16-bit music from Linn Records - free giveaway!"_

 

You must be referring to this.

LinnRecords - The 24bit Comparison - Alyn Cosker - SoundCloud


----------



## leeperry

exactly! can you hear a huge SQ difference? I can't say that I can...I'll try again tomorrow w/ fresh ears


----------



## xnor

Well anyone can post two different sounding files and call it "16/24 bit real-world comparison".
 That's why I asked for the original source.

 edit: Will also do some ABX tomorrow, but I will take the 24/96 file as source and convert it myself. I really don't trust these guys.


----------



## leeperry

well, as I wrote in their thread before it was closed...they should indeed let us know how these 2 files were created..

 if they come from the same 32float source? same dithering algorithm? if the 16bit was downconverted from the 24bit?

 I've got some 7.1 24/48 TrueHD lossless soundtracks that truly lack in 16bit...but it's due to the conversion, integer>integer is lossy as hell.


----------



## xnor

Yeah, well I did some tests with their 24/96 file. I converted it to 16/44.1 (with fb2k, dithering enabled) and created an audio diff with DiffMaker. The resulting file shows what I expected: *no difference*.


----------



## nick_charles

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_Well anyone can post two different sounding files and call it "16/24 bit real-world comparison".
 That's why I asked for the original source.

 edit: Will also do some ABX tomorrow, but I will take the 24/96 file as source and convert it myself. I really don't trust these guys. 
	

	
	
		
		

		
		
	


	


_

 

 Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_Yeah, well I did some tests with their 24/96 file. I converted it to 16/44.1 (with fb2k, dithering enabled) and created an audio diff with DiffMaker. The resulting file shows what I expected: *no difference*._

 

They said they had two different masters. When I ran some stats on them in Cool Edit Pro they were a bit different the 24 bit had 6db more dynamic range, but both were under 96db anyway. Looking at the spectrograms in Audacity the 24 bit hits close to max (0db) far more frequently than the 16 bit, this would make it appear more dynamic perhaps. These do not look quite the same to me.

 I downsampled with CEP and compared the 16 bits and 24 bit-down-sampled versions and then plotted the differences they average out at 0.18db, varying from 0.08db to 0.37db, but that may be my process, even so the differences are quite small though 0.18db might just be audible as an average loduness difference but I doubt it.

 Not sure that these samaples prove very much one way or another ?


----------



## Jim Collinson

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_some manufacturer posted some supposedly 16/24bit files w/ a huge SQ difference, but the mods killed it?!_

 

That was us. Linn Records. We didn't actually say there was a huge SQ difference.

 Don't now why the mods killed it ... perhaps I didn't ready the T's and C's closely enough, if anyone could shed any light on that it would be appreciated. 

  Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_exactly! can you hear a huge SQ difference? I can't say that I can..._

 

I suppose I should have pointed out that these are the exact files as we sell them on through our website, hence the reason I called it a 'real-world' comparison. We have not doctored the files in any way for effect, in fact quite the opposite...

 The 16-bit file is the same as we press to our CDs which are all hybrid HDCDs. Hence it goes through a slightly different mastering process and is encoded for HDCD. We actually try and get it as close to the 24-bit Studio Master as we can, within the limitations of the media.

 It is of no benefit to us to dumb down our CD releases in order to highlight the quality 24-bit files, that would just be daft!

 The whole idea of the 24-bit Studio Master format is to get as close to the original recording as we can, and have that reproduced for the listener at home. Yes you do need to have a level of investment in your hi-fi and listening environment for it to make the difference - having an in adequate DAC or soundcard can narrow the difference so much that there is little point. We do hope, and believe, that the technology will catch up, and that's part of what Linn as a hi-fi manufacturer aims to do too.

 My self and our chief engineer, Philip Hobbs, are devising a more rigorous test from an original analogue tape that should satisfy the audio engineering geeks amongst us. That is a little while away though, as Phil has a very busy recording schedule at the moment.

 It should be said though, that we still think CD quality is great. I love my CD collection! And all of our releases are available on hybrid SA/HDCDs. 

 There are listeners who want the to get as close to the original recording - as the engineer, and musicians intended it - though and have the hi-fi capable of reproducing it, and for them we have 24-bit ... (or of course vinyl 
	

	
	
		
		

		
		
	


	




).

  Quote:


  Originally Posted by *xnor;* 
_I really don't trust these guys._

 

A bit harsh! We have been making audiophile recordings since 1983 and take it very seriously. We aren't about to fudge something just for some slim marketing advantage. If you wanted to check me on that, download the tracks direct from the website and then analyse the HDCD, it's just what we have provided through this giveaway.

 24-bit isn't for everyone, much like vinyl it takes some time and investment, but for those who are interested the results are beautiful. And when the technology catches up, and storage prices drop, you won't have to buy your music over again! 
	

	
	
		
		

		
		
	


	




 Jim - Linn Records


----------



## leeperry

ok thanks for the clarifications...but HDCD is not true 24bit, it's actually dithered 20bit AFAIK?

 and saying that the only 16/24bit difference makes a very much audible difference in this very thread might require some more informations from your side


----------



## Jim Collinson

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_ok thanks for the clarifications...but HDCD is not true 24bit, it's actually dithered 20bit AFAIK?

 and saying that the only 16/24bit difference makes a very much audible difference in this very thread might require some more informations from your side 
	

	
	
		
		

		
		
	


	


_

 

Not quite sure what you mean here! The 16-bit file comes from the HDCD master, and the 24-bit has its own session.

 Ultimately the only thing that really matters in music is an audible difference surely! I'm not sure if most people care about a visual difference to an audio waveform! :
	

	
	
		
		

		
		
	


	




 This has been a very interesting and enjoyable thread, but some of the initial statements on which it was borne are slightly, well, wrong. We going to put together a document setting the case for 24-bit, from an engineering/science end. Just give us a little time - we have a packed recording and release schedule.

 The proof of the pudding is always in the listening (and enjoyment) and is always subjective. Hence the 24-bit comparison series we have started.

 The objectivity we will get to later,

 Jim - Linn Records


----------



## xnor

Quote:


  Originally Posted by *Jim Collinson* /img/forum/go_quote.gif 
_A bit harsh! We have been making audiophile recordings since 1983 and take it very seriously._

 

Yeah maybe that's a bit harsh, but I didn't want to doubt your competence or skills in any way! /apologizes

 The reason, that I was so skeptical, is that there was no information about how these files were "created" and from which source(s). (maybe I've overlooked something?) Also the post looked a bit like advertisement. 
	

	
	
		
		

		
		
	


	




 As you've said, both files were mastered differently so it's not a genuine test "to hear the difference between 16 and 24bit" (as stated on that soundcloud.com page). To really just compare 16 to 24 bits, the files should have a) the same source, b) same sampling rate and c) the conversion process should be specified.

 Nevertheless, I enjoyed listening those samples 
	

	
	
		
		

		
		
	


	




 and I usually don't listen to jazz.


----------



## leeperry

Quote:


  Originally Posted by *Jim Collinson* /img/forum/go_quote.gif 
_Not quite sure what you mean here! The 16-bit file comes from the HDCD master, and the 24-bit has its own session._

 

oh ok..so it's not 16bit either 
	

	
	
		
		

		
		
	


	




 as HDCD uses the 2 bottom bits of CDDA infos to encode HDCD 20bit dithered data AFAIK...and also the audio is +6dB louder than when it's properly HDCD decoded.

 so this indeed seems to be far from a 16/24bit genuine comparison?! it'd appear to actually be a 14/24 bit comparisons w/ different masterings....so please keep us posted whenever you will be able to provide us w/ genuine 16/24bit files dithered from the same exact 32float source w/ the same exact mastering/dithering algorithms.

 that's what annoys the OP of this thread, companies push 24bit....but the mastering is much different, and in that specific case it's even worse because your 16bit file has been HDCD encoded so it's not even true 16bit


----------



## barleyguy

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_oh ok..so it's not 16bit either 
	

	
	
		
		

		
			





 as HDCD uses the 2 bottom bits of CDDA infos to encode HDCD 20bit dithered data AFAIK...and also the audio is +6dB louder than when it's properly HDCD decoded.

 so this indeed seems to be far from a 16/24bit genuine comparison?! it'd appear to actually be a 14/24 bit comparisons w/ different masterings....so please keep us posted whenever you will be able to provide us w/ genuine 16/24bit files dithered from the same exact 32float source w/ the same exact mastering/dithering algorithms.

 that's what annoys the OP of this thread, companies push 24bit....but the mastering is much different, and in that specific case it's even worse because your 16bit file has been HDCD encoded so it's not even true 16bit 
	

	
	
		
		

		
		
	


	


_

 

From a technical perspective, I agree this is not accurate. But from a practical perspective, this is really what happens between a 16-bit and 24-bit version of a recording in the real world. Things are recorded and mixed at 24-bit. But when a 16-bit version is released, compression or limiting is used, which effectively removes the top 8 bits. When 24-bit releases are done, the top 8 bits are left intact, so what is given is a recording with more dynamic range.

 What you are asking for is either a 16-bit recording mastered without compression, or a 24-bit recording that is compressed to 16-bit standards. Neither of these happens in the real world. For a technical comparison between the two formats I understand why you're asking. But if you want a comparison between what is released in 16-bit and what is released in 24-bit, that's exactly what is being given.


----------



## leeperry

...so we'll never be able to compare 16/24bit properly.

 I've got some mind blowing 24/96 5.1 lossless recordings, but god knows whether they sound "better" than CDDA due to higher standards mastering or increased resolution?

 so it's like cable debates, there will never be a final answer...so let ppl believe what they like, what matters in the end is end users enjoyment after all 
	

	
	
		
		

		
		
	


	




 at some point I had some 32float masters that I got from Cubase VST, and 16bit sounded much less refined than 24bit...but then the OP made lenghtly explanations to prove us that 24bit is m00t, so I blamed it on the lousy SRC/dithering algorithms(UV22HR). ah well 
	

	
	
		
		

		
		
	


	




 I tend to believe that a properly dithered CDDA can almost sound as good as 24/96..I've got a MFSL remastered CD from Jean-Michel Jarre that just sounds too good to be true, you wouldn't believe that it's silly 16bit 
	

	
	
		
		

		
		
	


	




 the same way properly dithered RGB24 can look almost as good as RGB36


----------



## tosehee

The whole idea of OP, if I understanding from reading this entire thread, is that the original source cannot record anything higher than 20khz. There is no mic that can capture that, and our hearing capability is not beyond 60db in most cases. The dynamic range that is offered by 24bit is really not important factor since 16 bit has enough room to destroy our ears.

 24bit/96 or even 192khz is good for mixing and engineering process for extra rooms. Similar to why we are using the raw format in digital photos, but since the output cannot be reproduced, there really is no point of using 24/96+ in the actual listening format.

 For those who listens very loud to a point where it will destroy your ear in a short period of time, maybe having that extra 24bit might help a bit. But those like me who listens at a comfortable volume level will find the 16/44.1 perform exactly same as 24/96+.

 Isn't this what OP is saying?

 p.s.: is f2k with dithering the best way to down sample 24/96+ materials to 16/44.1?


----------



## barleyguy

Quote:


  Originally Posted by *tosehee* /img/forum/go_quote.gif 
_24bit/96 or even 192khz is good for mixing and engineering process for extra rooms. Similar to why we are using the raw format in digital photos, but since the output cannot be reproduced, there really is no point of using 24/96+ in the actual listening format.

 <snip>

 Isn't this what OP is saying?_

 

Yes, I think that is what he is trying to say. But from a practical perspective (rather than a purely technical one), 24-bit releases have 8 more bits of dynamic range. And those 8 bits are on the top, not the bottom. What he is saying is "if 16-bit releases were the same as 24-bit releases, they would sound the same." But the fact is, they aren't the same. If a studio released a 16-bit CD with 8 bits of extra dynamic range, it would be 48 Db or so quieter than all of your other CD's. People would ask "why is this broken?". (And it would be, because only the bottom 8 bits, or 6 bits after dithering [bits 3-8], would contain any music other than transients.) Therefore, the only releases with 24 bits of dynamic range are 24-bit releases.


----------



## tosehee

Quote:


  Originally Posted by *barleyguy* /img/forum/go_quote.gif 
_Yes, I think that is what he is trying to say. But from a practical perspective (rather than a purely technical one), 24-bit releases have 8 more bits of dynamic range. And those 8 bits are on the top, not the bottom. What he is saying is "if 16-bit releases were the same as 24-bit releases, they would sound the same." But the fact is, they aren't the same. If a studio released a 16-bit CD with 8 bits of extra dynamic range, it would be 48 Db or so quieter than all of your other CD's. People would ask "why is this broken?". (And it would be, because only the bottom 8 bits, or 6 bits after dithering [bits 3-8], would contain any music other than transients.) Therefore, the only releases with 24 bits of dynamic range are 24-bit releases._

 

Yes. But that 8 bit more of dynamic range is not within the domain of our hearing capability. Isn't that what OP is saying? What am I missing here?


----------



## barleyguy

Quote:


  Originally Posted by *tosehee* /img/forum/go_quote.gif 
_Yes. But that 8 bit more of dynamic range is not within the domain of our hearing capability. Isn't that what OP is saying? What am I missing here?_

 

If those 8 bits were on the bottom (least significant), they would be out of our hearing capability. But they're not. They're on the top, and they get removed with compression or limiting. (Side note: Limiting==fast, compression==slow.) So it does in fact sound different, because of the compression or limiting.


----------



## tosehee

Quote:


  Originally Posted by *barleyguy* /img/forum/go_quote.gif 
_If those 8 bits were on the bottom (least significant), they would be out of our hearing capability. But they're not. They're on the top, and they get removed with compression or limiting. (Side note: Limiting==fast, compression==slow.) So it does in fact sound different, because of the compression or limiting._

 

I read it as the upper limit is also not audible after 60db+ unless you wanna destroy your ears.


----------



## DoYouRight

So either way, would a vinyl rip to flac downverted to 16 sound identical on a higher end dac?


----------



## barleyguy

Quote:


  Originally Posted by *tosehee* /img/forum/go_quote.gif 
_I read it as the upper limit is also not audible after 60db+ unless you wanna destroy your ears._

 

I don't think you're understanding what I'm saying....

 When 16-bit audio is produced, it is generally recorded and mixed at 24-bit. When the conversion to 16-bit is done, it's not "let's do a straight conversion with dither, and remove the bottom 8 bits." It's "let's compress this until it sounds louder on a 16-bit medium, and remove the top 8 bits." So there are actual transients lost, and air and soundstage lost, and possibly artifacts from the compression.

 What is being asked as far as I can tell is, "if we did a straight conversion from 24-bit to 16-bit (removing the bottom 8 bits), would it sound the same?" The fact is, no commercial release of music is ever done that way. The standards and expectations for the two formats are different. So they sound different.

 EDIT: Another way to explain this is: You can't just throw away the top bits. If you do, you'll get a nasty clipping sound. You have to do something with them, and no matter what approach is taken, something is lost.

  Quote:


  Originally Posted by *DoYouRight* /img/forum/go_quote.gif 
_So either way, would a vinyl rip to flac downverted to 16 sound identical on a higher end dac?_

 

It depends how you do the downconversion. That's my whole point. If you record it into a 24-bit ADC, you can choose (during the conversion) to leave the dynamics intact and have it be really quiet compared to other 16 bit recordings, or you can do some sort of dynamic range compression (or simply normalization). The second choice may even be the best, because the original vinyl may not have enough dynamic range to need 24 bits.


----------



## leeperry

I think vynil is a different story because the more your quantize the crackles, the worst they will sound...but again it's source dependend, a very high quality vynil rip won't have hardly any audible crackles. but top notch vynil equipment costs an arm and a leg, plus you need a mint unplayed copy...and also need to clean it thoroughly, if it's been stocked for ages.


----------



## xnor

Quote:


  Originally Posted by *barleyguy* /img/forum/go_quote.gif 
_I don't think you're understanding what I'm saying....

 When 16-bit audio is produced, it is generally recorded and mixed at 24-bit. When the conversion to 16-bit is done, it's not "let's do a straight conversion with dither, and remove the bottom 8 bits." It's "let's compress this until it sounds louder on a 16-bit medium, and remove the top 8 bits." So there are actual transients lost, and air and soundstage lost, and possibly artifacts from the compression.

 What is being asked as far as I can tell is, "if we did a straight conversion from 24-bit to 16-bit (removing the bottom 8 bits), would it sound the same?" The fact is, no commercial release of music is ever done that way. The standards and expectations for the two formats are different. So they sound different.

 EDIT: Another way to explain this is: You can't just throw away the top bits. If you do, you'll get a nasty clipping sound. You have to do something with them, and no matter what approach is taken, something is lost._

 

I don't know if you understand what's going on.

 A single sample usually is represented by a floating point number in the range [0; 1].
 (0.0 is silence, 1.0 is clipping)
 Floats normally have 32 bits, so they allow a pretty fine resolution.

 Reducing the number of bits to represent these sample values just reduces the resolution (in other words: the distance between two consecutive numbers increases).

 As the OP stated, this doesn't make the sound worse, since the original waveform can be restored perfectly using dithering.

 You also say "let's compress this until it sounds louder on a 16-bit medium, and remove the top 8 bits.". Isn't that sheer nonsense? Why should there be a change in volume? (remember: the dynamic range changes and dynamic range != music volume)
 Why on earth would somebody make the samples much louder and cut off upper bits? That would result in one big clipping mess.


----------



## barleyguy

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_I don't know if you understand what's going on.

 A single sample usually is represented by a floating point number in the range [0; 1].
 (0.0 is silence, 1.0 is clipping)
 Floats normally have 32 bits, so they allow a pretty fine resolution.

 Reducing the number of bits to represent these sample values just reduces the resolution (in other words: the distance between two consecutive numbers increases).

 As the OP stated, this doesn't make the sound worse, since the original waveform can be restored perfectly using dithering.

 You also say "let's compress this until it sounds louder on a 16-bit medium, and remove the top 8 bits.". Isn't that sheer nonsense? Why should there be a change in volume? (remember: the dynamic range changes and dynamic range != music volume)
 Why on earth would somebody make the samples much louder and cut off upper bits? That would result in one big clipping mess._

 

I didn't say "cut off". I said "remove" and also "compress and limit". Which really is what is done in the conversion.

 This all has to do with the history of the 16-bit medium, and the recent history of the 24-bit medium. When CD's were released, they were mostly conversions from analog recordings. They had a relative volume and dynamic range compression similar to that analog medium. Then, as CD's became mainstream, there was a "loudness war", where producers and record companies compressed the crap out of everything to get the "loudest" recordings. (Of course the actual volume is controlled by the volume knob, but still, there was a loudness war. As in, make a particular CD sound louder at the same knob setting as another CD.) Thus, the standards and expectations for 16-bit recordings entail having a certain level, which is only about 10 Db RMS from full scale.

 24-bit recordings are typically audiophile targeted, and so they have that headroom and dynamic range restored closer to the original studio recording. A recording with more headroom has more room for transients and soundstage, so it should inherently sound better.

 I understand perfectly what you're saying. What I'm saying is that you won't see any music that is released like that, because the two formats have different expectations.


----------



## xnor

@barleyguy: Ok, I misunderstood you there.
 But still, what the OP said is true. You cannot hear the difference between a 24bit and 16bit file, except something has been "done" to it.

 So these 24-bit recordings are just another way to make more money..
 Guess they are selling expensive 32-bit recordings in a couple of years hehehe


----------



## xnor

Quote:


  Originally Posted by *DoYouRight* /img/forum/go_quote.gif 
_So either way, would a vinyl rip to flac downverted to 16 sound identical on a higher end dac?_

 

Basically, yes.


  Quote:


  Originally Posted by *barleyguy* /img/forum/go_quote.gif 
_It depends how you do the downconversion. That's my whole point. If you record it into a 24-bit ADC, you can choose (during the conversion) to leave the dynamics intact and have it be really quiet compared to other 16 bit recordings, or you can do some sort of dynamic range compression (or simply normalization). The second choice may even be the best, because the original vinyl may not have enough dynamic range to need 24 bits._

 

No, it isn't going to be really quiet?!
 Second choice is evil and what record companies are doing. (which is very sad)


----------



## barleyguy

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_@barleyguy: Ok, I misunderstood you there.
 But still, what the OP said is true. You cannot hear the difference between a 24bit and 16bit file, except something has been "done" to it.

 So these 24-bit recordings are just another way to make more money.._

 

Yes, that true. But, it's not some big conspiracy. It is about giving people what they expect. If you released a 16-bit recording with the proper amount of headroom, it would be quieter at the same volume setting as everything else in someone's collection. With 24-bit, the standard has been to give that headroom back resulting in a better sounding recording.

 Often the market has more to do with giving people what they expect than giving them what's best. People in general don't understand things well enough to know what's best.

 The big question behind this topic seems to be "Do 24-bit recordings sound better than 16-bit recordings?"

 Yes, they do. But not for the reason you would expect. They sound better because they are mastered differently.


----------



## leeperry

also for the fact that every -6dB you lose 1 bit...so, some very quiet CD recording would be less than 10bit.

 not an excuse for the loudness war, of course..24bit allows far more headroom, especially useful when recording.


----------



## tosehee

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_also for the fact that every -6dB you lose 1 bit...so, some very quiet CD recording would be less than 10bit.

 not an excuse for the loudness war, of course..24bit allows far more headroom, especially useful when recording._

 

Can't you just increase the volume then? Why is this an issue? 

 As for other comment where 24bit is mastered differently, I find that little unease. I was under the impression that the entire engineering/mixing part is done in 24bit/192khz or higher, and down sample to whatever format the buyers want. Be that CD or HDCD or whatever.

 Since the original engineering sample is same, the CD and HDCD should sound identical.

 Are you saying that they deliberately use different engineering/mixing for 16bit recording and 24bit? That sounds like extra work involved since they need to perform the same task twice for 16 and 24bit mastering.


----------



## barleyguy

Quote:


  Originally Posted by *tosehee* /img/forum/go_quote.gif 
_Can't you just increase the volume then? Why is this an issue?_

 

It's an issue because of market expectations. It's not a technical issue. People expect their recording to be a certain "volume". (Yes, they could just turn the knob to the right. But that's not what they expect to have to do.)

  Quote:


  Originally Posted by *tosehee* /img/forum/go_quote.gif 
_As for other comment where 24bit is mastered differently, I find that little unease. I was under the impression that the entire engineering/mixing part is done in 24bit/192khz or higher, and down sample to whatever format the buyers want. Be that CD or HDCD or whatever._

 

The mixing is done at native resolution. The mastering is targeted for whatever medium it is going to end up at.

  Quote:


  Originally Posted by *tosehee* /img/forum/go_quote.gif 
_Since the original engineering sample is same, the CD and HDCD should sound identical.

 Are you saying that they deliberately use different engineering/mixing for 16bit recording and 24bit? That sounds like extra work involved since they need to perform the same task twice for 16 and 24bit mastering._

 

Yes, if they are smart they will do two different masters for 16-bit and 24-bit. (And the posts earlier in this thread from a record company person confirm this.) This is mastering, not mixing. They are two different steps. The 16-bit master will have the level that normal consumers expect, and the 24-bit master will have additional headroom for audiophiles. (There are exceptions to this rule. Many classical recordings, for example, have headroom even at 16-bit, and classical consumers are accustomed to turning the volume knob to compensate.)


----------



## xnor

Quote:


  Originally Posted by *leeperry* /img/forum/go_quote.gif 
_also for the fact that every -6dB you lose 1 bit...so, some very quiet CD recording would be less than 10bit.

 not an excuse for the loudness war, of course..24bit allows far more headroom, especially useful when recording._

 

no... read the OP's post
 "each extra bit of data moves the noise floor down by 6dB (half). We can turn this around and say that each bit of data provides 6dB of dynamic range"

 so for every bit you remove, you move the noise floor up... (which you hear at over ~150 db on a normal CD, as the OP explained too)


 I don't think that the volume of the tracks is in any way related to what the OP described.
 As barleyguy explained before, the compression (thus loss of dynamics) is due to the loudness war, and nothing else.

 edit: and so the final conclusion is:
 Listening to a 24 bit song in 16 bit mode = no difference.


----------



## xnor

Seriously, on the 28th page people still don't understand what the OP said.. no wonder that he left ...


----------



## barleyguy

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_edit: and so the final conclusion is:
 Listening to a 24 bit song in 16 bit mode = no difference._

 

Generally true. But there are very specific examples where I'd argue that you could hear a difference. Let's suppose, for example, that there is a 5-second fade that fades from 7-bits to the noise floor of 2-bits on a 16-bit medium. That fade out will only have 32 volume levels. The same fade out on a 24-bit medium, the same distance from full scale, will have 8192 volume levels. It's possible that something like that could be heard.

 Remember that we are talking about volume levels, not frequency. Harder to detect and less critical for the most part.

 (Sorry to throw another point of contention in the mix. 
	

	
	
		
		

		
			





)


----------



## Arjisme

Quote:


  Originally Posted by *barleyguy* /img/forum/go_quote.gif 
_The big question behind this topic seems to be "Do 24-bit recordings sound better than 16-bit recordings?"_

 

That is indeed a good question, but is not the topic of this thread. This thread was started to explain whether or not there is a technical advantage in the choice of bit depth. That is, does 24 bit innately result in superior audio as opposed to 16 bit? The OP makes the case that it does not because the differences are either outside the audible range or are not usable w/o endangering ears & lives.

 As you've pointed out (and I think it was pointed out earlier in the thread too), mastering differences do occur and would explain the audible differences between the two choices.


----------



## leeperry

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_for every bit you remove, you move the noise floor up... (which you hear at over ~150 db on a normal CD, as the OP explained too)_

 

humm...moving the noise floor up means losing bitdepth resolution in my book.

 each time you lose -6dB from 0dB, you lose 1 bit resolution...reason why some very low parts in some old non-"loudness war" material sounds wonky...you're actually listening to 8bit resolution audio or so 
	

	
	
		
		

		
		
	


	




 a CD at 0dB offers 96dB of resolution, 16*6...so you can afford low level volumes in 24bit, not 16.
  Quote:


  Originally Posted by *barleyguy* /img/forum/go_quote.gif 
_It's an issue because of market expectations. It's not a technical issue. People expect their recording to be a certain "volume". (Yes, they could just turn the knob to the right. But that's not what they expect to have to do.)_

 

to the human ear: louder=better

 reason why headphones/speakers comparisons w/o a SPL meter are m00t...records are being clipped to please the human ear, the untrained human ear I should say. if it's loud, it's good!11!!


----------



## jcx

I see the dither discussion was lost on a few folks


----------



## xnor

Quote:


  Originally Posted by *barleyguy* /img/forum/go_quote.gif 
_Generally true. But there are very specific examples where I'd argue that you could hear a difference. Let's suppose, for example, that there is a 5-second fade that fades from 7-bits to the noise floor of 2-bits on a 16-bit medium. That fade out will only have 32 volume levels. The same fade out on a 24-bit medium, the same distance from full scale, will have 8192 volume levels. It's possible that something like that could be heard.

 Remember that we are talking about volume levels, not frequency. Harder to detect and less critical for the most part.

 (Sorry to throw another point of contention in the mix. 
	

	
	
		
		

		
		
	


	




)_

 

.. as the OP wrote 
	

	
	
		
		

		
		
	


	




 even complex orchestra recordings usually don't exceed a dynamic range of 60 db, so that example doesn't exist in practice, because (16-2) * -6 equals -84 db. Good luck doing ABX tests on a volume that would allow you to hear that (at least for a minute, or two) ... 
	

	
	
		
		

		
		
	


	




 Besides, 2^2 = 4 and 2^7 = 128 so we have 128 - 4 = 124 values, which would give a resolution of 1 / 124 = 0.008

 That's all _ I _ can do for you.
 /unsubscribes 
	

	
	
		
		

		
		
	


	




 edit: @leeperry: If you were bashing your head against a wall and I'd say, "Stop it, that damages your brain.", you'd bash even harder, it seems.


----------



## tosehee

What xnor wrote is what I was gonna say. Even the most complex orchestra recordings wouldn't extend beyond 60db. 

 Other than the fact that recording studios intentionally make the 16bit CD worse than the 24bit/HDCDs, there is no other explanation as to why we should prefer 24/96+ over 16/44.1


----------



## leeperry

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_@leeperry: If you were bashing your head against a wall and I'd say, "Stop it, that damages your brain.", you'd bash even harder, it seems._

 

hehe OK, I guess I'm overlooking something...the dB scale being logarithmic prolly changes the whole story 
	

	
	
		
		

		
		
	


	




 anyway, I'll take the highest quality stream the studio was kind enough to offer...I thing that's the best advice one's could give.

 TrueHD 7.1 24/48 lossless sounds out of this world! compressing it to DTS is like doing some 128kbit MP3 encoding, and downscaling to lossless 16/48 also makes the sound kinda lacking focus, and more "hollow"....but again I used very basic downscaling in eac3to, not some top of the line dithering..which would prolly sound WAY better 
	

	
	
		
		

		
		
	


	




 dithering is what makes 16bit worthy, and noise shaping too..you can play around w/ both in ffdshow audio(between 32FP & 16/24/32int), not that I can hear a difference though


----------



## barleyguy

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_.. as the OP wrote 
	

	
	
		
		

		
		
	


	




 even complex orchestra recordings usually don't exceed a dynamic range of 60 db, so that example doesn't exist in practice, because (16-2) * -6 equals -84 db. Good luck doing ABX tests on a volume that would allow you to hear that (at least for a minute, or two) ... 
	

	
	
		
		

		
		
	


	


_

 

This is assuming that the upper bits actually contain sound. If you've got a low level fade on a song that is quiet all the way through, this could actually happen. And you could possibly have it turned up loud enough to hear it.

  Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_Besides, 2^2 = 4 and 2^7 = 128 so we have 128 - 4 = 124 values, which would give a resolution of 1 / 124 = 0.008_

 

Since the you're starting at 7 bits, and the bottom 2 bits are noise, I did it as 2^5 = 32.


----------



## Jim Collinson

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_I don't know if you understand what's going on.

 A single sample usually is represented by a floating point number in the range [0; 1].
 (0.0 is silence, 1.0 is clipping)
 Floats normally have 32 bits, so they allow a pretty fine resolution.

 Reducing the number of bits to represent these sample values just reduces the resolution (in other words: the distance between two consecutive numbers increases).

 As the OP stated, this doesn't make the sound worse, since the original waveform can be restored perfectly using dithering._

 

This is the fundamental flaw in the OP's argument, and really the whole thread has flowed from it. 

 In reality, dithering does not reproduce a waveform perfectly. If this statement was true, and perfect dither was possible, then you would be able to produce a system with infinite resolution regardless of the bit depth.

 Jim - Linn Records


----------



## D. Lundberg

Quote:


  Originally Posted by *Jim Collinson* /img/forum/go_quote.gif 
_In reality, dithering does not reproduce a waveform perfectly._

 

Dither does not reproduce waveforms at all, it is used to decorrelate quantization errors from a signal.

  Quote:


 If this statement was true, and perfect dither was possible, then you would be able to produce a system with infinite resolution regardless of the bit depth. 
 

"[...] if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white,
 uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the channel – and that is that!"
 -- _J. Robert Stuart, Coding High Quality Digital Audio, Meridian Audio Ltd._

 I'm not sure what you mean by "resolution" in this context, but you can perfectly recreate waveforms regardless of bit dept (as long as you follow the rules of the Nyquist theorem). 

 Differences in bit depth will be in how much white noise you'll get in addition to the signal. 
 So bit depth is just a measure of statistical resolution, and ultimately the dynamic range of the signal.


----------



## Jim Collinson

Quote:


  Originally Posted by *D. Lundberg* /img/forum/go_quote.gif 
_Dither does not reproduce waveforms at all, it is used to decorrelate quantization errors from a signal._

 

Yes, I should have been more clear there.

  Quote:


  Originally Posted by *D. Lundberg* /img/forum/go_quote.gif 
_"[...] if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white,
 uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the channel – and that is that!"
 -- J. Robert Stuart, Coding High Quality Digital Audio, Meridian Audio Ltd._

 

Robert makes a pretty good case for higher than 16/44.1 bit-depth and sample rate in that paper. 

 Jim


----------



## sirmasterboy

I understand that more bits doesn't mean higher quality. It's obvious because DSD is only a 1bit signal yet it's much higher fidelity than 16bit CD.

 My question to this is how does a 1bit DSD recorded stream compare then to these 16bit or 24bit streams.

Direct Stream Digital - Wikipedia, the free encyclopedia


----------



## Jim Collinson

Hello Everyone,

 For anyone who is interested, the second in our series of 24-bit comparisions has gone out through Twitter. This time we even have a 192kHz Studio Master!

 This one is from the *Beethoven Piano Concertos by the Scottish Chamber Orchestra, with Sir Charles Mackerras and Artur Pizarro*.

 Download the 16-bit/44.1kHz, 24-bit/96kHz and the 24-bit/192kHz for free and compare.

 Please bear in mind that these are the files exactly as we sell them for download on the store - and for the 16-bit, exactly as it is sold on hybrid SACD. Hence the reason we are calling it a real-world test.

 When reporting back in, let us know your listening setup. But most importantly, ENJOY!

 Jim - Linn Records


----------



## tosehee

Thanks, Jim for a superb piano concerto.

 It was so peaceful that I almost fell asleep while trying to AB between the two files.

 My system do not support 192khz, so my AB testing was between 16/44.1 vs. 24/96khz.

 My setup is listed on my signature, so no need repeating here.

 I have done repeated tests hearing from one passage from start to finish, and several places where I thought I hear the differences in multiple times.

 Without going out in full detail, here is what I got to stay.

 When I AB between two files after several minutes in between, I could tell apart very much. That's being my honest impression. However, if I listen right after the other, then I notice more lively, less fatigue sound from 24/96. I have no clue if that's just a placebo or ego here.

 I was leaning toward OP, but after hearing this X number of times, I can't explain, but I favor the 24/96 by a small margin. The base seems to be slightly fuller, the piano is less harsh, and overall less fatigue in 24/96. Other than that, I can't notice a thing. 

 If done blind test, I don't know how many can actually distinguish one from the other. They are both exceptional quality. If I must though, I must say that I favor the 24/96 a small margin.


----------



## DoYouRight

alot of that might have to do with an affinity for your dac to be geared toward one than the other. if it upsamples and such.


----------



## grawk

That's a comparison of different masters, rather than a comparison of just sample and bit rates, of course, and a great marketing tool. Give me the original masters, and I could do the sample rate conversion in reverse and convince people that 44/16 was better than 96/24, I'd wager...


----------



## s4nder

I haven't read the whole thread but it seems to me that the only reason to go beyond 44,1 kHz sampling rate is if frequencies higher than 22 kHz have some effect on sound quality. 

 I've read that some instruments like the violin can have harmonics as high as 40 kHz or more and they affect the timbre of the sound. So basically a violin in a 192 kHz recording would sound slightly more realistic due to these subtle timbre and spatial cues.

 While humans can not directly hear frequencies over 20 kHz, they can still be perceived through skin and bones, so there might be some truth to that. 

 There's a good video about sound quality and why it matters with big names from the industry in a round table. Find "Deep Listening: Why Audio Quality Matters" on this list and press "watch video." It lasts for two hours but is very interesting.


----------



## jnewman

Gregorio,
 Thank you for taking the time to post throughout this thread. It was quite an interesting read and I didn't know as much about digital audio as I thought I did.


----------



## charlie0904

excellent read. i will not spend so much on audio from now on.


----------



## b0dhi

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_Seriously, on the 28th page people still don't understand what the OP said.. no wonder that he left ..._

 

This is the actual reason.


----------



## nnotis

"So, can you actually hear any benefits of the larger (48dB) dynamic range offered by 24bit? Unfortunately, no you can't."

 Why then, with a lot of listening, can I distinguish my own 24 bit master tracks from the dithered 16 bit versions in a blind test? Perhaps I’ve just learned to pick out the noise shaping. Or perhaps it’s something more. Subjectively, the dithered tracks never sound quite as good as the originals.

 I’m not disputing the theory you’ve articulated, as I wouldn’t dispute the Nyquist frequency. Unfortunately, in practice, these things turn out to be more complicated than any of us would like. In the case of Nyquist, roll off below the cut off frequency leads many to sample at extremely high rates. I have no explanation for why 16 bit files don’t sound as good as the 24 bit originals, but in practice, they don’t. Mind you, the differences are extremely subtle.


----------



## nick_charles

Quote:


  Originally Posted by *nnotis* /img/forum/go_quote.gif 
_"Why then, with a lot of listening, can I distinguish my own 24 bit master tracks from the dithered 16 bit versions in a blind test? ._

 

Can you post a couple of short samples of 16 and 24 bit verions ?


----------



## shigzeo

I've never heard 24-bit postulated over 16-bit other than in audiophile circles. But, that is beside the fact. We will never, ever have a proper debate here at head-fi about anything which is remotely fact/fiction, the platform simply doesn't exist. In fact, internet isn't a good place to debate because people are too emotional as they suck down coffee (or in my case, alcohol) and argue.

 One of the problems here is that many people came in to prove something which I will leave out of fact, or fiction. Anything can be proved, whether it be 16 bit or 24 bit is superior; whether analogue or digital; whether bose or skullcandy - everything can be proved by debate, and depending on who has the weaker argument, or the quieter voice.

 If Gregorio got flack, I would imagine it has nothing to do with reality, just the same as his or anyone's arguments are based on heat and temper; above all a desire to prove someone else wrong and oneself right; to appear more knowledgeable is paramount. 

 Nice read though


----------



## kwkarth

Quote:


  Originally Posted by *shigzeo* /img/forum/go_quote.gif 
_I've never heard 24-bit postulated over 16-bit other than in audiophile circles. But, that is beside the fact. We will never, ever have a proper debate here at head-fi about anything which is remotely fact/fiction, the platform simply doesn't exist. In fact, internet isn't a good place to debate because people are too emotional as they suck down coffee (or in my case, alcohol) and argue.

 One of the problems here is that many people came in to prove something which I will leave out of fact, or fiction. Anything can be proved, whether it be 16 bit or 24 bit is superior; whether analogue or digital; whether bose or skullcandy - everything can be proved by debate, and depending on who has the weaker argument, or the quieter voice.

 If Gregorio got flack, I would imagine it has nothing to do with reality, just the same as his or anyone's arguments are based on heat and temper; above all a desire to prove someone else wrong and oneself right; to appear more knowledgeable is paramount. 

 Nice read though_

 

Hi,
 Just postulating here... There are some here that freely admit we do not know everything there is to know in ANY given field of endeavor. I will go so far as to say that while I have compiled a body of knowledge, and even some wisdom in selected fields of study, I am fundamentally a student at heart and forever will be. I can and do learn from virtually everyone I interact with. I am sure that there are many others who frequent these forums who feel the same way.

 An intellectually honest discussion places some responsibilities on its participants. Some participants accept those responsibilities, some do not.
 We can even learn from those who are too immature or intellectually dishonest to accept those responsibilities.

 BTW, the term *debate* assumes opposing sides of an argument, whereas *discussion* assumes more of a position of neutrality in the quest of greater knowledge.


----------



## xnor

Quote:


  Originally Posted by *b0dhi* /img/forum/go_quote.gif 
_This is the actual reason._

 

And you even supported that, you should be ashamed of yourself.

 Btw, does this forum even have an ignore feature?


 On topic: There's no reason to debate/discuss this anymore.

 The only difference you hear is the difference between different masters. This has been shown, I've tested it (and provide the files free for download if you want to try it yourself) and we got confirmation that the files are mastered differently.
 Nothing more to add.


 So who has lost the debate, in several aspects?
 Not gregorio's side, that's for sure.


----------



## kwkarth

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_And you even supported that, you should be ashamed of yourself.

 Btw, does this forum even have an ignore feature?


 On topic: There's no reason to debate/discuss this anymore.

 The only difference you hear is the difference between different masters. This has been shown, I've tested it (and provide the files free for download if you want to try it yourself) and we got confirmation that the files are mastered differently.
 Nothing more to add.


*So who has lost the debate, in several aspects?
 Not gregorio's side, that's for sure.*




_

 

Well, if this thread was a debate and not a discussion, then everybody lost with that mindset. Head-Fi is first and foremost a discussion forum, not a debate forum. If there were anybody on the planet in possession of all knowledge on a given subject, I would listen to them, but soon even I would tire of listening to the soliloquy eventually, and probably not adsorb everything I read because sometimes I need to see things form different perspectives before they sink in. Such is learning.


----------



## nnotis

Quote:


  Originally Posted by *nick_charles* /img/forum/go_quote.gif 
_Can you post a couple of short samples of 16 and 24 bit verions ?_

 

I will try to find time to to do this sometime soon. Then, so long as I can be trusted to provide files that are equal outside of the bit depth, dithering, and noise shaping, people can decide for themselves.

 I posted files like this once before, and I recall that you did some interesting analysis of the two versions using Cool Edit or something like it. I believe that with the 16 bit version, there was a smaller difference between the loudest and most quiet samples. But even the 24 bit file had a range well within 96 decibels. Could this difference have been due to the noise shaping?


----------



## nnotis

Quote:


  Originally Posted by *shigzeo* /img/forum/go_quote.gif 
_I've never heard 24-bit postulated over 16-bit other than in audiophile circles._

 

Check out the mastering forum on Gearslutz.com - Powered by vBulletin. There's lots of in depth discussion about this being done there by people who would most certainly not consider themselves audiophiles.


----------



## leeperry

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_And you even supported that, you should be ashamed of yourself._

 

yeah, I'm pretty shocked that such b*tchy personal attacks have been tolerated on this forum


----------



## nwkid178

Wow very interesting post, but a question then, say you were to have an album ripped and stored as flac in 24 bit with a sound sample size of 96.000khz.. how could i change this so it is playable on my ipod classic, as I have run into this issue before when experimenting with, mostly "high res" albums before, mostly ripped from vinyl, which I have not tried doing for myself yet.


----------



## grawk

You'll need something to convert flac to alac, aiff, mp3, or aac and you need something to convert it from 96/24 to 48/16. dbpoweramp can handle the format conversion on pc, max on windows. Which software to do the sample rate conversion is something of a religious debate, so I'll let others chime in.


----------



## Justice Strike

people tend to forget that everything is relative. Ofcourse you wouldn't get 48db more pressure as you would just turn down the music. However, instead of a larger dynamic range you could just have more resolution in the sound.

 I can best describe it by taking the 48bit images and compare them with 32 or 24 bit images. Yes you might not see the difference. But there is more detail in the image which can be used in post processing... or could be used by compressing dynamic range without sacrificing resolution.

 ofcourse postprocessing is a really good reason to have 24 bit audio, which is something that is especially active for normal setups which need some post processing to make the audio fit the room.


----------



## Ashirgo

You haven't read the whole thread. This claim has been dismissed many times so far. Unfortunately, the comparison to the bit depth of pictures does not hold water; it is still true, that this additional dynamic range is useful for sound processing - but for some other reasons (as it has been mentioned.............)


----------



## leeperry

24/96 has been agreed to sound better than 16/44.1 sometimes, but noone really knows why basically 
	

	
	
		
		

		
		
	


	




 some high quality dithering algorithm to CDDA might sound very similar to 24/96...just like dithering for video works indeed.

 16bit dithered to 8 might look just as good as 12bit, it's been more or less proven on doom9.


----------



## Justice Strike

Quote:


  Originally Posted by *Ashirgo* /img/forum/go_quote.gif 
_You haven't read the whole thread. This claim has been dismissed many times so far. Unfortunately, the comparison to the bit depth of pictures does not hold water; it is still true, that this additional dynamic range is useful for sound processing - but for some other reasons (as it has been mentioned.............)_

 

it does hold water.... if you think it doesn't you should really explain why.


----------



## grawk

it's been explained repeatedly in this very thread. 

 The short answer: the bits mean the numbers get bigger, not that the numbers have more details in the middle. The added bits mean it handles more volume below -96db.


----------



## Xel'Naga

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_it's been explained repeatedly in this very thread. 

 The short answer: the bits mean the numbers get bigger, not that the numbers have more details in the middle. The added bits mean it handles more volume below -96db._

 

Read this: http://www.head-fi.org/forums/f133/b...lained-455688/. I didn't go much into why higher bit-depth is better, because I thought it's obvious. Apparently not...
 I guess a few pictures will show the difference.




 Musical Fidelity V-DAC, waveform of undithered 1kHz sinewave at –90.31dBFS, CD data (left channel blue, right red).




 Musical Fidelity V-DAC, waveform of undithered 1kHz sinewave at –90.31dBFS, 24-bit data (left channel blue, right red).
 Bottom of the line, the more bits you have the bettter you can represent the sample amplitude, which is a real number.




 Consider how this picture would look with 1 bit depth and then with 16. With 16, the difference between the digital representation and the red sine wave would be invisible to the naked eye at this scale.


----------



## emmodad

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_it's been explained repeatedly in this very thread. 

 The short answer: the bits mean the numbers get bigger, not that the numbers have more details in the middle. The added bits mean it handles more volume below -96db._

 


 well, that's what the thread author seems to have implied.

 grawk and others perhaps interested in a bit of ..... clarification.... as to issues concerning the concept of resolution as presented in this (lengthy, and sadly often technically misinformative) thread,

 you may wish to take a look at Pohlmann for a not-overly-technical explanation of quantization step size and quantization error:

 Principles of Digital Audio
 Chapter 2, Fundamentals of Digital Audio
 pp31-32: Quantization
 pp32-37: Signal-to-error ratio
 pp37-39: Quantization distortion

 this chapter is available for free preview on google books; depending on the day / cache refresh timing / phases of the moon, various pages may not be included in the preview (on last check, 31-39 were all visible except 37; then later 37 showed up...):

Principles of digital audio - Google Books

 hth


----------



## TStewart422

It's not a question of whether or not it's more accurate, it obviously is...

 ... but is it AUDIBLE? All signs point to "NO."


----------



## Justice Strike

Quote:


  Originally Posted by *TStewart422* /img/forum/go_quote.gif 
_It's not a question of whether or not it's more accurate, it obviously is...

 ... but is it AUDIBLE? All signs point to "NO."_

 

well ofcourse that's a matter of how you use the information.

 you can use the bits in 2 ways. The example of Xel'Naga.shows one kind of way (the way i was trying to explain)

 However more range is also an option.

 it's the same as with normal numbers. You can use 6 digits to represent a distance. for example 100.000 km Now adding digits, you can choose to add digits to the big end or de little end. so adding digits can make your precision better 1.000.000,00 or you can get more range 100.000.000. Now, if you want to represent a distance from any place to any place on a map you could opt for range. But this has little value of your map is only 50 by 50 km. Your maximum distance will be (50^2+50^2)^0.5 so instead of using the digits to represent range you could also use it to represent precision.

 further more. This precision can be used when dealing with post processing (DSP and stuff) it will reduce the amount of information that is lost. 

 To illustrate this i want to give an example.

 Just think of having only 1 digit of precision. now plain d/a conversion would be ok. However with a dsp most likely multiplications and divisions will be done. now dividng 9 by 3 will give you a nie round number of 3. However thigns go astray when you decide to divide 2 by 3. we all know that the anser is 0.66666 rounded up it would have been a 1. However due to the 1 digit precision it would be represented as a 0. 

 To make a long story short. This can be solved in 2 ways. The easy and best way is to indroduce more bits to get rid of those rounding things. The second is to have a dsp which uses a higher number of bits internallY. It will sotre the number 0.6666 and will round it of when it gives output.


----------



## D. Lundberg

Quote:


  Originally Posted by *Xel'Naga* /img/forum/go_quote.gif 
_I guess a few pictures will show the difference.
 Musical Fidelity V-DAC, waveform of undithered 1kHz sinewave at –90.31dBFS, CD data (left channel blue, right red)._

 

Did you miss the word "undithered"?

  Quote:


 Bottom of the line, the more bits you have the bettter you can represent the sample amplitude, which is a real number. 
 

If the signal is properly dithered the complete signal, including the amplitude, can be accurately represented (thanks to the Nyquist theorem).

 "[...] if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white,
 uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the channel – and that is that!"
 -- J. Robert Stuart, Coding High Quality Digital Audio, Meridian Audio Ltd.

 More bits -> less quantization errors -> lower noise floor -> higher dynamic range.

  Quote:


 Consider how this picture would look with 1 bit depth and then with 16. With 16, the difference between the digital representation and the red sine wave would be invisible to the naked eye at this scale. 
 

It's not about what it looks like, but about what it represents. 
 Using 1 bit means you have a very limited dynamic range (roughly 6dB), but if you have plenty of bandwidth you can move most of the noise above 20kHz (noise shaping). That's how DSD/SACD and Delta/Sigma-converters work.


----------



## Justice Strike

Quote:


  Originally Posted by *D. Lundberg* /img/forum/go_quote.gif 
_Did you miss the word "undithered"?
 "[...] if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white,
 uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the channel – and that is that!"
 -- J. Robert Stuart, Coding High Quality Digital Audio, Meridian Audio Ltd.

 More bits -> less quantization errors -> lower noise floor -> higher dynamic range._

 

higher dynamic range or higher resolution, it's a choice.


  Quote:


 It's not about what it looks like, but about what it represents. 
 Using 1 bit means you have a very limited dynamic range (roughly 6dB) 
 

how do you figure that? why not 3? or 2 db or 50 db? who says that 1 bit only represents 6 db of dynamic range?

 I'll tell you a little secret. I can encode 100db in one bit.


  Quote:


 but if you have plenty of bandwidth you can move most of the noise above 20kHz (noise shaping). That's how DSD/SACD and Delta/Sigma-converters work. 
 

that made absolutely no sense to me what so ever. You are mixing up technologies.


----------



## Justice Strike

Quote:


  Originally Posted by *D. Lundberg* /img/forum/go_quote.gif 
_Did you miss the word "undithered"?
 "[...] if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white,
 uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the channel – and that is that!"
 -- J. Robert Stuart, Coding High Quality Digital Audio, Meridian Audio Ltd.

 More bits -> less quantization errors -> lower noise floor -> higher dynamic range._

 

higher dynamic range or higher resolution, it's a choice.


  Quote:


 It's not about what it looks like, but about what it represents. 
 Using 1 bit means you have a very limited dynamic range (roughly 6dB) 
 

how do you figure that? why not 3? or 2 db or 50 db? who says that 1 bit only represents 6 db of dynamic range?

 I'll tell you a little secret. I can encode 100db in one bit.


  Quote:


 but if you have plenty of bandwidth you can move most of the noise above 20kHz (noise shaping). That's how DSD/SACD and Delta/Sigma-converters work. 
 

that made absolutely no sense to me what so ever. You are mixing up technologies.


----------



## Justice Strike

Quote:


  Originally Posted by *D. Lundberg* /img/forum/go_quote.gif 
_Did you miss the word "undithered"?
 "[...] if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white,
 uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the channel – and that is that!"
 -- J. Robert Stuart, Coding High Quality Digital Audio, Meridian Audio Ltd.

 More bits -> less quantization errors -> lower noise floor -> higher dynamic range._

 

higher dynamic range or higher resolution, it's a choice.


  Quote:


 It's not about what it looks like, but about what it represents. 
 Using 1 bit means you have a very limited dynamic range (roughly 6dB) 
 

how do you figure that? why not 3? or 2 db or 50 db? who says that 1 bit only represents 6 db of dynamic range?

 I'll tell you a little secret. I can encode 100db in one bit.


  Quote:


 but if you have plenty of bandwidth you can move most of the noise above 20kHz (noise shaping). That's how DSD/SACD and Delta/Sigma-converters work. 
 

that made absolutely no sense to me what so ever. You are mixing up technologies.


----------



## D. Lundberg

Quote:


  Originally Posted by *Justice Strike* /img/forum/go_quote.gif 
_higher dynamic range or higher resolution, it's a choice._

 

It's not a choice, it's the same thing! The "resolution" is just a statistical measure of the amount of quantization errors you'll get in addition to the signal. 
 The errors can either be correlated (distortion) or not correlated (white noise) to the signal. And if the signal is properly dithered you'll get the second type of errors, and thus white noise.

 So the (statistical) resolution determines the noise floor and dynamic range, nothing more.

  Quote:


 how do you figure that? why not 3? or 2 db or 50 db? who says that 1 bit only represents 6 db of dynamic range? 
 

From the amount of quantization steps involved.
 If Q is the number of quantization steps and D is the dynamic range:

 Q = 2^(D/6), or D = 6 Log[size=xx-small]2[/size] Q

 1 bit = 2 quantization steps = 6 dB of dynamic range.
 16 bits = 65536 quantization steps = 96 dB of dynamic range.

 That is of course only the total statistical dynamic range. In practice it can vary depending on sample rate, type of dither used, noise shaping, etc.

  Quote:


 I'll tell you a little secret. I can encode 100db in one bit. 
 

Of course you can, if you use noise shaping and have high bandwidth.
 You still have the same amount of noise, you just move large amounts of it to higher frequencies.

  Quote:


 that made absolutely no sense to me what so ever. You are mixing up technologies. 
 

It's the same technology, just different implementations.


----------



## Justice Strike

Quote:


  Originally Posted by *D. Lundberg* /img/forum/go_quote.gif 
_It's not a choice, it's the same thing! The "resolution" is just a statistical measure of the amount of quantization errors you'll get in addition to the signal._

 

nevermind... no use discussing this with you.


----------



## D. Lundberg

Quote:


  Originally Posted by *Justice Strike* /img/forum/go_quote.gif 
_nevermind... no use discussing this with you._

 

Sorry you feel that way, I just don't see how anything I wrote could be that upsetting. It's was just basic stuff you could find in any entry-level book on digital audio.


----------



## googleborg

Quote:


  Originally Posted by *majkel* /img/forum/go_quote.gif 
_6) It's not dynamics killing people and affecting hearing but sound pressure with the given numbers of 140dB = pain, 160~180dB = death_

 

cool!


----------



## Justice Strike

Quote:


  Originally Posted by *D. Lundberg* /img/forum/go_quote.gif 
_Sorry you feel that way, I just don't see how anything I wrote could be that upsetting. It's was just basic stuff you could find in any entry-level book on digital audio._

 

i've tried to explain that you can have more resolution with 24 bit sound and that therefore you can do much better postprocessing (such as compression). This is just a fact. You seem to keep repeating the same stuff without really acknowledging this fact. There is little use in discussing if there is no way our stances are coming togheter even a little bit.

 BTW i'm not upset, i just don't have the time to get into a circular argumentation.


----------



## xnor

Quote:


  Originally Posted by *Justice Strike* /img/forum/go_quote.gif 
_i've tried to explain that you can have more resolution with 24 bit sound and that therefore you can do much better postprocessing (such as compression). This is just a fact. You seem to keep repeating the same stuff without really acknowledging this fact. There is little use in discussing if there is no way our stances are coming togheter even a little bit.

 BTW i'm not upset, i just don't have the time to get into a circular argumentation._

 

The very first post of this endless thread already told us so, why do you have to repeat this? I don't think anyone said that _that _is wrong.


----------



## D. Lundberg

Quote:


  Originally Posted by *Justice Strike* /img/forum/go_quote.gif 
_i've tried to explain that you can have more resolution with 24 bit sound and that therefore you can do much better postprocessing (such as compression). This is just a fact._

 

Yes, and nobody has argued otherwise.
 24, 32 or 64 bits are usually used when processing digital audio. 
 It's because every time you process the signal you add noise, and you use the extra bits as headroom to keep the noise floor low in the final version.


----------



## parrot5

Quote:


  Originally Posted by *D. Lundberg* /img/forum/go_quote.gif 
_Yes, and nobody has argued otherwise.
 24, 32 or 64 bits are usually used when processing digital audio. 
 It's because every time you process the signal you add noise, and you use the extra bits as headroom to keep the noise floor low in the final version._

 

So 24-bit playback makes sense for people who use EQ or other digital manipulation then


----------



## slytown

We can't hear the difference between 24 and 16 but I remember hearing that we can feel the difference. Is that right?


----------



## grawk

if something used all 24 bits, and you played it so you could hear the quietest parts, you could definitely hear the difference, until you went deaf 3 seconds later


----------



## xnor

Quote:


  Originally Posted by *grawk* /img/forum/go_quote.gif 
_if something used all 24 bits, and you played it so you could hear the quietest parts, you could definitely hear the difference, until you went deaf 3 seconds later_

 

what a Moment of Glory


----------



## leeperry

Quote:


  Originally Posted by *slytown* /img/forum/go_quote.gif 
_We can't hear the difference between 24 and 16 but I remember hearing that we can feel the difference. Is that right?_

 

It might give more room for dynamic contrast. Actually reproduction volume is irrelevant. If they play something at 96db, and then crescendo to 120...you're not going to hear any difference with 16 bits. Lets say your system is set at a max volume of 80db. Well, at 16 bit it would crescendo from 80 to 80. But at 24 bit it could go from like 53 to 66 or so.

 or not?


----------



## TStewart422

Quote:


  Originally Posted by *xnor* /img/forum/go_quote.gif 
_what a Moment of Glory_

 

A ZILLION cool points for you for that one! Love the Scorps!


----------



## grokit

Quote: 





bojamijams said:


> W00t! Thank you for that indepth explanation.. NOW I KNOW MORE!


 

 QFT, glad I was able to see through the SACD BS a long time ago, iTunes all the way for me 
	

	
	
		
		

		
		
	


	



   
  (bumpity bump for a great thread and a _fantastic_ OP!)


----------



## chesebert

Nyquist sampling theorem is just that a theorem.
   
  1. In practice, to avoid aliasing, most signals are ran through a LPF to cut off any frequency above Fmax (in this case 20khz)
   
  2. However, in order to sample the 20khz signal, the sampling frequency has to be increased above the 2*Fmax (or Fs the sampling frequency) to avoid getting zero results.
   
  3. now, come the fun part, D/A conversion.  D/A can be perfect if it follows pure Nyquist reconstruction formula.  I don't believe a filter is needed if the actual Nyquist reconstruction formula can be used.  So since everyone uses some type of filter, I presume no one has figuered out how to actually use the damn formula.
   
  Here is the formula for your amusement:  Xa(t) = Summation [Xa * (n/(2B)) ( (sine2piB (t-n/2B))/ (2piB (t-n/2B))], from negative infinity to infinity.
   
  On to quantization errors:
   
  Theoretically, quantization of analog sigansl always results in a loss of information.  This is result of the ambiguity introduced by quantization.  Quantization is an _irreversibel or noninvertible process_ (many to one mapping). In a fixed dynamic range, increasing the level of quantization will reduce the quantization steps, thereby reducing the quantization error. (always remember its a many to one mapping, more levels means less error).  This has nothing to do with dynamic range which is really the difference between the highest and lowest value in the signal.
   
  So in conclusion: Nyquist works on paper, the reconstruction formula works on paper, but nothing is perfect in the real world and higher sampling frequency and higher quantization level means less crap to worry about later on (easier and more accurate non-perfect reconstruction and minimizaing of quantization errors). Or you can just use your own ear to hear if there is a difference and leave all the 'objective' discussions to the engineers.
   
  If you are interested in this D/A monbojumbo, take Digital Signal Processing at your local engineering college.  I had fun when I took it a long time ago, and the labs are usually great (like designing your own D/A chip using FPGA).


----------



## Isak

Disclaimer: I have read the thread until about page 10 since it seemed to deteriorate at that point and did not seem to answer the question I’m going to ask now – which might even be a stupid question.
  I’m not going to discuss whether or not we cannot perceive anything outside the frequencies that can be recreated at a 44.1 khz sample rate, a 96 khz, 192 or whatever.
   
  My question is whether or not rates above 44.1 khz will provide greater detail within the spectrum we can hear even though we cannot hear the ‘new’ frequenzies that they can reproduce.
  Let me give an example that at least makes me understand my question better (not being condescending here, just trying to keep things within my own limited knowledge!) Suppose we are only able of hearing frequencies from 1-4 herz (I’m quite aware that we cannot hear those but it makes the example much easier). In that case you would only need to sample what would effectively produce a high note of 4 hz, which is a sample rate of 8 hz, right? If that is the case wouldn’t you only be able to reproduce 4 tones (1, 2, 3 and 4 hz) on such a recording? And if this is true would it not be the case that while you would not be able to hear, say, a 7 hz tone on a 16 hz recording, you’d still be able to hear the 2.5 hz tone, and hence benefit from the doubling?
   
  Or does our inability to hear things beyond 1-4 hz also mean that we’re unable to distinguish tonal differences at less than 1 hz?


----------



## chesebert

see above post.
   
  The short of it is, in theory, 44khz is really all you need.  But in practice since Nyquist reconstruction formula is never used to do the interpolation between the samples, you end up getting a stepped/approximate/half-assed interpolation of signal between each sample.  The most basic one is the zero-order-hold D/A, which results in the stepped response many magazines like to print.  Then stepped signal is "smoothed" out by a LPF (yeah..like that's a really accurate representation of the analog signal). I believe we have moved beyond that, but still making crap up by not using the formula.
   
  oh if you can believe this, at 96khz sampling frequency, D/A has less crap to make up in between samples.  And a trivia, the Nyquist guy died in the 70s and we still can't make his formula work in the real world. But who really cares? All you really need is higher sampling frequency to solve most of the problems described.
   
  So essentially the commercial audio DACs now days are just making crap up (cough..intelligent sample reconstruction) between the samples, as I don't believe they actually implement the Nyquist reconstruction formula.
   
  I would like to stress that a perfect reconstruction of the analog signal is possible, but only by following the Nyquist reconstruction formula. (It really works on paper).  It's quite amazing to see the math all work out perfectly no matter how complex the starting analog signal is.
  
  Quote: 





isak said:


> My question is whether or not rates above 44.1 khz will provide greater detail within the spectrum we can hear even though we cannot hear the ‘new’ frequenzies that they can reproduce.


----------



## kwkarth

A couple of reasons we do not want to limit sample rate to the Nyquist freq. for the limits of human hearing...
   

 The higher the sampling freq. the less deleterious effects of brick wall filters on the passband, including aliasing, etc.
   

 Upper harmonics of fundamentals do lend "air" and aliveness to a recording
   

 Because upper harmonics are generally greatly diminished in amplitude from the fundamental, low level resolution and detail is also critical for the preservation of an accurate recording so not only is sampling frequency important, but bit depth as well so that low levels can be faithfully maintained and not lost in the dither.


----------



## Pio2001

Quote: 





> Originally Posted by *chesebert* /img/forum/go_quote.gif
> 
> So since everyone uses some type of filter, I presume no one has figuered out how to actually use the damn formula.
> 
> Here is the formula for your amusement:  Xa(t) = Summation [Xa * (n/(2B)) ( (sine2piB (t-n/2B))/ (2piB (t-n/2B))], from negative infinity to infinity.


 

 You presume wrong. The filter is there to implement the very formula. 
   
  Actually, the perfect implementation is what is called a brickwall filter, or Sinc filter. The sinc function is defined by sinc(t) = sin(t) / t). But these are not used in audio because they lead to unwanted artifacts. They introduce ringing at the Nyquist frequency. That's the normal behaviour of perfect sampling. First you lowpass your original, which introduce a lot of ringing, then you sample, then you reconstruct your original lowpassed signal with all its ringing.
   
  Using a lowpass filter with an cosinus attenuation profile, as viewed in the frequency domain, is much cleaner. If you can generate lowpass filters by yourself, compare a given sample lowpassed at 10 kHz with a perfect sinc filter, then with a cosine filter starting at 9500 Hz and ending at 10500 Hz.
   
  Quote: 





> Originally Posted by *chesebert*
> 
> 
> 
> ...


 
   
  Right. It introduce noise at -96 dB if you quantize with 16 bits, at -144 dB if you quantize at 24 bits etc.

  
  Quote: 





isak said:


> My question is whether or not rates above 44.1 khz will provide greater detail within the spectrum we can hear even though we cannot hear the ‘new’ frequenzies that they can reproduce.


 
   
  Yes, by using the additional bandwidth to add noise shaped dither, you can increase the resolution in you working bandwidth. It is the same thing as increasing the number of bits.
  
  Quote: 





			
				Isak said:
			
		

> sample rate of 8 hz, right? If that is the case wouldn’t you only be able to reproduce 4 tones (1, 2, 3 and 4 hz) on such a recording?


 

 Not at all. You can reproduce any frequency below 4 Hz with a sampling rate of 8 Hz. 3.255664 Hz is perfectly possible, for example.
  
  Quote: 





> Originally Posted by *chesebert* /img/forum/go_quote.gif
> 
> The most basic one is the zero-order-hold D/A, which results in the stepped response many magazines like to print.  Then stepped signal is "smoothed" out by a LPF (yeah..like that's a really accurate representation of the analog signal). I believe we have moved beyond that, but still making crap up by not using the formula.


 

 No ! The signal is oversampled first, which allow to apply the Nyquist formula with good accuracy.
   
  Only very high end DACs directly lowpass the zero-order hold. That indeed leads to crappy results (-2 dB of treble in the audible range). That's a coloration that they either believe is "more musical", either deliberately introduce to distinguish them from their competitors.
   
  Quote: 





> Originally Posted by *chesebert*
> 
> 
> 
> oh if you can believe this, at 96khz sampling frequency, D/A has less crap to make up in between samples.


 
   
  There is more crap between the samples ! All the ultrasonic noise, parasites from computer displays etc is recorded and fed into the amplifier during playback.
  
  Quote: 





kwkarth said:


> The higher the sampling freq. the less deleterious effects of brick wall filters on the passband, including aliasing, etc.


 
    
  There is no deleterious effects of brick wall filters in the audible range at 44.1 kHz.
   
  Quote:


kwkarth said:


> Upper harmonics of fundamentals do lend "air" and aliveness to a recording


 
    
  No, they don't. They're inaudible.
   
  Quote:


kwkarth said:


> Because upper harmonics are generally greatly diminished in amplitude from the fundamental, low level resolution and detail is also critical for the preservation of an accurate recording so not only is sampling frequency important, but bit depth as well so that low levels can be faithfully maintained and not lost in the dither.


 
   
  16 bits are enough in all practical situations for this purpose.


----------



## grokit

Quote: 





pio2001 said:


> 16 bits are enough in all practical situations for this purpose.


 

 While I don't completely understand all of the details of this ongoing debate as reflected in this thread, I do get the jist of it, and what you are saying makes sense to me. It also makes me trust my own conclusions, that the most important thing is the quality of the source recording and mastering, regardless of the bit rate and sampling frequency used for playback. Anyways I do appreciate all of the thoughtful posts lately, it's very interesting subject but sometimes it just makes me want to play a record!


----------



## chesebert

I didn't know the brickwall (that is, the brick wall is a frequency-domain brickwall) was even implmeneted.  It has an impulse reponse that extends for all time. 
   
  input signal -> perfect D/A -> impulse response -> brick wall -> perfect analog signal.
   
  let's see which one of these steps can't be implemented. . . all of it.
   
  One disclaimer: I haven't touched this DSP stuff in many many years, so my knowledge may be dated and certain things I can't remember that well.  in order words, I may not know what I am talking about 
  Quote: 





pio2001 said:


> You presume wrong. The filter is there to implement the very formula.
> 
> Actually, the perfect implementation is what is called a brickwall filter, or Sinc filter. The sinc function is defined by sinc(t) = sin(t) / t). But these are not used in audio because they lead to unwanted artifacts. They introduce ringing at the Nyquist frequency. That's the normal behaviour of perfect sampling. First you lowpass your original, which introduce a lot of ringing, then you sample, then you reconstruct your original lowpassed signal with all its ringing.


----------



## JaZZ

pio2001 said:


> You presume wrong. The filter is there to implement the very formula. Actually, the perfect implementation is what is called a brickwall filter, or Sinc filter. The sinc function is defined by sinc(t) = sin(t) / t). But these are not used in audio because they lead to unwanted artifacts. They introduce ringing at the Nyquist frequency. That's the normal behaviour of perfect sampling. First you lowpass your original, which introduce a lot of ringing, then you sample, then you reconstruct your original lowpassed signal with all its ringing.


 
   
  Why exactly are you concerned about a ringing at 22,050 Hz? In my understanding it would theoretically be better to accept it, as 22 kHz are known to be inaudible. Theoretically a steep filter with massive ringing barely affects transient response at audible frequencies, in contrast to your favored smoother filter with reduced ringing which moreover may affect amplitude response.
   
  Well, at least according to the established audio doctrine. In reality the ringing is most likely audible nonetheless, as the different filter settings on my Corda Symphony with different characteristic and intensity of the ringing show (note that some of them have identical amplitude response!). Which makes me think that – at least in view of finite perfection of the used electronics components – an abrupt and steep low-pass filter so close to the audio band is critical for sensitive and experienced ears. Actually I have passably made friends with the CD format since the HD 800 era, and due to an imcompatibility of my hearing with speaker-based recordings listened through headphones I'm forced to use crossfeed. The CD format is perfect for my own implementation, so it has become my audio standard since quite a while. But lately I have compared some double-discs again with the same recording once on DVD-A, once on CD (DGG), and the result is clear, although not night and day: Despite the plausibility of the Nyquist theorem and the reasonings of its proponents, in practice recordings with higher frequency bandwidth sound better to my ears.

  


> No, they don't. They're inaudible.


 
 No, not necessarily. Maybe (!) if they're in the ultrasonic range, but his statement didn't implicate that. I'm fairly convinced that preserved overtone transient accuracy/sharpness is the key to a non-digital sound (to avoid the term «analogue»). Whereas low-rez digital tends to smear them.
.


----------



## kwkarth

Quote: 





> Originally Posted by *Pio2001* /img/forum/go_quote.gif
> 
> You presume wrong. The filter is there to implement the very formula.
> 
> ...


 
  Well, I guess we've been told.


----------



## grokit

There's an "established audio doctrine"?
   
  That seems debatable; as far as I can tell there are many oppositional viewpoints that are extremely well-argued, without any apparent resolution.


----------



## kwkarth

I depends upon with whom you speak.


----------



## Pio2001

Quote: 





jazz said:


> Why exactly are you concerned about a ringing at 22,050 Hz? In my understanding it would theoretically be better to accept it, as 22 kHz are known to be inaudible. Theoretically a steep filter with massive ringing barely affects transient response at audible frequencies, in contrast to your favored smoother filter with reduced ringing which moreover may affect amplitude response.
> .


 

 To put numbers on it, filters used in DACs are very far from the regular 6 db per octave found in speakers. They don't start until 20 kHz, and reach minus infinite soon after. Compared to speaker crossovers, they can be viewed as brickwalls.
  But compared to Sinc filters, they are very smooth. A Sinc filter can let everything pass unaffected until 22049 Hz, and reach minus infinite at 22050 Hz.
   
  I prefer smoother filters (0 db at 21 kHz, minus infinite at 22 kHz, for example) because I had the possibility to compare them in the 12 -16 kHz range. At 13 khz, the ringing introduced by a brickwal filter was very annoying, while a filter with a transition band from 12500 to 13500 Hz produced a very good result. The treble loss was subtle, and there was no ringing at all.
   
  At 22.05 kHz, it shouldn't matter. I'm just keeping the same what sounds better at 13 khz.


----------



## Somnambulist

I read this thread from start to end in the early hours of last night. Gregorio should be praised for his consistently informative posting and his patience dealing with certain posters. Some of it went over my head from a technical standpoint, but I 'get the gist' of it and it'll surely help me with DAC choices in future.


----------



## nick_charles

Cough... Meyer and Moran 2007 ?... empirical evidence that stripping the upper (22k+) harmonics may be in fact be inaudible in playback, not as yet refuted by any latter empricial tests...


----------



## andysor

Thank you so much to Gregorio for his extremely informative and easy to follow OP and considerable patience, even when confronted by personal attacks from people intent on defying mathematical logic in favour of clinging to their placebos.
   
  After reading most of this thread I have saved myself from a future of unnecessary outlays and will be focusing on enjoying the music.


----------



## kingtz

Thank you! This was very educational and opened my eyes!


----------



## Deep Funk

In my search for a DAC this thread will prove valuable.


----------



## mike1127

Others have said it here---Nyquist sampling theory is just theory, and reality is notorious for not conforming to theory.
   
  The 44.1 KHz sampling rate was chosen on the idea that humans can't hear much about 20 KHz (or even 15 KHz). If you play a human a tone at 22.05 KHz, they can't hear it. But there's an assumption here---that the ear is a linear system. It's not.
   
  Say we have two impluse-like signals A and B. Maybe two recordings of rim shots. Further suppose that when Fourier-analyzed they are identical at frequencies below 20 KHz. But they differ above 20 KHz.
   
  Can the human ear tell these apart? If the ear were linear, we can settle that question immediately: nope. But the ear is nonlinear.
   
  I'm pointing out that these are distinct statements:
   

 No one can hear at 22 KHz tone.
 No one can tell the difference between impulse-like signals A and B which differ only in harmonic components above 22 KHz.
   
  Because the ear is nonlinear, one does not imply the other.
   
  Note that any A/D D/A system that incorporates any kind of filtering or processing that is unrelated to how natural-world system behave, will exhibit a transient response that is unlike anything in the natural world. Infinite impulse-responses (especially that have energy before the leading edge) are examples. I wonder if this explains a lot of the problems with digital.


----------



## nick_charles

Quote: 





mike1127 said:


> Can the human ear tell these apart?


 

 The best evidence we have is that the answer is no. This has been done frequently since the late 70s with filters removing harmonic content being undifferentiable from the unfltered content or inserting AD stages to remove 22K+ material;. You can try it youself, take a 96khz file and then low pass filter off the content above 22K and do a FooBar ABX. There are numerous anecdotes about the effect of higher harmonics but the empirical evidence to date just does not back this up.
   
   


>


----------



## Pio2001

A more direct experiment : play a 12 khz tone in your left speaker, and a 18 khz one in your right speaker.
  Then do it in mono.
  Be careful not to fry your tweeters if you try it at home.
   
  This kind of experiment, much more likely to reveal any relevant non-linearity than any musical content, has been done independantly by David Griesinger, by Nika Aldrich and by myself. Our results match : extended frequency response, though usually inaudible, can become audible using this kind of exceptional test tones, and the result is that the frequency extension *increases distortion*, because in this special case, thanks to the deafness of the ear at upper frequencies,* the hifi system introduces more non-linear distortion than the ear ! *
   
  David Griesinger's experiments seems to show that the intermodulation comes from the amplifier rather than the tweeter.


----------



## hscai

Thank you for the interesting and quality write-up. Unfortunately, I've been lazy recently, and have rarely read an article through completely and thoroughly - this one is an exception! That goes for the interesting comments in this thread as well.


----------



## dedero

Hello guys,
   
  After reading this article I will ask you a newbie question.
   
  After seeing that Apple (Beatles) release a USB stick with whole discography in 24/96, can this new format give a bigger gap to allow The Beatles still "fighting" in loudness war? (I'm asking for the beatles just to put an example, it applies to the other bands)
   
  Thanks
   
  Bruno.


----------



## leeperry

dedero said:


> After seeing that Apple (Beatles) release a USB stick with whole discography in 24/96, can this new format give a bigger gap to allow The Beatles still "fighting" in loudness war?


 
   
  CDDA can provide 96dB of dynamics...the weak link is not the medium, it's the record companies that ask the mastering engineers to make it loud...because ppl want it to be loud, as they'll listen to it on their ipod.
   
  I've heard the Beatles 24/96 USB key, SQ is clearer than on the CDDA versions...but it's still very much bloated to my ears. Compare those versions to the CDDA mastertape bootlegs that are floating around, and they sound processed/EQ's/compressed to death. But more than likely if they had released "audiophile" versions of those records, ppl would have whined that the sound was dull and lifeless.


----------



## Deep Funk

That would mean the 16 bit digital CD remasters are still better right? I prefer CD anyway...
   
  DACs, confusing too. Maybe a studio oriented DAC will cross my path in time...


----------



## kwkarth

Quote: 





deep funk said:


> That would mean the 16 bit digital CD remasters are still better right? I prefer CD anyway...
> 
> DACs, confusing too. Maybe a studio oriented DAC will cross my path in time...


 

 Again, it's not the medium that is the problem.  If the mix-downs that are committed to release are compressed/processed to death for AM radio's sake, nothing you can do will make it better.  Even dynamic range expanding will not unscrew it.


----------



## lucozade

Quote: 





grokit said:


> While I don't completely understand all of the details of this ongoing debate as reflected in this thread, I do get the jist of it, and what you are saying makes sense to me. It also makes me trust my own conclusions, that the most important thing is the quality of the source recording and mastering, regardless of the bit rate and sampling frequency used for playback. Anyways I do appreciate all of the thoughtful posts lately, it's very interesting subject but sometimes it just makes me want to play a record!


 
   x2


----------



## Deep Funk

X3, it is time for some music. 
   
  Kwkarth, yes I know that. Now all I can hope for is that bad pressings and bad mix-downs do not cross my path too often. Thanks anyway for clearing that up. That does imply that for hunts on the next CDs I will have to do more digging. Judas Priest already gave me head aches...


----------



## Danz03

Interesting thread, I've always thought that 24bit/96kHz is only ever used in a studio, I think it's the same reason why digital artists would normally use 30-32 bit colors when we can really only see 24 bit true colors.  But the thing about 24bit audio is that one can manupulate it and then downsample it to 16 bit with no loss of quality. Let say, someone were to use a digital pitch shifter on a 16bit/44.1kHz audio file and lower the pitch by 10%, then the highest frequncy that's audiable would not be 20kHz anymore but 18kHz, which can be heard by most people and the resulting 16bit curves will not be as smooth either. I'm not saying a lot of people are going to pitch-shift a CD but a lot of people would use EQs in their digital amps etc. That will degrate the sound of 16bit/44kHz audio. But if the source was 24bit/96kHz, altering it with a digital EQ would not degrate the sound at all as the altering process was done in the frequency and resolution range beyond what we could hear, and then down sampled back to our audio range. So I guess if one were just listening to music without messing around with it, 16bit/44.1khz is fine, but if someone were to manupulate the music in anyway, 24bit/96kHz is a must.


----------



## grokit

Danz03, I find your posts interesting but somewhat hard to read. The text is smaller, the kerning between characters and spaces is less, and the line spacing is compressed. Plus you tend to write longish paragraphs, which would have better readability with the default formatting. I wonder if you are changing the text style from the default on purpose, or if you are pasting it in from another program. If it is the former, please stop it as it is just too small and scrunched together compared to the surrounding text, requiring me to either zoom in to read it or ignore your post. If it is the latter, please change to plain text before copying it to your clipboard and into the edit box, thanks.


----------



## Danz03

Thanks, better? For some reason, that was the default format on my computer, sorry. 
	

	
	
		
		

		
			




I managed to change the size but don't know how to change the spacing and kerning here. I tried cut and paste before but somehow it didn't seem to work. It looked fine when I type but after I submitted, it all got changed somehow!


----------



## grokit

Hmm... now your text is bigger than normal, I wonder why it's different for you than everyone else! Do you type your text right into the edit box?


----------



## Ham Sandwich

Quote: 





danz03 said:


> Interesting thread, I've always thought that 24bit/96kHz is only ever used in a studio, I think it's the same reason why digital artists would normally use 30-32 bit colors when we can really only see 24 bit true colors.  But the thing about 24bit audio is that one can manupulate it and then downsample it to 16 bit with no loss of quality. Let say, someone were to use a digital pitch shifter on a 16bit/44.1kHz audio file and lower the pitch by 10%, then the highest frequncy that's audiable would not be 20kHz anymore but 18kHz, which can be heard by most people and the resulting 16bit curves will not be as smooth either. I'm not saying a lot of people are going to pitch-shift a CD but a lot of people would use EQs in their digital amps etc. That will degrate the sound of 16bit/44kHz audio. But if the source was 24bit/96kHz, altering it with a digital EQ would not degrate the sound at all as the altering process was done in the frequency and resolution range beyond what we could hear, and then down sampled back to our audio range. So I guess if one were just listening to music without messing around with it, 16bit/44.1khz is fine, but if someone were to manupulate the music in anyway, 24bit/96kHz is a must.


 

 16 bit vs. 24 bit and 44.1 kHz vs 96 kHz are separate issues.  You can have 16/96 or 24/44.1 if you want.
   
  The pitch shift example you give is a concern with sample rate.  You want a higher sample rate if you're going to do things like pitch shifting.  The bit depth doesn't matter to a pitch shift operation.
   
  What about if you upsample the 44.1 to 88.2 before doing the pitch shift?  I've been experimenting with doing a software based resample up to 88.2 before doing my digital EQ.  I'm not sure if it is making an audible difference.  I need to experiment some more and try some different EQs as well.


----------



## corsario

Hi everybody,
   
  I am replying here to a question arising from the following post in another topic :
http://www.head-fi.org/forum/thread/504567/hm-602-portable-music-preorder/120#post_6869534

  
  Quote: 





			
				gatz said:
			
		

> Sorry being of topic, but I find quite noticable differences between some commercial 16/44 and 24/96 tracks.  Anyway your statement may be true and *the difference may come from 24/96 mastering*. (You can try Rebecca Pidgeon "Spanish Harlem" on Chesky for instance)


 
   
  It sure must come from the mastering. Did you ABX this "noticable" difference by the way ?
   
  The right way to go is to *perform yourself the downsampling of the 24/96 track* using R8brain (use the free version which appears to be better than the PRO one for this purpose. It is complicated to explain but let's say that the default dithering performed by the free version of R8brain is the good one, just set the quality setting to "Very high"). This way you will have the 16/44 version of the 24/96 track : no mastering difference, etc...
   
  And there, go to foobar, use the ABX module and make the test*
   
  If you can ABX it, then it will be time to consider if R8brain did not deteriorate the 24/96 track or did not performed an inappropriate dithering. But, so far, no ABX was sucessful, so the question is not on the table yet 
	

	
	
		
		

		
		
	


	



   
   
  *If your soundcard can not switch automatically between 16/44 and 24/96 (this is the case for EMU1212m for instance) then you will need to use R8brain again and upsample your 16/44 sample (no problem it is juste zeros added to obtain 24/96 res, nothing is changed to the file - you can not recreate what was lost in the downsampling process). If your sound card can deal indifferently with any input format (e.g. Lynx2B) then do not bother to upsample and just make the 24/96 vs. 16/44 comparison. If you can ever ABX on one sample, please be kind to indicate the 24/96 sample you used, and I am sure a lot of people, including myself, will be happy to try to ABX it as well.


----------



## gatz

Interesting topic !

 I was always been convinced that 24/96 is superior to 16/44.
*Well, until today.*
  I will try to make the story short.

 It has started maybe 10 years ago when I was trying a SACD reissue of an old analog jazz recording. The SACD was in every aspect superior to the CD issue. Since then, I always thought that 24/94, DSD or any other “HD” audio formats are obviously superior. My last listening test was with my Hifiman HM801. It wasn’t to reconsider my position regarding HD/SQ but more to see if this 24/96 capability was really an asset for such a portable device. Then I purchased 24/96 reissue of some all time audiophile favorites I already own from Chesky label. Again, the 24/96 track was better than the (original) CD rip.
   

I made further testing today (thank’s to corsario remarks).


*Test 1* (my original test) :

 Track 1 : Spanish Harlem (Rebecca Pidgeon) 24/96 from “retrospective”
 Track 2 : Spanish Harlem (Rebecca Pidgeon) 16/44 CD rip from “The Raven”
 Result : Quite noticeable (well, I should say, obvious !) difference. No need to ABX.
  Anyway I have installed foobar ABX and I tried the blind test : I always find the right track, in few seconds.


*Test 2* :

 Track 1 : Spanish Harlem (Rebecca Pidgeon) 24/96 from “retrospective”
 Track 2 : same track converted to 16/44 with R8brain as corsario suggested
 Track 3 : same track downsampled and dithered using soundforge
 Result : No listening difference between the 3 using the HM801 + EM3pro.
 Then I move to my DAW (RME fireface + dynaudio BM5a) : *No obvious difference too *!


 But why my original 16/44 track is so different ?
 OK, maybe my CD rip wasn’t that accurate?
 I purchased (for the 3rd time...) “Spanish Harlem” from HD track, the 16/44 version, to compare with my CD rip.
  They are identical.
   

*The mastering of the original 16/44 track and the 24/96 one are obviously different.*


Last test, back to soundforge, testing the tracks dynamics :
 16/44 original track : peak level = -0,5dB, RMS level = -21,4dB
 24/96 original track : peak level = -0,1dB, RMS level = -19,5dB

 The 24/96 is slightly louder than the 16/44.
  Having a different mastering for the 24/96, I would have expected the opposite!
  This alone explains the more noticeable analog noise floor and soundstage I can ear on the 24/96 track.
   

 The 24/96 track was part of a "best of", so this could explain the different mastering beyond the format. My next test would be to purchase (for the 4th time !) the 24/96 track from the original album : but it doesn’t exist. The current album version is advertised as “Bob Katz 15th Anniversary Remaster” (strangely in 24/88), so probably different from all versions I already have.

  
   
  So, 24/96 (for the end listener) is just another marketing buzz ?
 Why not reissuing directly 16/44 audiophile remaster ?


----------



## xnor

Quote: 





gatz said:


> Why not reissuing directly 16/44 audiophile remaster ?


 

 Making (big) money doesn't work that way.


----------



## grokit

Quote: 





gatz said:


> I was always been convinced that 24/96 is superior to 16/44.
> *Well, until today.*
> I will try to make the story short.
> 
> ...


 

 When all else fails just play a record


----------



## leeperry

gatz said:


> So, 24/96 (for the end listener) is just another marketing buzz ?
> Why not reissuing directly 16/44 audiophile remaster ?


 
   
  Because it's far more complicated...most of the 16/44 crowd will listen to crappy mp3 on their ipod or on the radio or whatnot, and the higher the dynamics compression the better it will compress/decode. OTOH, the 24/96 crowd wants as little compression as possible as they have proper gear to boot. If I reuse the example I gave above...if they had released the untouched mastertapes dumps for the Beatles reeeditions, most ppl would have said that the sound was dull and lifeless. Audiophiles are prolly less than 0.1% of the population and the volume caps on the ipod will only make things worse.
   
  Also, this doesn't help and will only make music clip even further: http://www.gearslutz.com/board/tips-techniques/334385-intersample-peaks.html
   
  Ppl want to listen to clipped autotuned music these days...most of those singers are unable to sing w/o pitch correcting machines anyway


----------



## Danz03

[size=medium]I personally don't see much advantage with using 24bit with 44.8kHz (or 44.1kHz) sampling rate. It would be fine if the wave curve just happen to fall onto the grid conveniently, but more often then not, it doesn't, and what if the curve falls onto a point right in the middle of two bit points? Dither will kick in and randomly shift the point up or down to fit it to a bit point, but then when the 24 bit map is being down-sampled back to 16 bit, the same thing can happen, a point that has been previously shifted up by dither can be shifted up further again in the same way and creating more errors, and more errors in this case = more noise and distortions. However, if one were to use 24bt/96kHz, the mistakes would be halved in time, comparing to a 24bt/48kHz, as the time resolution is doubled.[/size]     [size=medium]Yes, if one were to just pitch shift a wave up or down by exactly one octave (double or half), then up-sampling it before-hand from 48kHz to 96kHz may be fine, as the bit positions will stay where they are, but what if one were to pitch shift the wave from an A4 (440Hz) to a C4 (261.63Hz)? Of course the bit positions would have to change, in that case, 24 bit would always be better than 16.[/size]     [size=medium]I don't think up-sampling would improve the sound quality of a digital EQ, since up-sampling would only affect the frequency and not the amplitude side of the EQ, ie: the Q factors or the selected frequency range would be a few milli-hertz more accurate. Probably a linear predictive coding would improve the sound quality more so than over-sampling.[/size]   
  Quote: 





ham sandwich said:


> 16 bit vs. 24 bit and 44.1 kHz vs 96 kHz are separate issues.  You can have 16/96 or 24/44.1 if you want.
> 
> The pitch shift example you give is a concern with sample rate.  You want a higher sample rate if you're going to do things like pitch shifting.  The bit depth doesn't matter to a pitch shift operation.
> 
> What about if you upsample the 44.1 to 88.2 before doing the pitch shift?  I've been experimenting with doing a software based resample up to 88.2 before doing my digital EQ.  I'm not sure if it is making an audible difference.  I need to experiment some more and try some different EQs as well.


 
   
There could be a difference between the two, but one would need to have an extremely good monitoring system and hearing above 20kHz to notice the difference. Or, if one were to use a digital EQ, in that case, the 24bit/96 kHz should sound better in theory.
 [size=small][size=medium]Like I said before, the main advantage of 24bt/96kHz is that one could manipulate the original sound signal and then convert it down to 16bit/44.1kHz with little or no degradation, so it is very useful for doing mastering.[/size][/size]   
  Quote: 





> Originally Posted by *gatz* /img/forum/go_quote.gif
> I was always been convinced that 24/96 is superior to 16/44.
> *Well, until today.*
> 
> ...


----------



## leeperry

> [size=medium]I don't think up-sampling would improve the sound quality of a digital EQ, since up-sampling would only affect the frequency and not the amplitude side of the EQ, ie: the Q factors or the selected frequency range would be a few milli-hertz more accurate. Probably a linear predictive coding would improve the sound quality more so than over-sampling.[/size]


 

 Well, some professional EQ plugins think otherwise: http://www.gearslutz.com/board/5101666-post296.html
   
_"The oversampling is necessary to avoid aliasing with the saturation, but also helps to get a more accurate response in the upper end. Just as you guys said, we use a steep IIR LP filter as oversampling filter."_
   
  Of course this EQ sounds out of this world, coz I've heard some upsampling EQ's that still sounded horrid ^^


----------



## Danz03

But you are talking about a piece of professional studio gear. Like I said before, most studios would use 24bit/96kHz at least for recording, so most professional EQs would be operating in 24 bit at least anyway, which is already better than what most people can hear, and the over-sampling is to make sure that it's accurate even in 24 bit. I was addressing @Ham Sandwich's comment on oversampling from 44.1 to 88.2kHz before going through a digital EQ, to me, that wouldn't make a lot of difference. I couldn't find any specs on how the A-Range EQ works but then I think most DAWs would up-sample the signal before going through any plug-ins anyway. It's quite different talking about studio gears and audiophile products, studio gears have to be accurate while audiophile products just have to sound good.
  
  Quote: 





leeperry said:


> Well, some professional EQ plugins think otherwise: http://www.gearslutz.com/board/5101666-post296.html
> 
> _"The oversampling is necessary to avoid aliasing with the saturation, but also helps to get a more accurate response in the upper end. Just as you guys said, we use a steep IIR LP filter as oversampling filter."_
> 
> Of course this EQ sounds out of this world, coz I've heard some upsampling EQ's that still sounded horrid ^^


----------



## Ham Sandwich

Quote: 





danz03 said:


> [size=medium]I personally don't see much advantage with using 24bit with 44.8kHz (or 44.1kHz) sampling rate. It would be fine if the wave curve just happen to fall onto the grid conveniently, but more often then not, it doesn't, and what if the curve falls onto a point right in the middle of two bit points? Dither will kick in and randomly shift the point up or down to fit it to a bit point, but then when the 24 bit map is being down-sampled back to 16 bit, the same thing can happen, a point that has been previously shifted up by dither can be shifted up further again in the same way and creating more errors, and more errors in this case = more noise and distortions. However, if one were to use 24bt/96kHz, the mistakes would be halved in time, comparing to a 24bt/48kHz, as the time resolution is doubled.[/size]     [size=medium]Yes, if one were to just pitch shift a wave up or down by exactly one octave (double or half), then up-sampling it before-hand from 48kHz to 96kHz may be fine, as the bit positions will stay where they are, but what if one were to pitch shift the wave from an A4 (440Hz) to a C4 (261.63Hz)? Of course the bit positions would have to change, in that case, 24 bit would always be better than 16.[/size]


 

 You're making arguments for 24 bit that are counter to gregorio's posts that began this thread.  The errors or difference caused by doing things at 16 bit shouldn't be audible or relevant to consumer processing and listening (listening to the music as an end product with a some digital post-processing being done, and not as tracks that will be further mixed or processed later).  In a studio you might be going through dozens of digital processes and if each of them bumped a 16 bit sample up to 24 bit to do its processing and then back down to 16 bit before sending the sample on to the next step then I could see some cumulative errors cropping in.  But for consumer style processing, even in cases of things like pitch shifting, I would need some convincing to see how 16-bit vs. 24-bit is an issue as long as the processing doesn't cause clipping at 16-bits.
   
  The reason I'm playing around with upsampling before my digital EQ process is to see if maybe the higher sample rate moves the digital filtering that the EQ is doing further out of the audible range and maybe, just maybe, that will have an effect.  I have some EQs that I think may be dulling the upper high frequencies so that's why I'm curious.  I don't know what those EQs are doing internally, whether they're upsampling or not.  I'm going on the assumption that they are not.  I'm not changing the bit depth during the resample.  The resampler I'm using is the native resampler in J River Media Center.  It is claimed to be an "audiophile quality" resampler.


----------



## leeperry

danz03 said:


> But you are talking about a piece of professional studio gear.


 

 I'm talking about a VST plugin that I'm using in my everyday media player.


----------



## Danz03

You are using the Softube Trident A-Range EQ plug-in for you media player?





leeperry said:


> I'm talking about a VST plugin that I'm using in my everyday media player.


----------



## Ham Sandwich

I'm using VST plug-in EQs as well.  Both Foobar and J River Media Center can use VST plug-ins.  So can some other media players.
   
   
  There is no reason to stick with the stock 5 to 10 band EQs that come with most media players.  Those EQs don't have enough control to be able to do corrective EQ fixes and they can sound poor as well.


----------



## PelPix

It's true that you can't hear the difference, but it makes a big difference when you're actually doing mixing and you may, in what we call a "worst case scenario," have to increase the gain until the noise is audible.  In these cases having a bit depth higher than 16 bit becomes practical.  Not for listening, but for processing.
  I master in "Overkill mode" (192khz/64-bit float).  It has no purpose whatsoever beyond compatibility.


----------



## kwkarth

Quote: 





pelpix said:


> It's true that you can't hear the difference, but it makes a big difference when you're actually doing mixing and you may, in what we call a "worst case scenario," have to increase the gain until the noise is audible.  In these cases having a bit depth higher than 16 bit becomes practical.  Not for listening, but for processing.
> I master in "Overkill mode" (192khz/64-bit float).  It has no purpose whatsoever beyond compatibility.


 
  And your final product after post, is no doubt, of higher quality as a result.


----------



## PelPix

That too, because it allows more smooth sound after high quality dithering than something natively 16-bit.  It's the same principle as video game anti-aliasing.


----------



## aspenx

Quote: 





danz03 said:


> You are using the Softube Trident A-Range EQ plug-in for you media player?


 

 I want to know exactly what EQ too. Currently I'm using Electri-'s parametric EQ with foobar2k through a VST wrapper from George Yohng.


----------



## xabu

Quote: 





gregorio said:


> 2 = .... It is in fact the fundamental tenet of the Nyquist-Shannon Sampling Theorem on which the very existence and invention of digital audio is based. From WIKI: “In essence the theorem shows that an analog signal that has been sampled can be *perfectly* reconstructed from the samples”....


 

  (I did not read the whole thread ... so if the following was already mentioned ... just ignore )
   
  Well, I'm somewhat with you regarding the whole bit mathematics but regarding Nyquist your cite is a little bit symplifying:
   
  "_In practice, neither of the two statements of the sampling theorem described above can be completely satisfied, *and neither can the reconstruction formula be precisely implemented*. The reconstruction process that involves scaled and delayed sinc functions can be described as ideal. *It cannot be realized in practice since it implies that each sample contributes to the reconstructed signal at almost all time points, requiring summing an infinite number of terms*. Instead, some type of approximation of the sinc functions, finite in length, has to be used. The error that corresponds to the sinc-function approximation is referred to as interpolation error. Practical digital-to-analog converters produce neither scaled and delayed sinc functions nor ideal impulses (that if ideally low-pass filtered would yield the original signal), but a sequence of scaled and delayed rectangular pulses. This practical piecewise-constant output can be modeled as a zero-order hold filter driven by the sequence of scaled and delayed dirac impulses referred to in the mathematical basis section below. A shaping filter is sometimes used after the DAC with zero-order hold to make a better overall approximation._"
   
http://en.wikipedia.org/wiki/Nyquist-Shannon_sampling_theorem
   
  Well, and there is also something more about the bit mathematics ...
   
  Quote: 





gregorio said:


> ... each bit of data provides 6dB of dynamic range ...


 
   
  So lets say
   
  with 1 bit we would resolve 6 dB into 2 values (e.g. 0 dB and 6 dB, nothing in between)
  with 2 bit we would resolve 12 dB into 4 values (e.g. 0,4,8,12 dB, nothing in between)
  with 3 bit we would resolve 18 dB into 8 values (more values, finer resolution)
  with 4 bit we would resolve 24 dB into 16 values (more values, finer resolution)
  .
  .
  .
  with 16 bit we would resolve 96 dB into 65536 values (that means 1 dB would resolve into 684 values, much finer resolution)
  .
  .
  .
  with 24 bit we would resolve 144 dB into 16777216 values (that means 1 dB would resolve into 116508 values, much much finer resolution, perhaps almost overkill 
	

	
	
		
		

		
			





)
   
   
  I would call that a step rise in volume resolution ...


----------



## khaos974

Or rather think of it this way, what is the tiniest difference of air pressure that you can hear, any difference of air pressure below that cannot be head? That will be the 2nd step on a scale from 0 to 65635.
  On 16 bit, it means that we hear a continuous variation of volume level from that single step to 96 dB higher.
  On 24 bit, it means that we hear a continuous variation of volume level from that single step to 144 dB higher.
   
  => The steps are not more detailed, you don't get more resolution, you get more dynamic range.


----------



## xabu

Quote: 





khaos974 said:


> Or rather think of it this way, what is the tiniest difference of air pressure that you can hear, any difference of air pressure below that cannot be head? That will be the 2nd step on a scale from 0 to 65635.
> On 16 bit, it means that we hear a continuous variation of volume level from that single step to 96 dB higher.
> On 24 bit, it means that we hear a continuous variation of volume level from that single step to 144 dB higher.
> 
> => The steps are not more detailed, you don't get more resolution, you get more dynamic range.


 

 Correct, there is a limitation of what difference you can hear.
   
_*If* the tiniest difference that could be heard would be 2 * (6 dB / 65635) = 0,0002 dB ..._
   
_you would be right, with this picture, there were some amount of non audible steps in bigger resolution ... but bigger resolution it were nonetheless ... (... and more dynamic range, no question about that ...)_
   
_But:_
   
_Edit: Don't get the meaning of what I wrote above anymore ... I think somehow I tried to make sense of what you wrote _
   
_------------------------- snip_
   
  Due to the fact that the dB scala (sound pressure level) is not linear but logarithmical based on the sound pressure measured in Pa we had to calculate the whole stuff better in Pa ... because it's not dB values what is stored ...
   
  So in fact 1 dB is the smallest amount of difference in sound pressure level the human ear can recognize (all related to 1khz),
   
  that would be 0,00002(244) Pa at 1 dB above 0 dB and e.g.  0,02244 Pa at 1 dB above 60 dB, etc.
   
  But to hold/represent these values which range from 0,00002 Pa to 20 Pa which translates to (0 - 120 dB) in a binary format, you would definetely need more than 16 bit.
   
  Even for 0 - 96 dB (0,00002 Pa - 1,26191 Pa) you would need more than 16 bit.


----------



## xnor

No you don't, and from the Handbook of Recording Engineering the dynamic range of music as normally perceived in a concert hall doesn't even exceed 80 dB...
  With popular music you're in the range of 10 dB..


----------



## xabu

Quote: 





xnor said:


> No you don't, and from the Handbook of Recording Engineering the dynamic range of music as normally perceived in a concert hall doesn't even exceed 80 dB...
> With popular music you're in the range of 10 dB..


 

  
  I don't know exactly what of my statements you refer to with "_No you don't_", if it is the last two you're wrong.
  That's just mathematics.
  You need more than 16 bits to hold all values between 0,00002 and 1,26191.
  There's nothing to discuss.
   
  Regarding the rest I've to admit I must pass until I learned more about the whole mastering stuff ... very interesting in any case!
   
  But nevertheless with 16 bit you have a pot of max. 65536 values available per sample.
  If that's enough for current mastering standards and digital domain specifics than it's enough.
  Though the audible range human ears can perceive exceeds this pot of values clearly ...


----------



## nick_charles

Quote: 





xabu said:


> Quote:
> 
> 
> 
> ...


 

 Mathemetically you are correct, as a matter of min and max levels perceived and we are talking about zero background noise then sure 16 bits is inusfficient to capture that range noiselessly. Pragmatically though in any environment there is so much background noise that several low order bits of resolution get lost in noise. Unless you can listen in an anechoic chamber the low level signals are drowned out and the difference between 16 bits and 24 bits is academic rather than essential.


----------



## D. Lundberg

Quote: 





xabu said:


> So lets say
> 
> with 1 bit we would resolve 6 dB into 2 values (e.g. 0 dB and 6 dB, nothing in between)
> with 2 bit we would resolve 12 dB into 4 values (e.g. 0,4,8,12 dB, nothing in between)
> ...


 
   
  You'll actually get *everything* inbetween, it's the noise floor that limits the dynamic range of a properly dithered digital signal.
  The amount of bits you have represents the amount of error signal you'll get from quantization (rounding to steps), and the error signal will (if the signal is properly dithered) be part of the noise floor.
  More bits will just lower the noise floor and give you more dynamic range. Nothing is lost unless the signal goes below the noise floor of the system.
   
  And you can shape the noise floor to greatly extend the dynamic range at certain frequencies. You can, for example, have 120dB of dynamic range (20Hz-20kHz) in a 1 bit system (like SACD).
  So when it comes to sound quality it doesn't matter if you have 16,7 Million or 2 qauntization steps, just that the noise floor is low enough to not be audible or mask the signal.


----------



## xnor

^ Precisely.
   
  xabu, here are some files I created to play around with:* (all 16 bit, 44.1 kHz)*
http://www.mediafire.com/?hy4n6e9cg1jt7s0 - 100, 1k and 10 kHz sine at below -80 dB
http://www.mediafire.com/?zq8t1ovnx7dn4q1 - the same, but below -100 dB
   
  And the spectrum of the 2nd file:
   

   
  As mentioned before, dithering (1 of the 16 bits) was used to keep the noise floor down. And there are a lot of options to shape that noise to make it less audible than white noise.
   
  Also (try to) listen to those files.


----------



## xabu

Quote: 





d. lundberg said:


> And you can shape the noise floor to greatly extend the dynamic range at certain frequencies. You can, for example, have 120dB of dynamic range (20Hz-20kHz) in a 1 bit system (like SACD).


 

  
  Because you have only 1 bit you have more samples to "emulate" the missing bits (2.8224 MHz)  ... DSD and Sigma Delta DACs work completely differently from Ladder DACs and PCM.
  And and as far as I know they use now up to 5 bits in modern Sigma Delta DACs.


----------



## xabu

Quote: 





xnor said:


> ^ Precisely.
> 
> xabu, here are some files I created to play around with:* (all 16 bit, 44.1 kHz)*
> http://www.mediafire.com/?hy4n6e9cg1jt7s0 - 100, 1k and 10 kHz sine at below -80 dB
> ...


 

 I don't know what you did expect ... there is a tone audible for sure in both files.
   
  What was it you wanted to state?
  That 16 bit is enough to store this low tones?
  ... sure it's enough to store "some" value ...
   
  Example:
  If youd had only 9 possible values at hand to store something in a range from 0 - 60 (just some mathematical example, whatever these values represent doesen't matter they are just integers) and let's say these 9 possible values were 0, 8, 15, 23, 31, 49, 46, 54, 60. If you needed to store the value 44 it would be stored as 46.


----------



## xnor

Quote: 





xabu said:


> I don't know what you did expect ... there is a tone audible for sure in both files.
> 
> What was it you wanted to state?
> That 16 bit is enough to store this low tones?
> ...


 

 The purpose was to show that it doesn't work the way you described it in your example... (you would have noticed if you'd actually taken a closer look at the files)
  and it's not just "some" value t_t 
   
  And your last example shows this misunderstanding again. It doesn't matter if single samples don't _exactly _'hit' the exact value. 
  Digital audio is a _bit _more complicated..


----------



## xabu

Quote: 





xnor said:


> Quote:
> 
> 
> 
> ...


 

 Well, now we get to the point 
   
  You wrote about the analog result of reproduction form the digital domain.
  Obviously this is continuous, no doubt about that ... would be a bad D A C if not ...
  But I wrote about the digital domain only so far and digital resolution, the A D C.
   
  Regarding the analog reproduction I'm still not convinced that more digital resolution does not result in better reproduction ... among other things because of the fact that it is not possible to implement the nyquist theorem perfectly in practice.


----------



## xnor

Quote: 





xabu said:


> Regarding the analog reproduction I'm still not convinced that more digital resolution does not result in better reproduction ... among other things because of the fact that it is not possible to implement the nyquist theorem perfectly in practice.


 
   
  That's good enough and it doesn't have to because nothing is perfect in analogue electronics.  
	

	
	
		
		

		
		
	


	



   
  Why not convince yourself? Take a 24-bit file with high dynamic range, reduce the sample size to 16 bits (with dithering) and compare the files.
   
  Done that .. good luck.
  
  Or regarding perfect, infinite filters: read a book about digital signal processing first, and then convert an IIR into a FIR filter, apply both and compare the results. 
	

	
	
		
		

		
		
	


	




 (my EQ makes use of this)


----------



## xabu

Quote: 





xnor said:


> Quote:
> 
> 
> 
> ...


 
   
  Correct, making decisions about sound quality by actually listening to something is a really good point.  But I also want to comprehend it mathematically / technically ... with *graphs *etc. 
	

	
	
		
		

		
		
	


	



   
  As I said Im not yet convinced in one or the other way ... I still need more information about the whole subject matter ... I'm still learning


----------



## xnor

edited previous post a bit
   
  Well, there's a lot of software (free) and also hardware out there to analyze and measure stuff. People have done that too. You'll just see tiny amounts of noise added, most probably completely inaudible in playback systems.
  Compare this to lossless -> mp3 conversion. That adds tons of noise in comparison, clips, distorts the waveform and cuts off higher frequencies and still, it's transparent in many/most cases...


----------



## D. Lundberg

Quote:  





> Because you have only 1 bit you have more samples to "emulate" the missing bits (2.8224 MHz)  ... DSD and Sigma Delta DACs work completely differently from Ladder DACs and PCM.


 
  You can't use more samples to emulate "missing" bits, and more importantly: there are no "missing" bits to emulate! More samples will just extend the bandwidth and more bits will just lower the noise floor.
  DSD (which is based on Delta/Sigma) is actually a PCM-format and works the same way. The difference is that it needs noise shaping to be of any use, and that is what the extra bandwidth is used for.
  The total dynamic range of a 1bit/2.8224MHz system is still 6dB, but lots of noise is moved upwards to lower the noise floor in the audible range.
   
   Quote:


xabu said:


> Regarding the analog reproduction I'm still not convinced that more digital resolution does not result in better reproduction ... among other things because of the fact that it is not possible to implement the nyquist theorem perfectly in practice.


 
  Thankfully you don't need to perfectly implement the Nyquist theorem (at least not for use in audio). You just need to implement it well enough.
  Here is an introduction to how the theorem works in practice:
 http://www.lavryengineering.com/documents/Sampling_Theory.pdf


----------



## xabu

Quote: 





xnor said:


> Or regarding perfect, infinite filters: read a book about digital signal processing first, and then convert an IIR into a FIR filter, apply both and compare the results.
> 
> 
> 
> ...


 


  Well ...  to late today ... now I'd rather listen to some nice music 
	

	
	
		
		

		
		
	


	




 (... convert ...you mean by actually doing this ... programmatically ? what both ? compare what with what ?  EQ ?  we're derailing ... ... ah, I think it's too late in the evening ... )


----------



## xabu

Quote: 





d. lundberg said:


> Quote:
> 
> 
> 
> ...


 
   
  Oooo.k. So the dynamic range of music stored on a SACD is only 6 dB  ... if you say so ...
   
  Using "more samples to emulate missing" bits was a figure of speech ...
   
  You're aware that 1 bit can hold only a quantity of 2 values? Can you actually explain how it is achieved that this 1 bit system nevertheless can handle/hold more than only 2 different values? It's achieved via the very fast switching of this 1 bit "switch".
   
  That's my last post regarding this matter, because the title of this Thread is "24bit vs 16bit, the myth exploded!"


----------



## xnor

Yup, one/each bit = 6.02 dB of dynamic range, without applying techniques like noise shaping of course.
   
  ...but PCM != PDM, and that's the explanation right there


----------



## D. Lundberg

Quote: 





> Originally Posted by *xabu* /img/forum/go_quote.gif
> Oooo.k. So the dynamic range of music stored on a SACD is only 6 dB  ... if you say so ...


 
  No, the total dynamic range of the system is ~6dB (0-1441kHz). The dynamic range in the audible range (20Hz-20kHz) is much larger (thanks to noise shaping).
   
  Quote: 





> You're aware that 1 bit can hold only a quantity of 2 values? Can you actually explain how it is achieved that this 1 bit system nevertheless can handle/hold more than only 2 different values? It's achieved via the very fast switching of this 1 bit "switch".


 
  Rounding to 2 steps means you'll get tons of quantization errors and those errors will represent white noise (if the signal is properly dithered). So less steps means more noise and thus higher noise floor.
  I know it is hard to grasp that 2 steps can represent all the compexities of a waveform, but they very effectively can! Even if the sample rate is 44.1kHz the waveform will still be accurately represented in a 1 bit system (but the dynamic range will be very limited).


----------



## xabu

Quote: 





> Originally Posted by *xnor* /img/forum/go_quote.gif
> 
> ...but PCM != PDM, and that's the explanation right there


 

 Exactly! And I'm glad we have some consent on that ... (... well, I pondered putting a 
	

	
	
		
		

		
		
	


	




 behind each of the following links ... 
	

	
	
		
		

		
		
	


	




 ...)
   
   
  And here's some more explanation of the rest ... but I still need some time to fully get the correlation between this 6 dB per bit stuff regarding voltage and accoustics.
  You have to scroll down a bit 
	

	
	
		
		

		
		
	


	




 to get more text on the following sites ...
   
http://www.experiencefestival.com/a/Decibel_-_Reckoning/id/4843526
http://www.experiencefestival.com/a/Decibel_-_Uses/id/4843524
http://www.experiencefestival.com/a/Decibel_-_Typical_abbreviations/id/4843525
http://www.experiencefestival.com/a/Decibel_-_Definition/id/4843522


----------



## Vitor Machado

There is one thing I don't understand though:
   
  Wouldn't it make more sense to use the extra bits not for increased dynamic range, but for more gradual steps in the quantization?
   
  Suppose we have 3 bits (possible values: 000, 001, ... 111), and they represent, in order, -30, -20, -10, 0, 10, 20, 30, 40.
  If we add one more bit, instead of going like -70, -60, ... , 70, 80 (which seems to be the analogous case for 16 vs. 24 bits if I understand correctly), why not -30, -25, -20, ... , 30, 35, 40, adding more steps instead of bigger range?
   
  If someone could clarify things with some practical example like this it would be great! 
	

	
	
		
		

		
		
	


	



   
  EDIT: Better numbers for my example.


----------



## xnor

Quote:  





> If someone could clarify things with some practical example like this it would be great!


 

 First post in this thread, followed by probably another hundred attempts of explanations to those who do not want or care to read a book about digital audio / signal processing, attend a course or at least read the Wikipedia articles that cover the basic fundamentals.
   
  To repeat myself, the topic's a bit more complex than just dividing some numbers.


----------



## Vitor Machado

Quote: 





xnor said:


> First post in this thread, followed by probably another hundred attempts of explanations to those who do not want or care to read a book about digital audio / signal processing, attend a course or at least read the Wikipedia articles that cover the basic fundamentals.
> 
> To repeat myself, the topic's a bit more complex than just dividing some numbers.


 

 1 - I did read the first post, and the first page. Did not read the whole thread because it's huge.
  2 - I'm not going to read a book about digital audio / signal processing because I am busy with college, and have no such time available to delve deeply into the matter.
  3 - I've read the Wikipedia articles and they did not cover my question with necessary detail.
   
  I did find this example by _xabu_, a couple of pages ago:
   
  "So lets say
   
  with 1 bit we would resolve 6 dB into 2 values (e.g. 0 dB and 6 dB, nothing in between)
  with 2 bit we would resolve 12 dB into 4 values (e.g. 0,4,8,12 dB, nothing in between)"
   
  But that's strange, because people usually say to use 24-bits playback even if your source is 16-bits, because the source will be *padded* to 24-bits, and you get more range to waste with the digital volume control. But in this example, you just can't pad the 1-bit source to 2-bit at all (not without rounding 6 to 4 or to 8). So is everyone wrong in using 24-bit playback then (for 16-bit sources)?


----------



## xnor

Quote: 





vitor machado said:


> "So lets say
> 
> with 1 bit we would resolve 6 dB into 2 values (e.g. 0 dB and 6 dB, nothing in between)
> with 2 bit we would resolve 12 dB into 4 values (e.g. 0,4,8,12 dB, nothing in between)"
> ...


 
   
  Forget the 1 bit stuff if you didn't read the DSD, PDM articles. And this example/question doesn't make sense to me.
   
  And please reconsider reading the first post...
   
  (and I'm not sure if you even read Wikipedia articles, sorry if that's harsh)


----------



## Vitor Machado

Quote: 





xnor said:


> Forget the 1 bit stuff if you didn't read the DSD, PDM articles. And this example/question doesn't make sense to me.
> And please reconsider reading the first post...
> 
> (and I'm not sure if you even read Wikipedia articles, sorry if that's harsh)


 
   
  I'm fairly sure he's talking about PCM in this example I quoted.
   
  But let's just keep it simple then: should I set my sound card to 24-bits, even if my source is 16-bits, or not? Will it be able to pad (in a lossless way) the data?
   
  Why am I asking? Because if the steps are different in different bit-depths, then you cannot convert from one to the other without having to round stuff.
   
*EDIT 2:*
   
  Did some serious tests here with an HEX editor and some wave files, and found out that you can pad perfectly from 16 bits to 24 bits.
   
  If you have, say, (big endian) FF 7F then pad it to 24-bits you get 00 FF 7F. The first 00 are the least significant part and account for the smaller steps. So for every step possible in 16-bits, you get exactly 255 new steps in between in 24-bits.
   
  I still don't understand how does this increase so much the dynamic range though, since the lowest point ("silence") is 00 00 00 (the same as 16-bits) and the highest is now FF FF 7F for the crest *, and FF FF 7F is just barely higher than FF 7F.
   
* It's not FF FF FF because after 7P it wraps around and represents the trough.
   
   
  This image is what I found out through testing (HEX viewer and looking at the wave):


----------



## khaos974

Suppose that you have a system where the values are 0, 1, 2, 3, this is effectively 2 bit. and that you was a system that does 0, 0.5, 1, 1.5, 2, 2.5, 3, 3.5, In your mind, you'd have doubled precision of the quantization.
  Let's say that the digital values 0, 1, 2 ,3 represent 0, 1, 2, 3 mV out of your microphone, you want to record 0.5 mV steps, you simply double the pre amp values for your mike, making the the output values go from 0 to 7 mV by 1 mV steps which means you have doubled the dynamic range necessary to record the data. Effectively, to diminish the quantization error, you simply increase the dynamic range. You now have to use a 3 bit encoding with 0, 1, 2, 3, 4 ,5 ,6 ,7 as values.
   
  Anyway, most recording are 24 bit today, allowing the smallest step to be set very low, and once digitized, every track (one for the piano, one for the singer...) is fed to the DAW (digital audio workstation) which performs its internal calculation on 32 and even 64 bit. And finally when you output your mastered song, you decide the values of your loudest and softest sounds considering that a variation smallest than the softest sound wouldn't be heard. This, for a CD is 16bit, ie. 96 dB of variation between the loudest and the softest sound, ie 65535 values, when you go up on and High Def, you get 144 dB, which means that either your loudest sound gets louder, or that you softest sound get softer, increasing dynamic range is the same as decreasing quantization error.
   
  Except that, your listening room has noise for, even in an anechoic room you, if your hearing is very good, you'd hear your heart beat, the flow of your blood... So the softest sound, the smallest variation doesn't have to be that small, there's a limit for that. Now consider your 16 bit CD, your room noise floor is 25 dB (that's a very quiet room), you set the volume so that the highest sound reaches 120 dB (That's really loud), your softest sound is played at 24 dB, ie. lost in the noise floor. If you play so that the loudest sound is 90 dB, the softest sound plays below 0 dB which is below the audibility threshold. Of course this does not take into account dithering, which brings the perceivable dynamic range up to to 120 dB with a 16 bit encoding, considering a loudest level of playback at 120 dB, the small detail is right at the audibility threshold

 As an addendum, your question was mostly answered in the first two paragraphs, the 3rd was mostly about why 16 bit is usually enough for playback in household conditions. That's with PCM encoding, SACd is a different story.
   
   


vitor machado said:


> There is one thing I don't understand though:
> 
> Wouldn't it make more sense to use the extra bits not for increased dynamic range, but for more gradual steps in the quantization?
> 
> ...


----------



## xnor

Quote: 





vitor machado said:


> I still don't understand how does this increase so much the dynamic range though, since the lowest point ("silence") is 00 00 00 (the same as 16-bits) and the highest is now FF FF 7F for the crest *, and FF FF 7F is just barely higher than FF 7F.


 

 FF FF 7F = 2^23-1 = 8388607 is just _barely _higher than FF 7F = 2^15-1 = 32767 ???
  The multiplier is 2^8 = 256, or +48 dB !!!
   
  And please, I'm begging you, read the first post.


----------



## xabu

Quote: 





khaos974 said:


> Suppose that you have a system where the values are 0, 1, 2, 3, this is effectively 2 bit. and that you was a system that does 0, 0.5, 1, 1.5, 2, 2.5, 3, 3.5, In your mind, you'd have doubled precision of the quantization.
> Let's say that the digital values 0, 1, 2 ,3 represent 0, 1, 2, 3 mV out of your microphone, you want to record 0.5 mV steps, you simply double the pre amp values for your mike, making the the output values go from 0 to 7 mV by 1 mV steps which means you have doubled the dynamic range necessary to record the data. Effectively, to diminish the quantization error, you simply increase the dynamic range. You now have to use a 3 bit encoding with 0, 1, 2, 3, 4 ,5 ,6 ,7 as values.


 
    
  Doesn't that confirm what I stated earlier.
  With 2 dB and 0,1,2,3 in 1mV steps and then with 3 dB and doubled preamp values you have still 1mV steps but you have now 8 of theses steps for the same stuff reaching the microphone as before.
  What ever comes after like mastering, filtering .... Effectively you have more values for the same recorded analog sound, more "resolution".
  The point of view of the microphone is still having 1mV values so no more resolution than before, but more dynamic range.
  It is that mV and SP stuff I can't get congruent with each other yet.
   
  I'm going now, buying some Webers or Dickreiter ...
   
   
  Quote:


khaos974 said:


> Except that, your listening room has noise for, even in an anechoic room you, if your hearing is very good, you'd hear your heart beat, the flow of your blood... So the softest sound, the smallest variation doesn't have to be that small, there's a limit for that. Now consider your 16 bit CD, your room noise floor is 25 dB (that's a very quiet room), you set the volume so that the highest sound reaches 120 dB (That's really loud), your softest sound is played at 24 dB, ie. lost in the noise floor. If you play so that the loudest sound is 90 dB, the softest sound plays below 0 dB which is below the audibility threshold. Of course this does not take into account dithering, which brings the perceivable dynamic range up to to 120 dB with a 16 bit encoding, considering a loudest level of playback at 120 dB, the small detail is right at the audibility threshold


 
   
  You can't hear tones which dB value is below noise floor dB value (or, if you only have your breath and heartbeat you couldn't hear sound below that ?) ?
  Wouldn't that mean you couldn't here e.g. a triangel if it played simultaneously with a trumpet, because the trumpet is much louder ?
  I know that noise floor contains many frequencies. I'm just curious, not stating you were completely wrong or something.
   
  And I'm with you regarding the rest so far.


----------



## nick_charles

xabu said:


> You can't hear tones which dB value is below noise floor dB value (or, if you only have your breath and heartbeat you couldn't hear sound below that ?) ?
> Wouldn't that mean you couldn't here e.g. a triangel if it played simultaneously with a trumpet, because the trumpet is much louder ?
> I know that noise floor contains many frequencies. I'm just curious, not stating you were completely wrong or something.
> 
> And I'm with you regarding the rest so far.


 

 Masking requires that the two frequencies are in proximity, the further apart they are the less masking works. You can hear some music beneath the noise floor but I cannot remember offfhand how much, I think 20db below the noise floor is not possible, but I may be wrong..


----------



## emmodad

also, pls recall that this entire "24bit vs 16bit, the myth exploded" thread contains numerous posts (from beginning to most-recent) presenting significant misinformation wrt fundamentals of sampling theory and digital signal processing -- neither of which are particularly simple topics...
   
  for some reasonably-understandable texts providing technically-correct info -- and specifically to assist in understanding the relationships between bit depth, quantization error, resolution and dynamic range (which in this thread, sadly are oft-misrepresented) -- see the linked references in these posts:
   
  from this thread:  #463
   
  from the thread (short, easy reading) "Bit depth and sampling frequency explained":  #15, #19 and #21
   
   
  when i last checked, majority of the excellent and straightforward Pohlman text (Principles of Digital Audio) can still be accessed on the link to Google books; and the Analog Devices "Introduction to Sampled-Data Systems" presentation is available at the noted analog.com link


----------



## Vitor Machado

Quote: 





xnor said:


> Quote:
> 
> 
> 
> ...


 
   
  Dude, I've read the first post, I already told you I did. Seriously, you're just making yourself seem arrogant. No one else in this thread is behaving like this, chill out...
   
  I was considering a different step size too.
  If you pick up a 16-bits file with an FF 7F value, then pad it to 24-bits, you'll get 00 FF 7F. Now if you change it to FF FF 7F in an editor, all you'll *see* is a marginal, almost imperceptible difference in the displayed wave form. Because now, if you view it like I did, the possible steps are also 256 times smaller. You do not *see* the waveform shrink 256 times when you pad it to 24-bits, it's the steps that are reduced.
   
  So I guess the problem here is that you can view it in two ways: the one above, considering the steps to be 256 times smaller, then the dynamic range is barely changed (this is the way I've been viewing it, because it feels more natural to me). Or if you consider steps of the same size, then dynamic range is indeed 256 times larger. This seems to be the correct approach, if I understand _khaos974_ explanation correctly.
   
  Here is what you will *see,* if you pad a 2-bit source to 4-bit:

   
  IMO, the most *natural* way to interpret it, is to think the possible steps have become smaller, and dynamic range has barely increased from 0011 to 1111 (it's a small difference *here*, because the steps are also 4 times smaller).
   
  But it seems the *correct* way to interpret it is: the steps are the same size, but dynamic range quadrupled. When the file was padded, it was just "stretched" to space out in this new dynamic range.


----------



## khaos974

Quote: 





xabu said:


> Doesn't that confirm what I stated earlier.
> With 2 dB and 0,1,2,3 in 1mV steps and then with 3 dB and doubled preamp values you have still 1mV steps but you have now 8 of theses steps for the same stuff reaching the microphone as before.
> What ever comes after like mastering, filtering .... Effectively you have more values for the same recorded analog sound, more "resolution".
> The point of view of the microphone is still having 1mV values so no more resolution than before, but more dynamic range.
> ...


----------



## khaos974

What's your point exactly?

 If your output is your computer, padding to 24 bit is useful if you need a digital volume control, otherwise outputing 16 bit is roughly equivalent, you DAC may do the padding automatically pre conversion anyway.
  
  Quote: 





vitor machado said:


> ...
> IMO, the most *natural* way to interpret it, is to think the possible steps have become smaller, and dynamic range has barely increased from 0011 to 1111 (it's a small difference *here*, because the steps are also 4 times smaller).
> 
> But it seems the *correct* way to interpret it is: the steps are the same size, but dynamic range quadrupled. When the file was padded, it was just "stretched" to space out in this new dynamic range.


----------



## Vitor Machado

Quote: 





khaos974 said:


> What's your point exactly?
> 
> If your output is your computer, padding to 24 bit is useful if you need a *digital volume control*...


 
  This is my point.


----------



## khaos974

Based on your sig, I suppose it's a question for foobar?
 I believe foobar volume control is 32 bit precision, it then outputs at the selected bit depth. You may want to double check the info though.


----------



## Vitor Machado

Quote: 





khaos974 said:


> Based on your sig, I suppose it's a question for foobar?
> I believe foobar volume control is 32 bit precision, it then outputs at the selected bit depth. You may want to double check the info though.


 

 Not really, I'm asking because I've been setting my Windows (Seven) mixing bit-depth to 24-bits (shared mode), so I can use the digital volume control without losing so much useful data.
   
  It's also for curiosity's sake.


----------



## xabu

Quote: 





khaos974 said:


> Quote:
> 
> 
> 
> ...


 
   
  1. How is this fixed 6 dB/1 bit thing to mV to SP related, and is this a physical founded relation or could you do it differently?
  2. When I'm playing a continous volume sweep e.g from 30 dB to 80 dB why is it that with 24 bit there is not more information about that volume sweep recorded as with 16 bit ? Is this just because of the mastering technics used or is this some kind of mathematical/physical law? I' m aware that also the dynamic range grows as makes your previous example clear (because the 7 mV is something more than before with the 3 mV max. which doubled is only 6 mV, so we have more range for a higher volume value as before, am I right?)
   
  These are my main problems of understanding currently.
   
  (The Dickreiter did not offer much information on that specific theme ... but I will read the links given in the post somewhat above)
   
  EDIT: O.k. I'm out. I started to read the first 4 to 5 pages of the thread and the mentioned related threads and will continue by and by, also read the mentioned Posts and adjacent posts.
  As it seems to me, the main problem in the discussion is, that some people like me argue theoretic and others regarding practical aspects.
  So there is a "resolution" increase with more bits, but it is not used because it is not considered practical necessary due to other technics used to get it right, etc. etc.
  Which technics are then considered better or even make audible difference seems to be more a matter of taste, belief or academical.
  As you say in German: Die Augen essen mit.
   
  Also I think no one here (of course including me) is able to fully comprehend the involved mathematics e.g. Nyquist et al.
   
  I'm reading on ... 
   
  And I'm glad my Hifiman has a (no, two) PCM1704


----------



## Gospel Guy

Makes perfect sense. I'm unable to hear any difference. Another example of SQ voodoo.


----------



## ramicio

24 bits are 16,777,216 possible values, 16 bits are 65,536.  So you suggest that the other 16,711,680 values are all below 96 dB?  If things are linear, then each decibel of a 24 bit file can be broken down to 1/116,508.  Each decibel in a 16-bit file can be broken down to 1/682.  So yes, there is more resolution, and it's not all about dynamic range.  I can't stand the view that 16/44.1 is all anyone will ever need, so let's not progress and stay with an ancient format forever.  It could be the same view that people can survive without needing computers, and people can live without technology, so why bother with it?  People also seem to think there will be no benefit from higher than 16/44.1 if each instrument was only captured in such a "resolution."  I find this false and stupid.  It would only matter if the song is just that one instrument and nothing else.  But you mix other sounds together, how could it not benefit from say 24/192?  Everything is just math here.  In my mathematical opinion a song is going to sound more natural, and a DAC is going to have to do less work trying to reconstruct a waveform at 24/192 than 16/44.1.


----------



## xnor

Quote: 





ramicio said:


> 24 bits are 16,777,216 possible values, 16 bits are 65,536.  So you suggest that the other 16,711,680 values are all below 96 dB?
> 
> If things are linear, then each decibel of a 24 bit file can be broken down to 1/116,508.  Each decibel in a 16-bit file can be broken down to 1/682.  So yes, there is more resolution, and it's not all about dynamic range.
> 
> ...


 

 1) 20 * log10( 1 / 65536 ) equals guess what? -96... dB. Feel free to fill in all the numbers up to 16777216 into the same formula. Just math, eh?
   
  2) Oh, linear decibels again, how hilarious!
   
  3) Right, 192000 samples per second with larger sample size are much simpler and faster to process than only 44100, especially in playback devices without much processing power. And the hardware to process it is cheaper too!
   
  t_t'


----------



## nick_charles

Quote: 





ramicio said:


> 24 bits are 16,777,216 possible values, 16 bits are 65,536.  So you suggest that the other 16,711,680 values are all below 96 dB?  If things are linear, then each decibel of a 24 bit file can be broken down to 1/116,508.
> 
> *The db scale is not linear it is logarithmic*
> 
> ...


----------



## JerryLove

Quote: 





> Originally Posted by *nick_charles*
> 
> 
> 
> The db scale is not linear it is logarithmic


 
   
  In regards to power yes, in regards to audiability no. 3db (thereabouts) is the threshold of audiability: whether we are discussing the difference between 75 and 78 db or 105 and 108db.
   
  Since hearing aberrations is linear in db (if not, we need to toss out those useless FR charts now), it makes sense that you would set resolution linear in db.
   
  Mind you, you could also cheat and set a variable resolution, after all the different values below the noise floor are irrelevent, as is any noise below the hearing threshold: but things would get a bit complex. Simpler to use enough bits for the desired dynamic range at the desired resolution from end to end.
   
   
  Quote: 





> What you have is a voltage that can be split into different numbers of slices. A full scale CD line-out signal is a nominal 2V so you can do the maths on that. At that point you need to demonstrate that this extra subdivision of voltages is audible. With a 2V signal sliced by 16 bits the difference between say 32766 and 32767 is 0.00003V , I wonder how audible this difference is , I'll give you a clue it isn't. It is way below human discrimination.


 
   
  Here I agree: though I think we actually have two questions.
   
  Dealing solely with end-user listening: the resolution exceeding the ear is the standard. When we start discussing something which may undergo more processing: working with as high a resolution as you can get away with helps minimize problems in the end product.
   
  I think too we agree on the rest. Higher resolution is closer to reality: but generally not in a useful way. If anything, I'd think you might want to play with the 44Khz, to allow for VHF harmonics with minimal distortion: but I'm just guessing that it might matter.


----------



## wyager

I'm a little confused by the OP-It said that each bit represented a 6dB increase in maximum range-but doesn't this assume the same scale for all recordings?
  Let's say you are playing a song on your ipod, and the output reaches a peak voltage of ±1.28v (just as an example). The total voltage range is 2.56 volts.
 If you have 8-bit audio, there are 256 discreet voltage levels that the DAC can produce. This means that every voltage "step" is .01v. This would probably sound pretty grainy. Here is an accurate graph of a 100hz audio signal played at 8-bit resolution (assume that y=voltage and x=time in seconds).

  As you can see, the audio is visibly "choppy", and this would probably not sound so great.
  Now, if I understand what the OP is saying, it means to tell me that  16-bit audio looks like this (graph scale is adjusted along time axis to make steps visible):

  where the range is exponentially wider, but scale stays the same... I have to be misunderstanding this part. Because if the OP really says what I think it does, then it says that the number of bits determine the maximum volume, which is just plain wrong...
  The way I think of it, if you have 16-bit audio then there are 65536 discreet voltage levels between the same two peak voltages, and it looks more like this:

  ie. smoother and less "digital".
   
  Am I misunderstanding how this works?


----------



## xabu

Ah, well, the thing is ... the membran of a speaker can only do two things ... move into one direction or into the reverse direction ... in between there's continous motion of that membran ... so there is no such thing as digital sound ... sound is always analog and continous.
  What you're visualizing is the side of an ADC.


----------



## wyager

Quote: 





xabu said:


> Ah, well, the thing is ... the membran of a speaker can only do two things ... move into one direction or into the reverse direction ... in between there's continous motion of that membran ... so there is no such thing as digital sound ... sound is always analog and continous.
> What you're visualizing is the side of an ADC.


 


  So by that logic, square wave sound sounds just the same as sine wave sound? That's not the case... The response time of a small speaker is so fast that relying on it to perform mechanical low-pass filtering and/or mechanical signal smoothing is stupid (which is why we need high-resolution non-PWM ADCs).


----------



## xnor

No, see http://en.wikipedia.org/wiki/Anti-aliasing_filter


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## wyager

Quote: 





xnor said:


> No, see http://en.wikipedia.org/wiki/Anti-aliasing_filter


 

 Are you responding to me, or Xabu?
  Can you explain what you mean? I don't understand what you are negating with the link to the AA filter article...


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## xabu

Quote: 





wyager said:


> Quote:
> 
> 
> 
> ...


 

 A membran is actually not able to produce the motion of air equivalent of an ideal square wave ... accoustical square waves are physically impossible ... you only get approximations.
   
  But thats not what I wanted to get at. I just doubt that a speaker is able to reproduce the signal which your first graph shows.
   
   
   
  ... and I'm still trying to get the picture of the relation between a range of possible volume levels depending on the bit depth and the bit dependency of the "resolution" of a specific wave at a specific volume level ...


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## wyager

Quote: 





xabu said:


> Quote:
> 
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> ...


 


  Ah, OK, I understand now.
  Yes, I recognize an acoustical square wave is impossible (for now), but what I was saying is that the response times of small speakers are so fast that you can get VERY close to a true audible square wave. The response times of a small speaker are also so fast, and our ears so sensitive, that the small 1/256 volume jumps are, in fact, audible and decrease overall sound quality. Even if speakers did act as perfect mechanical smoothing filters and square waves at x frequency sounded just like sine waves at x frequency, what happens when you try to overlap multiple waves? Have you ever tried to add separate waves with 1 bit audio resolution (in a non ∆∑ configuration)? It just doesn't work. (If you want me to make a snazzy graph showing why 1-bit wave addition doesn't work, I'd be happy to.) 
	

	
	
		
		

		
			




   
   
  And as for your last sentence, look at this... the OP said "So, 24bit does add more 'resolution' compared to 16bit but this added resolution doesn't mean higher quality, it just means we can encode a larger dynamic range." and "4 = Dynamic range, is the range of volume between the noise floor and the maximum volume."
   
  OP seems to be suggesting that more bits=more volume, which is wrong... more bits=more discreet volume levels (a smoother curve).


----------



## xabu

Quote: 





wyager said:


> Ah, OK, I understand now.
> Yes, I recognize an acoustical square wave is impossible (for now), but what I was saying is that the response times of small speakers are so fast that you can get VERY close to a true audible square wave. The response times of a small speaker are also so fast, and our ears so sensitive, that the small 1/256 volume jumps are, in fact, audible and decrease overall sound quality. Even if speakers did act as perfect mechanical smoothing filters and square waves at x frequency sounded just like sine waves at x frequency, what happens when you try to overlap multiple waves? Have you ever tried to add separate waves with 1 bit audio resolution (in a non ∆∑ configuration)? It just doesn't work. (If you want me to make a snazzy graph showing why 1-bit wave addition doesn't work, I'd be happy to.)
> 
> 
> ...


 

 You may be able to differentiate between more than 256 different volume levels, but not "inside" one wave and at the scale you mentioned ... try to translate that to dB ... you're max. able to differentiate approx. 1 dB differences in volume levels (...hmmm ... actually that would translate to only 140 max. steps before you you experience irreversible damage to your ears ... on the other side, sensitivity varies depending on frequency and volume ... so you may be able to differentiate 0.25 dB in some areas ...)
   
  Hm ... and if we take this practical approach with anti aliasing filter into account ... there should practically no problem at all.


----------



## wyager

Quote: 





xabu said:


> Quote:
> 
> 
> 
> ...


 


  I beg to differ... I just edited an app I had that generated a sine wave-I set it to 100 hertz. Then I added a line that effectively made the audio 8 bit, and the difference was VERY obvious to my ears, even with just a single wave. And I'm not sure what you're saying about an AA filter, all an AA filter does is remove frequencies above the Nyquist frequency (usually 22050hz) during sampling I believe... it has nothing to do with smoothing the audio.


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## khaos974

What is usually done when a * bit signal is generated is that the signal is actually dithered down to 8 bit, ie quantization error is replaced by noise.
  So a higher bit depth indeed means a greater dynamic range.


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## wyager

Quote: 





khaos974 said:


> What is usually done when a * bit signal is generated is that the signal is actually dithered down to 8 bit, ie quantization error is replaced by noise.
> So a higher bit depth indeed means a greater dynamic range.


 

  
  Umm, turning something from >8bit to 8bit doesn't replace quantization error, it _creates it_... LOL. From the wiki article on quantization error-"This error is either due to rounding or truncation." When you turn audio from >8bit to 8bit, you either round or truncate it. If you mean the other way around, then true, higher bit depths tend to have more noise, but saying that quantization error is better than noise is like saying that having a calculator that does integer-only math with 100% minimum accuracy is better than a calculator that does 10-digit floating point math with 99% minumum accuracy. I'm not sure if that _is_ what you are saying, as text doesn't convey tones of voice properly, but if so then I might as well respond to that argument.
   
  And again, I'm not sure why you say it means a greater dynamic range... the maximum volume stays the same. The computer just scales the voltage value of a single bit.


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## khaos974

From wiki, dither article:
   
  Quote: 





> > In an analog system, the signal is _continuous_, but in a PCM digital system, the amplitude of the signal out of the digital system is limited to one of a set of fixed values or numbers. This process is called quantization. Each coded value is a discrete step... if a signal is quantized without using dither, there will be quantization distortion related to the original input signal... In order to prevent this, the signal is "dithered", a process that mathematically removes the harmonics or other highly undesirable distortions entirely, and that replaces it with a constant, fixed noise level.[10]


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## wyager

Quote: 





khaos974 said:


> From wiki, dither article:
> 
> Quote:
> 
> ...


 

 Ah, I see what you are saying now! However, the dither noise is not necessary... why don't you just keep it at 24 bit and dither that? And this still doesn't change the fact that the maximum volume stays the same...


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## Confispect

Interesting......


----------



## JerryLove

Quote: 





xabu said:


> You may be able to differentiate between more than 256 different volume levels, but not "inside" one wave and at the scale you mentioned ... try to translate that to dB ... you're max. able to differentiate approx. 1 dB differences in volume levels (...hmmm ... actually that would translate to only 140 max. steps before you you experience irreversible damage to your ears ... on the other side, sensitivity varies depending on frequency and volume ... so you may be able to differentiate 0.25 dB in some areas ...)
> 
> Hm ... and if we take this practical approach with anti aliasing filter into account ... there should practically no problem at all.


 

 8-bit = 127 volume levels. remember that the wave is both positive and negative. (256 -1 (for zero) then /2 (for sign) and rounded down to the nearest integer (since you'd want equal resolution in both directions))
   
  Remember also that (electrically) the smaller waves ride on the larger waves: so the wave amplitude goes through a larger set of fluxuations than the end volume does.
   
  If I understand you: you are saying that a 1db warble sounds the same as a 1db pure tone?


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## wyager

Quote: 





jerrylove said:


> Quote:
> 
> 
> 
> ...


 


  True about the zero. Also, technically since eg. -.05 and .05 volts are discreet, there ARE 254 volume levels (I believe a positive-only wave sounds different from a +/- wave. I'll test right now).  If the device did indeed see "zero" as a level (which I'm not sure they actually do), there would be both +0 and -0, even though they are the same. So 254 different voltages for 8 bit, I guess.
   
  Edit:
  Yep, a "rectified" sine wave sounds a lot more like a square wave at 2x that frequency.
  By "rectified", I mean


----------



## xnor

Quote: 





wyager said:


> Can you explain what you mean? I don't understand what you are negating with the link to the AA filter article...


 

 Such filters are part of the reconstruction process, so that the outcome is not a staircase-shaped mess (like you assumed), but a smooth curve.


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## xabu

Quote: 





jerrylove said:


> Quote:
> 
> 
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> ...


 

 Then you missunderstood 
   
  I simply stated that I doubt you will hear the difference in sound between this "sine" in the first graph compared to the second graph in the post I releated to.
   
  And again ... I think we're still confusing volume resolution and wave resolution. And as far as I gathered so far 
	

	
	
		
		

		
			





 all these bit values are not mainly used to resolve the wave. But maybe I still don't get the whole picture yet 
	

	
	
		
		

		
		
	


	




 but I try hard 
	

	
	
		
		

		
		
	


	



   
   
  Edited: ... was a little bit in a hurry while typing ...


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## wyager

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xnor said:


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  This is not true... For two reasons:
  1)Smoothing filters, in most cases, will only muddle and blur the audio. What happens if the song contains rapid, small voltage spikes (clicks)?
  2)As I said, I actually tried listening to the output last night, and it does sound like a "staircase-shaped mess". I'm actually glad, because if it didn't that would mean I have an awful DAC that can'd handle fast voltage switches.
   ​[/size]



  Quote: 





xabu said:


> And again ... I think we're still confusing volume resolution and wave resolution. And as far as I gathered so far
> 
> 
> 
> ...


 

  Hmmm... I thought "volume resolution" and "wave resolution" are the same thing? Can you explain the difference?


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## xnor

Quote:  


> [size=medium]This is not true...​[/size]


 
   
  So, you're telling me there are no filters in DACs. Right, I'm done here.


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## wyager

Quote: 





xnor said:


> Quote:


 


> > [size=medium]This is not true...​[/size]
> 
> 
> 
> ...


 

 Um, no.... I'm telling you that DAC designers aren't stupid enough to stick a huge capacitor on the output. I'm not really sure how you can try to dispute this, I literally listened to an 8 bit sine wave and noticed the voltage steps. No need to be dismissive.... (especially when you don't even really have an argument...)


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## xnor

Quote:  





> Um, no.... I'm telling you that DAC designers aren't stupid enough to stick a huge capacitor on the output. I'm not really sure how you can try to dispute this, I literally listened to an 8 bit sine wave and noticed the voltage steps. No need to be dismissive.... (especially when you don't even really have an argument...)


 
   
  Sorry but you gotta be kidding.
   
  a) there's _*DIGITAL*_ filters
   
  b) you don't need huge capacitors for a low-pass filter (you confuse this with big DC blocking caps = high-pass?)


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## wyager

Quote: 





xnor said:


> Quote:
> 
> 
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  I'm aware of this... but why would anyone use a digital filter to make the audio less sharp?
   
  Also, ideally low-pass filters wouldn't smooth out the audio at all. A capacitor would both filter out highs and smooth the wave, but again, no one in their right mind would filter the output enough to smooth out an 8-bit wave.
   
  But the point remains, *how do you explain the fact that I CAN hear the individual steps in a sine wave when listening to 8-bit audio? *Stop beating around the bush, just answer the question. If what you are saying is true, why does the output of an 8-bit wave on my laptop sound noticeably staggered (ie. *not smooth*)?


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## xabu

Quote: 





wyager said:


>


 


> 1)Smoothing filters, in most cases, will only muddle and blur the audio. What happens if the song contains rapid, small voltage spikes (clicks)?  2)As I said, I actually tried listening to the output last night, and it does sound like a "staircase-shaped mess". I'm actually glad, because if it didn't that would mean I have an awful DAC that can'd handle fast voltage switches.
> ​


 
  1) ... it happens to be a bad song ... (just kidding)
   
  I think you're the first person who is actually able to hear whats happening in a 1/100 second between max and min of a wave.
   
  As far as I know all possible accoustical sound is based on sine waves (even a square wave) ... combined and overlapping whatsoever ... and thats also why it's possible to reconstruct it almost perfectly.
   
  I don't know what you hear and how it is produced what you here, but I doubt sincerely it's actually "steps" ... (because accoustical steps are almost as impossible as accoustical square waves)
   
  Also these sine curves are just mathematical based graphical representations of voltage fluctuations ... how the hitten air behaves which than hits your eardrum would look differently.
   
   
  Apart from that.
   
  8 bit means you can hold values up to 256. But it's still only *one *value per sample (a value between 0 and 255 or -125 and + 124 or whatever). So with a given wave you don't have more values per second (as suggests your graphs) with 16 bit then with 8 bit or 1 bit. More values per second, thats the sampling frequency. So with more bits you can get a grip of a wider range of volume (what translates to more dynamic range ? ...  I think I get it a bit 
	

	
	
		
		

		
		
	


	




 of it now)
   
  This is also an interesting read. Try to read it all and understand all what's said there. (I couldn't yet)
   
http://www.lavryengineering.com/documents/Sampling_Theory.pdf


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## wyager

Xabu-
  Yes, you are correct, it's not like I am actually counting every single step, but the audio sounds "grainy". It's a little hard to explain, but the difference is very obvious. 
   
  And yes, you are correct again, I never meant to suggest that there are more samples per second with a higher bit depth, only that there are more possible values for the sample. My graphs weren't meant to suggest more samples per second. Think about it like this. Let's say that the maximum voltage your ipod can produce is 1.27 volts (as we learned might be more accurate). This means that, with an 8 bit audio file, the minimum voltage change is .01v. Now, if we used a 16 bit audio file instead, the sample rate stays the same (44100hz) and the max voltage stays the same, but the minimum voltage change is now .0000390625v, which is a much more accurate reproduction of the real sound than 8 bit audio can provide.


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## xabu

Quote:  





> Think about it like this. Let's say that the maximum voltage your ipod can produce is 1.27 volts (as we learned might be more accurate). This means that, with an 8 bit audio file, the minimum voltage change is .01v. Now, if we used a 16 bit audio file instead, the sample rate stays the same (44100hz) and the max voltage stays the same, but the minimum voltage change is now .0000390625v, which is a much more accurate reproduction of the real sound than 8 bit audio can provide.


 

  
  Well, that's the question, if it really works that way ... and shouldn't we consider the kind of DAC used, 16 bit or 8 bit or even 24 bit?
   
  Your simple mathematics seem right (well, see below), it's also almost the same as mine in my first posts here.
  But I'm not sure anymore if it is really handled that way by the ADC/DAC and the other components analog and digital involved.
  Also I have the impression as if a 24 bit file on my 801 sounds more loud than a 16 bit file with the same volume setting.
   
  And what happens if you plug an amplifer into your ipod which is capable of 5 volts or more (... and let*s not forget playing such a song via a dektop amplifier and 200 watt speakers ...)... would it just thin the sound  
	

	
	
		
		

		
		
	


	




 ?
   
  (... 8 bit = 256 values ... 1.27 / 256 = 0.005 ... or do I miss something ?)


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## wyager

Quote: 





xabu said:


> Quote:
> 
> 
> 
> ...


 

 That's true, the kind of DAC used may affect the output. You say that a 24 bit file sounds louder-but does it sound 256 times louder? That would mean it is acting as the OP says. And yes, if that amp increased the volume then I see no reason why the steps wouldn't be even more noticeable. And yes, for your last equation you are forgetting that the (hypothetical, but realistic) voltage range is actually +1.27 to -1.27, which makes for a total difference of 2.54 volts or .01v per step. 
   
  I'm not sure if this will make sense to anyone here, but the way I think of it whichever device you're using just puts the MSB of your 24/16/8/whatever bits in the MSB of the DAC's output register and makes the rest zeros. The way the original post says it would work is if the LSB of the audio file was matched to the LSB of the DAC (which seems unrealistic). So, let's say that your DAC is some 8 bit microcontroller (as that is both likely and simple to match up to 24/16/8 bit audio), and the song you are listening to is a 24 bit audio file. The current voltage in 24 bit binary could be 10101010, 10101010, 10101010 (.666666627v). With the the way I believe it works, if this file was downgraded to an 8 bit file, the first 8 bits of the original audio file would be copied and the binary value in the DAC would be 10101010, 00000000, 00000000 (.6640652v). As you can see, the volume change would be minimal (with lower quality audio). The way the OP is suggesting, the value instead would be 00000000, 00000000, 10101010, or about .00001013v. The volume in this case would be 65,536 times lower than the original 24 bit audio file, with no advantages in terms of processing power or even quality. It just doesn't make sense to me that anyone would choose option 2. Also, sorry if that was confusing, that just makes sense to me from a low-level software standpoint.


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## Lenni

I've not read the most of the thread as it gets too technical, but if it's of interest I've uploaded two flac files of the same song, for educational purpose (I hope is not against the rules).
   
  one is a Studio Master 24bit 96kHz from Linn Records
   
  the other is a CD rip
   
  the difference between these two file is quite drastic. one is indeed much louder than the other. but hear it for yourself link
   
  note: if for some reason the sharing of these file is not permitted, even for educational purpose, and you're interested in hearing the files, pm and I'll pm back the link.


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## ramicio

You won't convince people that way because mastering is quite different between the CD and HD tracks.  I still believe and have heard differences between 2 equally mastered 16/44.1 and 24/96 files by simply taking a 24/96 file and down
  sampling and dithering down the bit depth.  It's not drastic, but as I said before, we have the storage technology, so why not increase quality?  Everything else in the world has gone to increasing quality.  Hell most people who think 16/44.1 is enough also think you can't tell the difference between a 128 kbps AAC and the uncompressed source.  We used to have lossy audio on DVD, then another storage format came along and just multiplied the image resolution by 6 and we gained better video compression, so now we have room for lossless audio. They did it, why can't the music industry.  It would help protect them from theft because the majority of dumb people out there don't like dealing with large files over the internet and waiting.


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## Lenni

yes, the whole mp3 development was due low Internet speed and storage... this in not longer justifiable. however I struggle finding differences between mp3 and lossless these days myself.
   
  btw I don't wish to convince anybody, I have the files and thought might be relevant to the thread


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## ramicio

Lossy formats have stuck because the majority of people are going to internet downloads, either through purchase through places like iTunes or illegally.  Lossy is in the interest of the providers because it saves bandwidth.  The source end of anything could give 2 craps about the quality of the production and format as long as it sells.  The only way music seems to sound good anymore is if the artist cares at all, has their own label, or is an unpopular type of music.  Just about every artist with their own label are rapper entrepreneurs, so all they care about is money anyway, and it's part of pop so it's going to be loud.  I just think MP3 was one of the worst codecs as far as bitrate vs. quality.  Everyone has always said 128 kbps MP3 was transparent to CD audio, then when AAC came out it was 128 kbps with that.  I don't care for lossy at all and I can hear differences between lossy and lossless.  I only ever use lossy for portability and non-quiet environments.


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## xabu

Quote: 





wyager said:


> Quote:
> 
> 
> 
> ...


 
   
  My comments in *bold *above.
   
   
  Removed my thoughts ... not completely coherent yet 
	

	
	
		
		

		
		
	


	




, but I get it better and better ...


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## wyager

Quote: 





xabu said:


> > *No and it doesn't need to, if one accepted that the gain is only 6 dB per bit, so this would be 48 dB more. E.g. at 1 kHz you have a doubling of sensed volume per 10 dB increase (which translates to 10 phon for every frequency)*
> > *But it also doesn't had to be 4 - 5 times louder because other aspects like differnt mastering might apply ... and perhaps there is still something wrong with the whole notion regarding the amped volume.*
> >
> > OK, so does it sound anywhere near 48dB louder? Since decibels are a logarithmic unit, I believe that would be roughly 63,000 times louder. (even after the logarithmic compensation of your ear in the inverse direction, it would still be many times louder). That still seems ridiculous to me...
> > ...


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## xabu

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wyager said:


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## D. Lundberg

Quote: 





wyager said:


> Quote:
> 
> 
> 
> ...


 
  0dBFS is the highest possible peak for a digital signal, regardless of bit depth. And at the other end you have the noise floor, which works as a (gradual) limit.
  So a 16 bit signal can be just as loud as a 24 bit signal, the difference is in the dynamic range.
   
  Quote: 





> I'm not quite sure I get this... Yes, increasing the bit depth would make all sounds sharper and clearer.


 
  Higher bit depth will just lower the noise floor, so it won't make a difference unless the noise floor is high enough to be audible and/or mask parts of the signal.
  There are no "steps" in a properly reconstructed signal, just a band limited (and relatively accurate) version of the original signal and some background noise.


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## wyager

[size=medium]

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d. lundberg said:


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  ​[/size]

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xabu said:


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## xabu

Well, my main interest is in unerstanding how it actually works in today applications and systems and not how it could be realized differently.
  The thread was not about how 24 bits could be used in another way as they are used in current systems, but that with the way they are used there is not that kind of difference in sound proclaimed by many.
  And if your idea of how it should be done would lead to an overall better end result is not yet proven.


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## wyager

It's really proven IMO... dynamic range stays the same, overall range stays the same... Unless you think that lack of volume is a good thing, I fail to see how it could possibly be any better to adopt a system where less bits=less volume.


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## xabu

Quote: 





wyager said:


> It's really proven IMO... dynamic range stays the same, overall range stays the same... Unless you think that lack of volume is a good thing, I fail to see how it could possibly be any better to adopt a system where less bits=less volume.


 

 I meant proven by applying it in a realworld system, not theorethical. Build such a system and if it sounds much better than existing ones you will find your market.


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## wyager

What I'm saying is that I think "real world" systems DO work the way I've described... so far, none of my testing has suggested that the way you seem to be saying these things work IS the way they work... In fact, looking at sites where people actually make sound and don't just listen to it (like stackoverflow) I see nothing to suggest that there is a volume difference between 16 and 8 bit. So again, what I'm saying isn't theoretical, I believe it's factual and I haven't heard any convincing evidence otherwise.


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## xabu

Well, seems we have a problem here Houston 
	

	
	
		
		

		
		
	


	



   
   
  It's not aboute volume it's about dynamic range. Whatever, we go in circles ... or nowhere (by the way it's interesting that the word "nowhere" can be fragmented as "now here" which gives it a complete new sense, by itself and in relation to the original word)
   
  So apart from our understanding or thinking or proving, it works somehow


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## mongolianfly

Just a minor correction, 144db + a 50db noise floor does not equal 180db. That's like saying a few people talking at 60db each is as loud as a lawn mower.
  
  Quote:


gregorio said:


> ...144dB + say 50dB for the room's noise floor. But 180dB is the figure often quoted for sound pressure levels powerful enough to kill and some people have been killed by 160dB.


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## D. Lundberg

Quote: 





wyager said:


> *Increasing the dynamic range also decreases the difference between steps... *


 
   
  Audio waveforms have four characteristics: frequency, amplitude, phase (relative to other waveforms) and dynamic range. The dynamic range is the _only_ one that is affected by the bit depth.
  Increasing the amount of quantization steps means less quantization error. Less quantization error means less quantization noise. Less quantization noise means lower noise floor. And lower noise floor means greater dynamic range.
   
_"If Q is the number of quantization steps and D is the dynamic range then_
   
_Q = 2^(D/6). or D = 6 Log2 Q"_[1]
   
  And the noise floor can be shaped to suit your needs. That's why modern D/A and A/D-converters can work with just a few bits and still have >100dB of dynamic range 20Hz-20kHz.
   
  Quote: 





> *and as I said, I have, in experimentation, listened to an 8 bit audio signal and even "properly reconstructed" the voltage jumps were noticeable.*


 
  You won't hear any voltage jumps. If the signal is properly reconstructed (and properly dithered) you'll get the original signal and some amount of white background noise that may or may not be audible (depending on SPL, room, noise shaping, etc.) and/or mask parts of the signal.
  And if it isn't properly reconstructed (truncation, bad or no reconstruction filter(s), etc.) you'll get a signal that is (to some degree) distorted, and that is probably what you heard.
   
  [1] Aldrich, Nika. _Digital Audio Explained: For The Audio Engineer_, 2005, Sweetwater.


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## D. Lundberg

Quote: 





mongolianfly said:


> Just a minor correction, 144db + a 50db noise floor does not equal 180db. That's like saying a few people talking at 60db each is as loud as a lawn mower.


 
  I assume he meant that you'll need to play at a level exceeding 180dB *SPL* to hear every part of a signal with 144dB of *dynamic range* in a room with 50dB *SPL* background noise, and that is accurate.


----------



## khaos974

Are you sure about that? If I generate a -10 dB pink noise file and add a -15 dB 440 Hz sin wave to that file, won't I be able to hear the 440 Hz signal?
   
  Quote: 





d. lundberg said:


> I assume he meant that you'll need to play at a level exceeding 180dB *SPL* to hear every part of a signal with 144dB of *dynamic range* in a room with 50dB *SPL* background noise, and that is accurate.


----------



## wyager

@D. Lundberg
  The entire first part of your post I agree with... You act like you are correcting me, but that's all that I am saying... Those equations you posted describe the relationship I am talking about, if D goes up then Q goes up. If Q goes up, the difference between steps becomes smaller.
   
  The second part, about the signal being "properly reconstructed", I'm not so sure... Even if a "proper" signal is dithered, it still won't sound as good as the original. The amount of noise is going to be proportional to the quantization error you hope to replace, which is a lot at 8 bits.


----------



## khaos974

No one is saying than an 8 bit signal sounds as good as a 16 bit one, it's just that if it is properly reconstructed, there should be no jumps.
   
  I remember a CD track dithered down to 2 bit once, the noise was as loud as the song itself, but there was some "jumping around", but it was far less drastic than I expected with a file with 4 possible values...
  
  A visual example of dithering, in order, the original file (16.7 million colors possible), a 16 colors with withering, a 16 colors with no dithering.


----------



## wyager

Wait-in the second file, the cloud is still visible but in the non-dithered file it is not. If the second file was truly a dithered version of the third file, there would be no cloud, right? Are you sure you didn't dither first and compress second?


----------



## khaos974

To answer your question, all files were saved with the same jpeg compression, and I checked that they were visually very close to the uncompressed version in my photo editor.

 That's the whole point of dithering actually, the whole range of white to blue values is not present in a 16 color file, the dithered file compensated this by alternating too white and too blue pixels to create a blueish white cloud, the undithered version simply truncates the color values to the nearest color.
   
  And there are even multiple algorithms of dithering that 'decide" how to reduce bit depth the best possible way. The DSD coding of SACD has a 1 bit depth with some advanced dithering and noise shaping.


----------



## wyager

So then it was dithered before compression, not after, right? I believe audio dithering is different, the wave is dithered during decompression. Otherwise all you get is quantization error with extra noise.


----------



## Canuckabroad81

absolutely! 
  I for one agree with your research. Sticking recordings on a 24bit setting as apposed to 16bit doesn't accomplish much unless you are mixing.
  I will agree that once sound goes through a compressor or external equipment we start to degrade the audio quality, so the extra bit depth might help reduce the noise added to the audio waveform (transistor noise, RF transients...) such is the case with tube amps which warms up the audio via distortion.  The nice kind of distortion!  
  I for one find digital *almost* all Solid state amps to be lacking in depth of sound, as if something appears to be missing. I guess that's what you Audiophiles call COLD sound? Am I right? 
   
   
   Sorry, I am a newbie.


----------



## khaos974

Quote: 





wyager said:


> So then it was dithered before compression, not after, right? I believe audio dithering is different, the wave is dithered during decompression. Otherwise all you get is quantization error with extra noise.


 
   
  I don't know what compression you are talking about, for the picture above, it was color reduced/dithered from the original uncompressed file, and then jpeg compressed.


  
  Quote: 





canuckabroad81 said:


> absolutely!
> I for one agree with your research. Sticking recordings on a 24bit setting as apposed to 16bit doesn't accomplish much unless you are mixing.
> I will agree that once sound goes through a compressor or external equipment we start to degrade the audio quality, so the extra bit depth might help reduce the noise added to the audio waveform (transistor noise, RF transients...) such is the case with tube amps which warms up the audio via distortion.  The nice kind of distortion!
> I for one find digital *almost* all Solid state amps to be lacking in depth of sound, as if something appears to be missing. I guess that's what you Audiophiles call COLD sound? Am I right?
> ...


 
   
   
  For modern audio, the signal is usually recorded at 24 bit undithered, since no A/D converter achieves real 24 bit depth due to thermal noise. The tracks are mixed/mastered inside a DAW is 32/64 bit precision. The output file is disthered down to 16 bit with some noise shaping.


----------



## wyager

Quote: 





khaos974 said:


> Quote:
> 
> 
> 
> ...


 
  Think about it. If the third picture was dithered, there would be no cloud. If the first picture was dithered and THEN subjected to the same compression as the third picture, the cloud would be visible (and it is).
 It's just that I don't think you really want that when reconstructing audio.


----------



## khaos974

The first pictured is "high resolution", the second (with the cloud) dithered down to 16 colors, the third (no cloud) is truncated down to 16 colors.
  The jpeg compression is invisible in all pictures.
   
  It a photo that I took myself, the whole processing chain is known to me.


----------



## wyager

Yes, I got that... I'm just saying that what you call "dithering down" appears to be dithering then truncating versus truncating then dithering. I believe audio, however, is often truncated then dithered (dithering occurring at the time of playback).


----------



## khaos974

Hmmm, I actually don't know whether it's truncate then inject noise or inject noise then truncate, the former makes more sense to me, my understanding is that the term dither refers to both steps.
   
  But the key point is also that dithering (truncate + inject noise) happens at the mastering stage, ie. the file on the CD/computer is already dithered, there is no dithering at the time of playback. The only situation where you dither at the time of playback id if you need to convert a 24bit file to 16 bit because your DAC doesn't support 24 bit.


----------



## Canuckabroad81

Thanks for explaining this VISUALLY! 
  I got your point easily! 
   
  Again keep the explanations coming as this is all food for the brain. 
   
       
   --- Let's go back to the tube amplifier.  I have a tube amp that is getting HQ .flac 96Khz audio from my ipod doc "unmodified" and unfiltered through the IPOD doc. The audio in fed via component into the inputs. There is a considerable change in audio quality. It sounds fuller to the ear. This is distortion of the waveform or injected noise or both?
   
   Help me here.  
   
   I think I am understanding this. Just clarify please.


----------



## khaos974

Look up tubes harmonics second order on Google, tube warmth is often attibuted to second order harmonic distortion.


----------



## D. Lundberg

Quote:


khaos974 said:


> Are you sure about that? If I generate a -10 dB pink noise file and add a -15 dB 440 Hz sin wave to that file, won't I be able to hear the 440 Hz signal?


 
  You can hear through a noise floor, but as the signal goes below it will be masked to some extent. At som point (depending on the signal, type of noise, etc.) it will be completely masked by the noise and you won't be able to hear it.
  Ideally you want to keep all of the signal above the noise floor.
   
   Quote:


wyager said:


> @D. Lundberg
> The entire first part of your post I agree with... You act like you are correcting me, but that's all that I am saying... Those equations you posted describe the relationship I am talking about, if D goes up then Q goes up. If Q goes up, the difference between steps becomes smaller.


 
  True, I was just clarifying. So then we agree that the bit depth only affects the dynamic range and not the "sound quality" (unless the noise floor is audible or masks the signal)?
   
  Quote: 





> Even if a "proper" signal is dithered, it still won't sound as good as the original.


 
  Except for the amount of background noise, the signal will be (close to) identical. And if you use some form of noise shaping you can minimize the effect of the noise, but you'll probably need to extend the bandwidth a bit as well to make room for the noise.
  If you lower the noise floor in the audible range (20Hz-20kHz) enough there will be no audible difference between a 24bit recording and the same recording reduced to 8bits (or less)!


----------



## wyager

Alright, how about this: 'Depending on the situation, bit depth may or may not affect audible signal quality'? I think that's appropriately ambiguous... 
	

	
	
		
		

		
			




   

 Quote: 





khaos974 said:


> . the file on the CD/computer is already dithered, there is no dithering at the time of playback. The only situation where you dither at the time of playback id if you need to convert a 24bit file to 16 bit because your DAC doesn't support 24 bit.






   
  [size=medium] 
That makes no sense to me... why mess up the waveform if you just get the exact same quantization error plus noise? If it's going to be 16 bit anyway, why mess up the wave before you compress it?​[/size]


----------



## wyager

Oops, double post.


----------



## Canuckabroad81

Quote: 





khaos974 said:


> Look up tubes harmonics second order on Google, tube warmth is often attibuted to second order harmonic distortion.


 


  looked it up. I understand what is going on now. I was correct in my thinking. 
  The tube adds "a harmonic/noise" to the sound/music and our brains interpret this as the sound opening up or warming up.


----------



## D. Lundberg

Quote:  





> Originally Posted by *wyager* /img/forum/go_quote.gif
> That makes no sense to me... why mess up the waveform if you just get the exact same quantization error plus noise?


 
  Dither doesn't "mess up" the waveform, in fact it is is used to prevent quantization (rounding to "steps") from messing up the waveform. The basic form of dither is random noise and it is used to make sure there is enough noise in the waveform to randomize the errors from quantization.
   
  Random quantization errors -> white noise
  Quantization errors that are correlated to the signal -> distortion
   
  Adding dither does actually raise the noise floor by a small amount (for example 0.5bits of dither will raise it by 3dB), but that is a small price to pay to preserve the waveform.
   
  The basic form of dither is used during many stages of processing, but the final quantization from 24bit to 16bit is often done with some form of "colored" (not random) dither and/or noise shaping (which allows you to preserve much of the 24bit noise floor).
   
  Here's a sample of what noise shaping can do:
   
  Silent part (noise floor) of the original master: 24bit/192kHz





   
  Reduced to 16bit/192kHz with iZotope MBIT+ Noise shaping





   
      Quote:  





> If it's going to be 16 bit anyway, why mess up the wave before you compress it?


 
  I'm not sure what you mean. Changing the bit depth is not compression.​


----------



## xnor

@Canuckabroad81: Of course, imagine a pure 100 Hz sine wave vs. a 100+200 Hz harmonic sine wave - the pitch will be higher.
   
  @wyager: I've even posted dithered example files.. see post #539. And I'm still recommending you to read a book about the topic if you want to understand what, why and how things work.


----------



## khaos974

Going back to the example with the pictures, a 4 bit (16 colors) quantization undithered has the pixels values simply truncated or rounded of from the actual values, it creates digital artifacts, see how it simply divides the sky into 2 distinct parts, on the other hand the dithered image has some kind of randomization going on in the sky and globally gives an artifact free and more realistic sky.
   
  Quote:


wyager said:


> > That makes no sense to me... why mess up the waveform if you just get the exact same quantization error plus noise? If it's going to be 16 bit anyway, why mess up the wave before you compress it?​


----------



## xabu

I just got an idea how it is better possible to picture why bit depth got nothing to do with wave resolution.
 Imagine a sine at 10 kHz. Imagine it sampled with 44,1 kHz.
  Picture 1/10000 second (which shows just one pass of the sine through max and min).
 There are only around four values taken for this pass. The sine must look really ugly now, with only four steps.
 Regardless how big the bit depth, even with 100 bit there are still only four values taken.
 But e.g. with 24 bit compared to 16 bit you will be able to take greater max and smaller min values, which would be the greater dynamic range as far as I understood.

 Correct me somebody if I'm wrong, please.


----------



## xnor

Quote: 





xabu said:


> I just got an idea how it is better possible to picture why bit depth got nothing to do with wave resolution.
> Imagine a sine at 10 kHz. Imagine it sampled with 44,1 kHz.
> Picture 1/10000 second (which shows just one pass of the sine through max and min).
> There are only around four values taken for this pass. The sine must look really ugly now, with only four steps.
> ...


 

 The sine doesn't look ugly at all, it looks fine (READ!). What looks ugly is if you link the 4 sample values with a straight line, but that has nothing to do with how the waveform is reconstructed.
   
  There's not really greater max values.
  128 in 8-bit format = 32768 in 16-bit format = 8388608 in 24-bit format = 1.0 normalized (floating point) = 0 dBFS
  (all integers signed)
   
  The problem is quantization noise. Imagine a sample value (normalized) of 0.333, even in a 24-bit format you cannot exactly represent that number, because it would have to be 2793406.5 which is a problem with integers. So you introduce a small amount of noise to the signal because you either have to pick 2793406 or 2793407.
   
  Both values are very close to what we want, but if you repeat quantization with 8-bit numbers you'll see that the error is greater. More noise. Less dynamic range.
   
  Convert the files I uploaded from 16 to 8 bit, with dithering. All you'll see is that the noise floor (around -120 dB) will move up by quite some dB. Everything below that new noise floor will probably be drowned = less dynamic range.


----------



## wyager

@xnor, if I understand that wiki right, the sin wave is shifted so that either the peak values or the (close to) zero values (I'm not sure which) occur at the moment of sampling. Do I understand it correctly?
  Also, please answer me this please-let's say that you have the hypothetical 8-bit +/-1.28v system I described earlier, with .01v resolution. You want a voltage of .635v, but do not have the resolution. Does the noise introduced by dithering, in effect, make the DAC rapidly switch between .63v and .64v to achieve the illusion of .635v for a certain amount of time? This is how I have come to interpret it, but I am not at all sure if this is correct.
   
  @xabu
  while that may be right (But maybe xnor disagreed? And I have no idea.), the problem is that there are plenty of waves well below 10khz. As I said, a non-dithered 100hz 8bit wave sounded really grainy.


----------



## xnor

Quote: 





wyager said:


> @xnor, if I understand that wiki right, the sin wave is shifted so that either the peak values or the (close to) zero values (I'm not sure which) occur at the moment of sampling. Do I understand it correctly?
> Also, please answer me this please-let's say that you have the hypothetical 8-bit +/-1.28v system I described earlier, with .01v resolution. You want a voltage of .635v, but do not have the resolution. Does the noise introduced by dithering, in effect, make the DAC rapidly switch between .63v and .64v to achieve the illusion of .635v for a certain amount of time? This is how I have come to interpret it, but I am not at all sure if this is correct.


 
   

  I really don't know why it's so hard to understand, but to help you I made this little graph:
   

   
  That's a 10 kHz sine wave, sampling rate is 44100 Hz.
   
  We start by looking at each sample, center the sinc function around that sample and scale it down to the sample value. In the end we sum those functions up, et voilà we have our continuous sine wave back.
   
  This step is the same for 8, 16, 24 or whatever bit sample size, the waveform always is going to be continuous like this.
   
  A DAC will try to follow this continuous waveform as close as possible, how the DAC achieves this, however, is implementation specific. (there's a lot more to read 
	

	
	
		
		

		
		
	


	




)


----------



## wyager

OK, so the sum of all the sinc functions from the samples taken is the current amplitude of the sine wave? I guess that makes sense.
   
  However, as xabu pointed out, won't the wave not look smooth, like in your graph, but rather blocky since there are only around four total samples per hertz?


----------



## xnor

Quote: 





wyager said:


> However, as xabu pointed out, won't the wave not look smooth, like in your graph, but rather blocky since there are only around four total samples per hertz?


 

 No, I have pointed that out in my answer clearly, haven't I?
   

   
  This has nothing to do with the reconstructed waveform (see above, I used the same sample values for this graph, it looks nothing like the waveform). You cannot graphically connect samples and hope you get an accurate waveform, that's nonsense.
   
  But either of the display modes shown in my graph are very simple to implement and fast, that's why some audio editors choose to display waveforms that way.
   
   
  (Only some ancient or very simple DACs would output something exactly like the hold line above.)


----------



## xabu

Quote: 





xnor said:


> Quote:
> 
> 
> 
> ...


 
   
  Cool down , read slowly and assume positively . You're "fighting" (mind the quotes) the wrong one. You missunderstood ...
  The first part of your second sentence states exactly what I wrote and what I meant.
  This was a reply to this steps thing graph of wyager and my doubts about it, not my support ...
  I wanted to show/picture that the bit depth got nothing to do with the reconstruction of the wave because e.g  if it would determine the reconstruction of a 10 kHz sine it would sound gruesome, *but it doesn't*, so his assumption has to be wrong !
  I never said that the resulting wave looked bad after all processing is done.
   
  But as I said before ... sometimes it's difficult not to be misunderstood ...
   
   
  The rest of your post I appreciate a lot, because the picture gets more clear for me with that explanation. I almost have this bit depth, noise, dynamic range thing clear now 
  (This wave reconstruction thing I have clear ... otherwise I wouldn't have posted this link: http://www.lavryengineering.com/documents/Sampling_Theory.pdf)


----------



## xnor

I'm not fighting, my response was meant to be sober, illustrative, explanatory.
   
  I don't think I misunderstood you. I wanted to point out that, since you mentioned it "looks ugly", that thinking of it that way is not the right approach and will just cause confusion. (no matter where this idea came from)
   
  Hope I could be of some help.


----------



## xabu

Quote: 





xnor said:


> I'm not fighting, my response was meant to be sober, illustrative, explanatory.
> 
> I don't think I misunderstood you. I wanted to point out that, since you mentioned it "looks ugly", that thinking of it that way is not the right approach and will just cause confusion.
> 
> Hope I could be of some help.


 


 You could, as I mentioned 
  But your other remark unfortunately was just confusing wyager ... and you did misunderstood, as I tried to explain.
  My example was meant to illustrate the wrongness of wyagers assumptions, therefore I used a language based on his vocabulary ... so  this "looks ugly" was meant to be ironic, sarcastic ...whatever.
  And *I* know that there are no steps in the reconstructed wave ... I thought that should be clear regarding my prior replies to wyager.


----------



## wyager

xabu, please don't be passive-agressive and claim that you were asking that question for my benefit. You were just as wrong (if not more so) than I was.
   
  xnor, thank you for the explanation. I think I get it now. Your graphs have been indispensable.


----------



## xabu

Quote: 





wyager said:


> xabu, please don't be passive-agressive and claim that you were asking that question for my benefit. You were just as wrong (if not more so) than I was.
> 
> xnor, thank you for the explanation. I think I get it now. Your graphs have been indispensable.


 
   
  O.k. it's futile and your remark is ridiculous.
  I was not asking for your benefit, but because I suddenly had an idea how to make it clear in layman terms that your steps idea is wrong.
  Next time I post something I will accompany it with a big explanation how it is meant so everybody get's it 
	

	
	
		
		

		
		
	


	




.
  (Perhaps some do already, reading all the posts in order)
   
  Oh my.


----------



## emmodad

surveying the recent pages in this thread, one is reminded of an observation by Bob Katz... (paraphrasing) posting many of those messages, while perhaps having not killed many trees, has certainly inconvenienced a lot of electrons.
   
   
  for those who would like an easily-digestible but accurate overview of sampling theory, quantization, sources and impact of errors etc, digital signal processing and digital audio systems engineering; the most-recent sixth edition of Ken Pohlman's "Principles of Digital Audio" is currently available from amazon for just over $31.00 (shipped US).
   
   
  highly-recommended for anyone interested in a factually-correct reference.  chapter 2 alone is worth the price of admission, and could prove particularly valuable for some recent prolific posters who appear to be running on assumptions, analogies, loose syllogisms and incomplete information.
   
   
  hth
  chuck
   
  edit: formatting


----------



## grawk

thanks for that, it's on my amazon wishlist now


----------



## AndresVilla

interseting post, i like it


----------



## xabu

Thanx for that link. Ordered. Surprisingly (or not) it cost less ordering via amazon.com compared to ordering via amazon.de ....


----------



## wyager

Quote: 





grawk said:


> thanks for that, it's on my amazon wishlist now


 


  Me too.


----------



## kwkarth

Food for thought...
   
  If the author of that book, Ken Pohlman's "Principles of Digital Audio", posted here in this Head-Fi thread, his postings would likely be questioned, denigrated, and otherwise covered over by hearsay and innuendo.  
   
  You know what they say, innuendo and out the other...


----------



## wnmnkh

Wow at this thread has been strong for nearly two years...


----------



## spagetka

Really nice piece of ....
   
  I did a blind test where I had *9s *of
   
   

[*]   
   
   

 mp3 320
 16bit
 24bit
   
  of the same piece of music added in the play-list and played randomly. During the playback, I made some tea/coffee because I want to be sure that I do not know by any chance which piece is playing.
   
  As you mentioned before I was not able to here the difference in amount of detail. BUT I can hear the change of dynamic. Just keep in mind that I had only 9+9 seconds to figure out which  one was better (!!! ones again I was not able to tell if were playing 320/24 or 16/24 or 320/16 !!!!).
   
  5 out of 5 was the result.
   
  nb->usb->DAC1->K701
   
  Quote: 





gregorio said:


> It seems to me that there is a lot of misunderstanding regarding what bit depth is and how it works in digital audio. This misunderstanding exists not only in the consumer and audiophile worlds but also in some education establishments and even some professionals. This misunderstanding comes from supposition of how digital audio works rather than how it actually works. It's easy to see in a photograph the difference between a low bit depth image and one with a higher bit depth, so it's logical to suppose that higher bit depths in audio also means better quality. This supposition is further enforced by the fact that the term 'resolution' is often applied to bit depth and obviously more resolution means higher quality. So 24bit is Hi-Rez audio and 24bit contains more data, therefore higher resolution and better quality. All completely logical supposition but I'm afraid this supposition is not entirely in line with the actual facts of how digital audio works. I'll try to explain:
> 
> When recording, an Analogue to Digital Converter (ADC) reads the incoming analogue waveform and measures it so many times a second (1*). In the case of CD there are 44,100 measurements made per second (the sampling frequency). These measurements are stored in the digital domain in the form of computer bits. The more bits we use, the more accurately we can measure the analogue waveform. This is because each bit can only store two values (0 or 1), to get more values we do the same with bits as we do in normal counting. IE. Once we get to 9, we have to add another column (the tens column) and we can keep adding columns add infinitum for 100s, 1000s, 10000s, etc. The exact same is true for bits but because we only have two values per bit (rather than 10) we need more columns, each column (or additional bit) doubles the number of vaules we have available. IE. 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024 .... If these numbers appear a little familiar it is because all computer technology is based on bits so these numbers crop up all over the place. In the case of 16bit we have roughly 65,000 different values available. The problem is that an analogue waveform is constantly varying. No matter how many times a second we measure the waveform or how many bits we use to store the measurement, there are always going to be errors. These errors in quantifying the value of a constantly changing waveform are called quantisation errors. Quantisation errors are bad, they cause distortion in the waveform when we convert back to analogue and listen to it.
> 
> ...


----------



## grawk

There's a high likelihood that the change in dynamics you hear is an artifact of the conversion process you used...


----------



## spagetka

I do not think so...
   
  It can be true only if they go from 16->24 or mp3(320)->16/24.. Even if it is bad recording, I should not be able to tell the difference, should I?
   
  Quote:


grawk said:


> There's a high likelihood that the change in dynamics you hear is an artifact of the conversion process you used...


----------



## ramicio

[size=medium]Umm I don't see how you can do a test with bit depth with mp3.  Makes no sense to me...  It's still all getting dithered down to 320 kbps.​[/size]


----------



## grawk

16 bits should give you 96dB of dynamic range.  There's no chance your recordings are using that.  If you're playing things loud enough to HEAR 96dB of dynamic range, then you'd be deaf VERY quickly.


----------



## spagetka

this is very funny
   
  If somebody did not pass the blind test, the result is wrong.
  If somebody pass the blind test, the result is wrong.
   
  I took the samples from http://www.linnrecords.com/linn-downloads-testfiles.aspx
   
  @ranicio: I did comparision between mp3 (320) vs 16bit vs 24 bit


----------



## dfkt

Did you downsample the 24/96 yourself, with proper tools? Or did you use a readily available "normal"-definition file?
   
  Edit: I see, they provide different formats on that site. I wouldn't trust those - I would definitely make my own conversions from the 24/96 to be 100% sure I get a quality transcode.


----------



## grawk

They have a vested interest in making 24/96 files seem better.


----------



## dfkt

Indeed, that's why I was asking.
   
  I conducted my own tests on that matter, and with a properly downsampled file there is absolutely no audible difference at all to my ears.


----------



## nick_charles

Quote: 





spagetka said:


> this is very funny
> 
> If somebody did not pass the blind test, the result is wrong.
> If somebody pass the blind test, the result is wrong.
> ...


 
   
  The Linn samples are different in more than just codec/bit-depth, I've tested them in Audacity and CEP, for starters the mp3 file is not quite the same length as the other two, in CEP the min levels are lower on the mp3 file by 4 - 5 db as compared to the flac files. also in CEP if you do a FR analysis on the mp3 and the 24 bit flac and compare the levels up to 20K there is an average difference of 4.22db (do a 8K FFT on the mp3 and a 16K on the 24 bit which gives you the same frequency points up to 22K). casually they look the same but they are very different in the audible spectrum.
   
  To make sensible comparisons you must take the reference file and convert yourself either downsampling with dither or encoding to mp3.

  
  In fairness to Linn the mp3 file is a very good encode it maintains high frequencies very well right up to 20K when it falls off the proverbial cliff.
   
  NB statistically you really want to do at least 10 trials, Foobar has a very good ABX plug-in.


----------



## spagetka

If somebody can send me the reference files, I can make a blind test.
   
  I have found Linn as extremely good with lot of beautiful recordings. My DAC1 accept "only" 24/96 via usb... I am using JRiver (WASAPI Event) and I have to say - great! 
   
   
   
  Quote: 





nick_charles said:


> Quote:
> 
> 
> 
> ...


----------



## nick_charles

Quote: 





spagetka said:


> If somebody can send me the reference files, I can make a blind test.
> 
> I have found Linn as extremely good with lot of beautiful recordings. My DAC1 accept "only" 24/96 via usb... I am using JRiver (WASAPI Event) and I have to say - great!


 

 http://www.divshare.com/download/13844696-99c]DivShare File - recit24bit.wav
  

 http://www.divshare.com/download/13844697-e47]DivShare File - recit16bit-converted-to-mp3-then-back-to.wav
  

 http://www.divshare.com/download/13844698-a98]DivShare File - recit16bit.wav
  
   
  Above are links to 24 bit file, 16 bit file (downsampled) and 320K mp3 file all are in wav containers. Load any two into the FooBar ABX comparator and run 10 trials or more.


----------



## monterto

Fantastic write up!
   
  I have a question that I'm a bit hung up about. This quote,
   
  "*In essence the theorem shows that an analog signal that has been sampled can be perfectly reconstructed from the samples*."
   
  Assuming I am not mis-understanding this then there are no advantages to listenning to vinyl vs CD?


----------



## khaos974

Could you post them on mediafire? I can't access them from China.
  
  Quote: 





> Originally Posted by *nick_charles* /img/forum/go_quote.gif
> 
> http://www.divshare.com/download/13844696-99c]DivShare File - recit24bit.wav
> http://www.divshare.com/download/13844697-e47]DivShare File - recit16bit-converted-to-mp3-then-back-to.wav
> ...


 
   
   
  If your goal is fidelity to the master, there is no advantage to listening to vinyl instead of CDs.
  Baring that, the questions of catalog content, different masterings, cost, emotional attachment, rituals... But if the question is purely about the accuracy to the recorded material, CDs beat vinyl by a fairly large margin.
  Yes, the content can be theoretically perfectly reconstructed fro the samples up to 22.5 kHz.
   
  Quote: 





> Originally Posted by *monterto* /img/forum/go_quote.gif
> 
> "*In essence the theorem shows that an analog signal that has been sampled can be perfectly reconstructed from the samples*."
> 
> Assuming I am not mis-understanding this then there are no advantages to listenning to vinyl vs CD?


 
  .


----------



## ramicio

I just love how people think because something is supposedly proven mathematically, on paper, that it means it will work in real life.  Good luck with your theorems applying to physical electrons.


----------



## khaos974

I love how you _interpret _my words, I never said that reconstruction filters were perfect, did I?


----------



## spagetka

Thank you very much. The files were downloaded. I'll do the test as soon as possible.
   
   
  BTW: 

 How can I verify that the files are not the same (only the name has changed)? 
 Is there ABX plug-in available for Jriver Media center?
   
   
  Thank you.
   
   
  Quote:


nick_charles said:


> Quote:
> 
> 
> 
> ...


----------



## monterto

Quote: 





khaos974 said:


> If your goal is fidelity to the master, there is no advantage to listening to vinyl instead of CDs.  Baring that, the questions of catalog content, different masterings, cost, emotional attachment, rituals... But if the question is purely about the accuracy to the recorded material, CDs beat vinyl by a fairly large margin.
> Yes, the content can be theoretically perfectly reconstructed fro the samples up to 22.5 kHz.
> .


 

 This is very interesting, thank you for the insight. I suppose we all need to be more careful of the assumptions we make.


----------



## nick_charles

Quote: 





spagetka said:


> Thank you very much. The files were downloaded. I'll do the test as soon as possible.
> 
> 
> BTW:
> ...


----------



## nick_charles

Quote: 





khaos974 said:


> Could you post them on mediafire? I can't access them from China.
> 
> 
> http://www.mediafire.com/?0n1hnkegv3z7ahv
> ...


----------



## khaos974

Thanks.
   
  foo_abx 1.3.4 report
 foobar2000 v1.1.2
 2011/01/24 02:45:42

 File A: C:\Users\Khaos\Desktop\recit24bit.wav
 File B: C:\Users\Khaos\Desktop\recit16bit.wav

 02:45:42 : Test started.
 02:46:50 : 00/01  100.0%
 02:46:56 : 01/02  75.0%
 02:46:59 : 02/03  50.0%
 02:47:00 : 03/04  31.3%
 02:47:02 : 04/05  18.8%
 02:47:04 : 05/06  10.9%
 02:47:05 : 06/07  6.3%
 02:47:08 : 06/08  14.5%
 02:47:11 : 07/09  9.0%
 02:47:13 : 07/10  17.2%
 02:47:26 : 07/11  27.4%
 02:47:27 : 08/12  19.4%
 02:47:28 : 08/13  29.1%
 02:47:29 : 08/14  39.5%
 02:47:30 : 08/15  50.0%
 02:47:31 : 09/16  40.2%
 02:47:32 : 10/17  31.5%
 02:47:41 : Trial reset.
 02:47:42 : 00/01  100.0%
 02:47:43 : 00/02  100.0%
 02:47:44 : 00/03  100.0%
 02:47:46 : 01/04  93.8%
 02:47:47 : 02/05  81.3%
 02:47:48 : 03/06  65.6%
 02:47:49 : 04/07  50.0%
 02:47:50 : 05/08  36.3%
 02:47:51 : 06/09  25.4%
 02:47:53 : 07/10  17.2%
 02:47:55 : 08/11  11.3%
 02:47:55 : 08/12  19.4%
 02:47:56 : 09/13  13.3%
 02:47:57 : 09/14  21.2%
 02:47:58 : 10/15  15.1%
 02:47:59 : 10/16  22.7%
 02:48:00 : 11/17  16.6%
 02:48:01 : 11/18  24.0%
 02:48:02 : 12/19  18.0%
 02:48:03 : 12/20  25.2%
 02:48:04 : 13/21  19.2%
 02:48:07 : Trial reset.
 02:48:10 : 00/01  100.0%
 02:48:11 : 01/02  75.0%
 02:48:12 : 02/03  50.0%
 02:48:14 : 03/04  31.3%
 02:48:14 : 03/05  50.0%
 02:48:15 : 03/06  65.6%
 02:48:16 : 03/07  77.3%
 02:48:18 : 03/08  85.5%
 02:48:19 : 04/09  74.6%
 02:48:21 : 04/10  82.8%
 02:48:22 : 04/11  88.7%
 02:48:24 : Trial reset.
 02:49:01 : 01/01  50.0%
 02:49:21 : 02/02  25.0%
 02:50:04 : 03/03  12.5%
 02:50:27 : 04/04  6.3%
 02:51:44 : 04/05  18.8%
 02:51:57 : 05/06  10.9%
 02:52:22 : 06/07  6.3%
 02:54:25 : 07/08  3.5%
 02:55:14 : 08/09  2.0%
 02:56:24 : 09/10  1.1%
 02:56:30 : Test finished.

  ----------
 Total: 36/59 (5.9%)
   
  Actually, the very first tests were just my playing with the ABX plugin, check out the last 10 tests,after the last rial reset, 1 error out of 10 trials, 1.1 % chance that I was guessing.


----------



## nick_charles

Quote: 





khaos974 said:


> Thanks.
> 
> foo_abx 1.3.4 report
> foobar2000 v1.1.2
> ...


 

 I'd be inclined to ignore all the previous tests and do another big run from scratch of say 20 trials, otherwise what you are doing is cherry-picking , i.e deciding to use only the best data. In the last tests what did you home in on was one perceptibly louder, shriller, less disrtorted, what cues were there?
   
  For the record I have guessed 7/8 in Foobar always saying X is A. 9/10 does looks solid though


----------



## khaos974

Hmmm what are those figures? dB below full scale, if they are those would be inaudible at standard volume (not to mention that 16 bit goes only to -98 dB). the values of the samples maybe?
   
  Actually, it's more trivial than that, using foobar's replay gain, the 24 bit track is at +2.61 dB while the 16 bit track is at +2.91 dB.
  0.3 dB difference => *distinguishable*.
   
  I didn't use replay gain for the ABX, had I used it, I 'd doubt I'd get the same results.


----------



## nick_charles

Quote: 





khaos974 said:


> Hmmm what are those figures? dB below full scale, if they are those would be inaudible at standard volume (not to mention that 16 bit goes only to -98 dB). the values of the samples maybe?
> 
> Actually, it's more trivial than that, using foobar's replay gain, the 24 bit track is at +2.61 dB while the 16 bit track is at +2.91 dB.
> 0.3 dB difference => *distinguishable*.
> ...


 
   
  Sorry those figures were db below full scale and difference between 16 bit sample and 24 bit sample, but I dissed tham as I am not convinced they are reliable


----------



## khaos974

Actually, before the last series of 10 trials, I was clicking at random without even listening to the samples to see the kind of percentages I'd get.
 I thought that the reset button erased all previous data, it turned out not to be so, so the "playing around" data was kept and included in the final report, all in all, it's just a series of 10 consecutive trials with one mistake, and I don't want to do it again since it's 4 am in Beijing. Good morning, I suppose.

 Notice how without the last series of 10 trials, I get a wooping 27/49 by randomly guessing. (ironical)
  Also, the only thing I proved was that a .3 dB difference is audible, hence the necessity to level match to at least 0.1 dB in ABX, fact which was known long before, no need to do another run of ABX to confirm a known fact. I'm going to sleep now.
   
  zzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzzz
   

  
  Quote: 





nick_charles said:


> I'd be inclined to ignore all the previous tests and do another big run from scratch of say 20 trials, otherwise what you are doing is cherry-picking , i.e deciding to use only the best data. In the last tests what did you home in on was one perceptibly louder, shriller, less disrtorted, what cues were there?
> 
> For the record I have guessed 7/8 in Foobar always saying X is A. 9/10 does looks solid though


----------



## spagetka

Thank you very much. I plan to make that test on Wednesday or Thursday...
  
  Quote: 





nick_charles said:


> Quote:
> 
> 
> 
> ...


----------



## senndroid

Hi All:
   
  I've browsed through this thread, but haven't really found a consensus, and the physics / wave theory is over my head (econ major). Anyway, here's my question (and ultimately, the reason I stopped by).
   

 For Headphone listening, does the bit rate that a given DAC is capable of (16 v. 24) affect the sound quality (detail or dynamics) in any audible way? Assume FLAC is being played.
 As you can see in my sig, I'm running FLAC --> E7/E9 via USB --> HD650. Is the 16 bit-rate in the E7 holding me back here?
   
  I appreciate everyone's eagerness to share knowledge about this subject, sorry I'm a n00b with this.


----------



## monterto

I'm no audio engineer, but I did a fair bit of reading after finding this thread, and I think I have a pretty good understanding of this now. Hopefully someone can correct me if I make a mistake.
   
  Here goes:
  Like the OP stated, 24-bit audio allows for a greater dynamic range than 16 bit. This is irrelevant however, because the dynamic range of 24 bit audio is 144dB which is the equivalent of a gunshot. 16-bit audio has a dynamic range of 96 dB which is also very loud. Dynamic range is the difference between the sound floor and the loudest possible un-distorted sine wave, the OP estimates a sound floor of ~50dB for speakers and ~30dB for headphones, which means that the full dynamic range of a CD will effectively deafen you and the full dynamic range of 24 bit audio will kill you.
   
  The Nyquist-Shannon sampling theorum states that a sound can be accurately reconstructed from samples up to a maximum frequency equal to one half of the sample rate, 44.1khz sampling frequency = 22.05khz highest frequency that can be reconstructed; 96 khz ---> 48khz etc. A child can hear up to 20khz and that range decreases throughout life.
   
  Both of these mean that, in a practical application, CD quality audio (16 bit @ 44.1kHz) is as high as you will ever need for listenning to music. Recording and mastering is a different story and I can't comment on that.
   
  (I'll edit my post if anyone spots a mistake)


----------



## senndroid

Thanks for summing that up for me, montero--that definitely makes sense. Does anyone else agree/disagree with his summary?
 Looks like I will be upgrading amps before I upgrade my DAC


----------



## ILikeMusic

His summary is essentially correct. There are some reasons higher bit rates/depths are used during the recording and mastering process, but for distribution there is little reason to go beyond the 44 kHz/16 bit Redbook standard. The only thing a higher bitrate will give you is a higher upper frequency limit (and 20 kHz is plenty if you happen to be a human being) and the only thing a greater bit depth will give you is a dynamic range way beyond what is practical or what your playback equipment (no matter how good it is) is capable of supporting. The common imperfect understanding of these facts by consumers does make a lot marketing departments happy though.


----------



## spagetka

Here is my test results:
   
*test1:*
  foo_abx 1.3.4 report
 foobar2000 v1.1.2
 2011/01/27 18:38:22
  File A: C:\Users\marian\Desktop\recit16bit.wav
 File B: C:\Users\marian\Desktop\recit24bit.wav
  18:38:22 : Test started.
 18:40:36 : 01/01  50.0%
 18:48:47 : 01/02  75.0%
 18:49:09 : 02/03  50.0%
 18:49:17 : 03/04  31.3%
 18:49:28 : 03/05  50.0%
 18:50:07 : 04/06  34.4%
 18:50:24 : 04/07  50.0%
 18:50:52 : 05/08  36.3%
 18:51:08 : 06/09  25.4%
 18:51:48 : 07/10  17.2%
 18:56:47 : Test finished.
   ----------
 Total: 7/10 (17.2%) 
  ********************************************************** 
*test2:*
  foo_abx 1.3.4 report
 foobar2000 v1.1.2
 2011/01/27 19:06:17
  File A: C:\Users\marian\Desktop\recit16bit.wav
 File B: C:\Users\marian\Desktop\recit24bit.wav
  19:06:17 : Test started.
 19:06:28 : 01/01  50.0%
 19:06:42 : 02/02  25.0%
 19:08:14 : 03/03  12.5%
 19:09:24 : 03/04  31.3%
 19:10:14 : 04/05  18.8%
 19:10:24 : 04/06  34.4%
 19:10:34 : 04/07  50.0%
 19:11:00 : 05/08  36.3%
 19:11:19 : 06/09  25.4%
 19:11:34 : 07/10  17.2%
 19:11:57 : Test finished.
   ----------
 Total: 7/10 (17.2%) 
  **********************************************************
*test3:*
  foo_abx 1.3.4 report
 foobar2000 v1.1.2
 2011/01/27 19:18:52
  File A: C:\Users\marian\Desktop\recit16bit.wav
 File B: C:\Users\marian\Desktop\recit24bit.wav
  19:18:52 : Test started.
 19:19:02 : 01/01  50.0%
 19:19:50 : 02/02  25.0%
 19:20:53 : 03/03  12.5%
 19:22:27 : 04/04  6.3%
 19:23:20 : 04/05  18.8%
 19:23:48 : 05/06  10.9%
 19:24:33 : 06/07  6.3%
 19:24:55 : 07/08  3.5%
 19:25:22 : 08/09  2.0%
 19:26:07 : 09/10  1.1%
 19:26:21 : Test finished.
   ----------
 Total: 9/10 (1.1%) 
   
  Best,
  Marian


----------



## monterto

Thanks for posting those reults, I might do this today myself...


----------



## wavoman

Quote: 





ilikemusic said:


> ...The only thing a higher bitrate will give you is a higher upper frequency limit ...


 
   
  Well not quite true.  The Nyquist theorem does not imply this; some posters here do not understand the theorem fully I'm afraid.
   
  The theorem is an exact result in a theoretical world that is only an approximation to reality.  You see, the Fourier transform of a bandwidth-limited function has infinite support in the time domain.  So unless you believe the song you are listening to began at the dawn of time, and will go on forever, then Nyquist does not apply precisely.
   
  Read http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem#Practical_considerations
   
  It's a really good model of nature, so Nyquist packs a lot of practical wisdom ... but no guarantee that you will get an exact reconstruction from 0 to 22.05 if you sample at 44.1 .  You won't.
   
  There's more to say:  if the original recording A-to-D step is done at 96, which it often is, you will get a better reconstruction of the analog waveform in the D-to-A step if you keep the 96 digital sample.  Down converting to 44.1 will lose information that results in a less-perfect analog output after D-to-A.
   
  The difference may or may not be audible.  I make no claims about that.  But computer audio makes it trivial to store and use music at the same sampling rate it was recorded at, so this is where the world should go, and is: the future is 96 downloads.
   
  I am sure the difference between 16 bits and 24 bits is not audible, but that's not the point.  The point is: don't convert.  Every conversion loses something, making the final D-to-A less accurate.  Again, it may not be audible, but if recording hardware runs at 96/24, then why not run your playback software like that. 
   
  Real-world D-to-A conversion algorithms typically begin with an interpolation step that upsamples dramatically, so that the final modulation resulting in the analog signal is more accurate.  (Not all DACs do this, but most do, and the ones that don't typically sound worse, most people agree -- some disagree, but that's life).  This is the problem with having both the 44.1 / 88.2 / 176.4 world vs the 48 / 96 / 192 world.  Upsampling (and downsampling for storage considerations if you have devices with limited disk) across those two families introduces a lot more error than within those families -- interpolation vs simple decimation (really hexamation) and replication.
   
  Half-sampling a 96 file to store as Apple Lossless (48) sounds better to me than the same track off a redbook CD at 44.1 re-sampled by SOX to Apple Lossless at 48.  So maybe the difference in sampling rates is audible, but the test was not blind so I don't know for sure. Often playing directly the 96 on a computer (instead of 48 on an iPod) doesn't sound different that playing the 44.1 on the same hardware, so who knows.  My belief is the non-congruent conversion is a bigger factor than the increased sampling rate.  BTW, I never hear any difference when I convert 24 bit 96 to 16 bit 96.
   
  Back to the main point.  In the real physical world, a higher sampling rate, if preserved throughout the playback chain right up to the final D-to-A, can indeed lead to more accurate reconstruction of the original analog signal, and NOT just at higher frequencies.  This does not violate the Nyquist theorem.  There is no guarantee of exact reconstruction (in the real world) of frequencies at or below half the sampling rate, although you can get very close.  I have no idea if faster samplng makes an audible difference.  Most published blind tests suggests it does not.  So do my own non-blind tests on myself.  But my own non-blind tests on myself suggest that sample rate conversion does introduce (small, and only sometimes) audible effects, so that 96 recordings should be kept at 96.


----------



## spagetka

wavoman,
  I almost completely agree with you. Almost.
   
  The key thing is, that you transform one problem to another,because you say that IF the result is audible than it is caused by transformation. But you simply do not give an example.
   
  Math is very tricky in these things because it is not explicit process. There were,are and will be the errors in very complex parts because of limits.
   
  The up-sampling process is the black box for me so I do not trust.


----------



## wavoman

I agree -- I made an assumption.  The degraded sound quality I hear could be due to something other than the bit-rate conversion process, you are right.
   
  But I wanted to make a point that still stands -- in the real world, music sampled at 96 can be more accurately reconstructed than music sampled at 44.1, and this not just about the frequencies above the Nyquist folding frequency (half the sampling rate).  This statement is not in conflict with Nyquist's theorem.  Nyquist's mathematical result suggests that in the real world the audible gains from such sampling would be minimal, if they exist at all.  This is why 44.1 was considered acceptable back in the day.
   
  Some claim they can hear a difference, others don't.  However, this is not terribly important once music is commercially available at 96.  Because then we just leave it at 96.  Surely downcoverting it will lead to less accurate re-construction, especially to a non-congruent number like 44.1.  And in my own experiments I can hear the 96-to-44.1 downconverstion loss of fidelity, but I cannot hear the 96-to-48 loss of fidelity.  The test was not blind, so it is not conclusive.  However, I have no reason to convert to anything but 48, and I do that only for the iPod.
   
  I am forced to downconvert to 176.4 or 88.2 to rip SACDs ... that really bugs me.  But downloads and DVD-A's are 96 (some at 192), so life is good.  44.1 is so yesterday.
   
  Over and over, when people listen to Metallica on my iMod their jaws drop open. This would happen even if Claude and Harry listened (that's Shannon and Nyquist).  It's because I ripped the DVD-A with DVD-Explorer, carefully decimated the 96 to 48, and put in on the iMod as ALAC.  It sounds better than the 44.1 rip done with EAC and accurip, then converted from WAV and up-sampled to 48 ALAC by Xrecode II.
   
  The conversion, or the higher sampling rate -- which made the difference?  Don't know and don't care.


----------



## Ham Sandwich

Quote: 





wavoman said:


> The conversion, or the higher sampling rate -- which made the difference?  Don't know and don't care.


 


  Another possibility is that the DVD-A audio and the CD audio are not the same to begin with.  The CD audio version may have gotten some subtle compression or other processing during the process of converting to CD format.


----------



## wavoman

Could be, for sure.  You mean other than the processing to make it 44.1, the engineers might have further downgraded it. 
   
  Most likely true. There's alot of stuff over at the SH forum about how the redbook masterers didn't give a damn, and the hi rez ones did, back then.


----------



## emmodad

Quote: 





wavoman said:


> But I wanted to make a point that still stands -- in the real world, music sampled at 96 can be more accurately reconstructed than music sampled at 44.1
> 
> [ snip ]


 
   
  in the real world, it is also perfectly possible to have a situation where music sampled at 44.1 can be more accurately reconstructed than music sampled at 96.....
   
  (albeit a degenerate case which presumes poor signal processing system design)
   
  just harry, claude et al at work......


----------



## wavoman

Agree 100% ... which is why it's always all about listening with our own ears (and a good set of headphones!).
   
  Most of the time, a redbook CD at 44.1 ripped any old way and lossy compressed to MP3 at 256 sounds just fine, played on a stock iPod with cheap but reasonable headphones (Senn PX-100s, say). 
   
  But then there are tracks for which that's not true, and the 96 DVD-A version ripped carefully and decimated to 48, lossless converted to ALAC, and played on an iMod 5.5 Gen with the Wolfson DAC, thru an iQube using Grado HF-2's will take your breath away -- and maybe the DVD-A version was better mastered, but I find compromising on any part of that chain will degrade the audio quality I experience. 
   
  And finally there's that one-in-a-thousand track that played at full SACD resolution with a world class DAC (Wadia in my case), listening via Orpheus, or O2+BHSE, is life-changing.
   
  So IMHO resolution matters. It doesn't matter much -- in the same sense that you need only air, food, and water to live, you don't need sex, but why would you live without it if you had the option?  So it is with Hi Res.


----------



## ex0du5

Quote: 





wavoman said:


> So IMHO resolution matters. It doesn't matter much -- in the same sense that you need only air, food, and water to live, you don't need sex, but why would you live without it if you had the option?  So it is with Hi Res.


 

 I'll take sex over high resolution audio any day of the week.


----------



## ILikeMusic

Quote: 





wavoman said:


> ...
> The difference may or may not be audible.  I make no claims about that.
> ...


 

  
  But whether the difference is audible or not is the _entire point_. I understand the 'just don't do it' argument in the theoretical sense, but considering the not-insignificant size difference of 96/24 and 44/16 files I don't see why one would devote resources if there is no demonstrable purpose. "I don't know, and I don't care' just doesn't cut it for me as a rationale, but I guess that is just my way of looking at things. Engineering classes will do that to you.


----------



## ex0du5

The size really isn't that big of a deal, is it?
   
  With fully uncompressed files, we're talking about 3x the data (2x sample rate * 1.5x bit depth).
   
  Storage is cheap enough. A $80 2TB HDD will still fit 1000 96/24 discs. That's $0.08 a disc, instead of around $0.03 a disc. Not a very big expense at all.


----------



## OKtave

I know this thread is two years old... 
   
  There's the math and the mathematicians.  There's the music and the musicians.
  There's the audiophile and his expensive stuff, and there's the music lover who knows what to listen for.
   
  Here's a practical approach to the whole thing by someone who's been mastering in analogue and digital.
   
  When we were recording on tape, we knew right away what the difference was between the live and the recorded.  Even in the best studios there was a clear difference between 15ips and 30ips.  Your album would sound entirely differently should you choose an option over the other.  No question about it.
   
  The digital domain made us look for the differences in the same places, but we were looking at the wrong place.  For example, it became clear that the closest you'd work to the limit of the 16bits, you would get loud clipping, with no warning.  So we'd give ourselves some "space" not to go over, therefore loosing a little of the 16bit dynamic range.  From this standpoint, here's my first rule:  24bit allows you to work further from its full dynamic range without loosing quality (I'll get to the "quality" down-to-earth definition later), but stricly from a "practical" point of view.
   
  - One of the first CD's I heard was "Sting - Nothing like the sun".  My first edition CD was lousy at best and I prefered the vinyl version.  Listening to "Be still my beating heart" intro with a fade-in, it was clear in my head what we had to look for to detect "quality" differences in digital masters.  In this very example, it becomes clear that we hear the floor of the track much higher than it should be, letting us hear what poor dithering sound like.   Without that dithering, we would have hear the actual 16bit steps of amplitude one by one until a pivot point where the amplitude of the mix overshadows the artifact itself.
   
  - Testing with recordings made at 16bits VS 24bits, I've discovered that the break points of fade-ins and the dying point of fade-outs are where 24bits recordings really stand out.  There are only a few recordings in which I've noticed that 24bits would have been better and only 1 is of analogue source, i.e. Supertramp - Crime of the century, between "School" and "Bloody Well right".  The combination if a fade-out and a note glissendo created that artificial second "note" that a 24bit would not have, because of the "smaller steps" between the specific amplitude levels of 24bits, much smaller than those of 16bits.  This is subtle, but it is there.  On the vinyl, I cannot hear it, simply because it's not there.  So rule #2:  24bits is better than 16bits at LEAST when we expect to record and reproduce notes, chords, strings, that will either make a up or down glissendo AND an up od down crescendo.
   
  - Today's albums are mastered and compressed so they can be heard loud and clear on the car radio, in your iPod, table top system...  Most sound LOUZY when played on a HI-END system, capable of better nuances and subtleties, because the CD is recorded that way.  20 years later though, when they'll release the "remastered" versions of these recordings, they'll be happy to have chosen (or unhappy if they haven't) to record in 24 bits when they'll decide NOT to make it sound as "punchy", but the 24bit original recordings will let them master the whole thing without any audible compromise on definition.  Rule #3:  Use it now if you have it, you're making choice upwards.
   
   - I've heard re-mastered albums from the 70's when digital wasn't around yet, put on SACD...  They sound good.  They sound JUST AS GOOD when digitally burned on 16bits because the tape resolution and noise floors are way ABOVE those of 16bits/44kHz.  If they "sound" better, it's because they've added EQ, filters and whatnot.  They would have come out just as good on 16bits with the same corrections.  Your original vinyl in perfect shape on a hi-end turntable is even better than those remasters, but such conditions are rare, so very few people can compare.  Rule #4:  Old masters won't sound any better whether they're put on 16bits OR 24bits OR Sacd.  The original quality is just not there yet.  It would be like expecting colors to appear from a B/W movie when copying it on DVD, or Blu-Ray.  When they sound "better" than, say, the MFSL version of it, they've just made different EQ's and filtering decisions.  All of which will sound just as good on any 16-24-Sacd format.
   
  Very few know what to listen for when attempting to compare bit depths.  For instance, a know-it-all seen-il-all put up a site where you could listen to 4 wav's files, recorded at different bit depths and the author asks to guess which one is which.  The problem is, the author revealed his lack of knowledge when he chose the musical passage he did, in which it was impossible to determine that with such little mucical information.  I'm not going to mention his name but for those of you who have search for answers like me, you've heard of the whiner I'm sure.. My point is, you're only going to hear the difference IF you know what you are searching for AND if you KNOW what MUSIC should sound like in the first place.
   
  *I've attempted to be as little technical as possible to reach as many music lovers as possible.


----------



## nick_charles

Quote: 





oktave said:


> - Testing with recordings made at 16bits VS 24bits, I've discovered that the break points of fade-ins and the dying point of fade-outs are where 24bits recordings really stand out.  There are only a few recordings in which I've noticed that 24bits would have been better and only 1 is of analogue source, i.e. Supertramp - Crime of the century, between "School" and "Bloody Well right".  The combination if a fade-out and a note glissendo created that artificial second "note" that a 24bit would not have, because of the "smaller steps" between the specific amplitude levels of 24bits, much smaller than those of 16bits.  This is subtle, but it is there.  On the vinyl, I cannot hear it, simply because it's not there.  So rule #2:  24bits is better than 16bits at LEAST when we expect to record and reproduce notes, chords, strings, that will either make a up or down glissendo AND an up od down crescendo.
> 
> *To date extensive DBTs of high res and redbook standard recordings do not support your assertion, it is possible to hear any difference you like when you know which is which, being able to do it blind is a very different matter, this is where the evidence for the superiority of high res vs redbook is lacking to date. There is not one verifiable set of DBTS to support the audibility of the difference between high res and downgraded redbook for the same source (at normal listening levels).*
> 
> ...


----------



## Arnaldo

*AES E-Library*
*Sampling Rate Discrimination: 44.1 kHz vs. 88.2 kHz*
  (http://www.aes.org/e-lib/browse.cfm?elib=15398)
   
"It is currently common practice for sound engineers to record digital music using high-resolution formats, and then down sample the files to 44.1kHz for commercial release. This study aims at investigating whether listeners can perceive differences between musical files recorded at 44.1kHz and 88.2kHz with the same analog chain and type of AD-converter. Sixteen expert listeners were asked to compare 3 versions (44.1kHz, 88.2kHz and the 88.2kHz version down-sampled to 44.1kHz) of 5 musical excerpts in a blind ABX task. Overall, participants were able to discriminate between files recorded at 88.2kHz and their 44.1kHz down-sampled version. Furthermore, for the orchestral excerpt, they were able to discriminate between files recorded at 88.2kHz and files recorded at 44.1kHz."
   
*Authors:* Pras, Amandine; Guastavino, Catherine
*Affiliation:* McGill University, Montreal, Quebec, Canada
*AES Convention:* 128 (May 2010)
*Paper Number: *8101
*Subject: *Audio Coding and Compression
   
  Quote: 





nick_charles said:


> To date extensive DBTs of high res and redbook standard recordings do not support your assertion, it is possible to hear any difference you like when you know which is which, being able to do it blind is a very different matter, this is where the evidence for the superiority of high res vs redbook is lacking to date. There is not one verifiable set of DBTS to support the audibility of the difference between high res and downgraded redbook for the same source (at normal listening levels).


----------



## nick_charles

Quote: 





arnaldo said:


> *AES E-Library*
> *Sampling Rate Discrimination: 44.1 kHz vs. 88.2 kHz*
> (http://www.aes.org/e-lib/browse.cfm?elib=15398)
> 
> ...


 

  
  I have that paper, the stats are *extremely dubious *, and the authors refuse to release the raw data, it is an interesting paper though, I'd forgotten about that one, there is a lively discussion about it on Hydrogen Audio, C/F however Meyer and Moran who ran 60 subjects through over 500 trials and not one managed a statistically significant result.


----------



## Arnaldo

There was a rather extensive discussion over the methodology and validity of the M&M tests at SA-CD.net (http://sa-cd.net/showthread/42987/42987/ and http://sa-cd.net/showthread/58757/58757/). It seems that M&M unknowingly used mostly low-rez recordings in the tests, thus making the results statistically flawed to say the least. Regardless, I'd rather not get involved into that discussion (again)...
  
  Quote: 





nick_charles said:


> I have that paper, the stats are *extremely dubious *, and the authors refuse to release the raw data, it is an interesting paper though, I'd forgotten about that one, there is a lively discussion about it on Hydrogen Audio, C/F however Meyer and Moran who ran 60 subjects through over 500 trials and not one managed a statistically significant result.


----------



## justanut

And of course it took a panel of "experts" to do it... How many of us claim to be one?


----------



## wingman1659

This is exactly why I don't understand why apple wants 24bit downloads in their store. I would undoubtedly prefer a 48kHz download option. Heck I'd even take 44.1kHz loss-less. Pushing 24bit downloads really doesn't make sense to me.


----------



## Permagrin

wingman1659 said:


> This is exactly why I don't understand why apple wants 24bit downloads in their store. I would undoubtedly prefer a 48kHz download option. Heck I'd even take 44.1kHz loss-less. Pushing 24bit downloads really doesn't make sense to me.




I know the iPod supports 24 bit depth but do most soundcards? If it gets truncated down the line it's going to sound worse than 16 bit. Isn't that right?


----------



## maverickronin

Quote: 





permagrin said:


> I know the iPod supports 24 bit depth but do most soundcards? If it gets truncated down the line it's going to sound worse than 16 bit. Isn't that right?


 


  I've got a nearly decade old soundcard in my PC that I only use for my so-so speakers and it does 24bit/48khz.  Even the onboard soundcard on my netbook does 24/48.


----------



## Griffinhart

If it gets _truncated_ from 24 to 16-bit, then yeah, I'd imagine it'd sound like crap. If it gets downsampled though, then there won't be as huge a (noticeable) quality loss.

 -- Griffinhart


----------



## Sylverant

Quote: 





wingman1659 said:


> This is exactly why I don't understand why apple wants 24bit downloads in their store. I would undoubtedly prefer a 48kHz download option. Heck I'd even take 44.1kHz loss-less. Pushing 24bit downloads really doesn't make sense to me.


 

 Its either all or nothing with apple huh? They can't just convert the entire store to CD quality ALAC; they have to offer the premium experience at high cost alongside a small catalogue? Really now. They shouldn't take advantage of people like that.
   
  I checked some of the info in Gregorio's post and checked it on google and most music really is recorded/mixed at 48 bits, then downsampled to 24 bit for proccessing and then 16 for the CD release. If the dithering process doesn't get messed up and is from the same mix there shouldn't be an audible difference. Just give us 16bit ALAC apple. Have they even seen how hard we try to love them despite our fangs? They don't even know how many people's day they'd make if ALAC became the iTunes standard.
   
  Edit: assuming the RIAA wouldn't make them drm lossless, anyway. Get the feeling that might happen.


----------



## Griffinhart

>ALAC
 >standard
   
  /facepalm

 FLAC. I honestly don't understand why most big-name DAPs (e.g., Apple iDevices, MS Zunes) don't have native support for FLAC. I mean, that "F" in the name? It stands for "Free".
   
  (I mean, it'd make sense if, say, ALAC supported DRM, and Apple used ALAC for that... but it currently doesn't, and they don't.)
   
  -- Griffinhart


----------



## nanaholic

Quote: 





griffinhart said:


> I honestly don't understand why most big-name DAPs (e.g., Apple iDevices, MS Zunes) don't have native support for FLAC. I mean, that "F" in the name? It stands for "Free".


 
   
  You just answered your own question by raising what the F stands for.
  "Free" means no lock-in, which means consumers are also "free" to move between software/hardware, and that is NOT good business for the individual mega corps as their aim is to set up little barriers that makes it inconvenient to switch to other platforms so the consumers would just "stick to what they know" for "the seamless experience".  Sony used to be really good at this game but now Apple is undoubtedly king in this area.


----------



## maverickronin

Quote: 





nanaholic said:


> You just answered your own question by raising what the F stands for.
> "Free" means no lock-in, which means consumers are also "free" to move between software/hardware, and that is NOT good business for the individual mega corps as their aim is to set up little barriers that makes it inconvenient to switch to other platforms so the consumers would just "stick to what they know" for "the seamless experience".  Sony used to be really good at this game but now Apple is undoubtedly king in this area.


 


  Except that they all already support mp3 which doesn't have any of that either...


----------



## Griffinhart

Incidentally, using MP3 would also mean having to pay licensing/patent fees because MP3 _isn't_ a free codec standard. And yet, people still use that, and can use it to move between software/hardware.
   
  Also, your (nanaholic) argument is predicated on the idea that if you have something in a non-free format (e.g., ALAC), you are _locked into_ that format - which is obviously untrue, as it's entirely possible to move between formats (as I would know from first-hand experience, having gone from CD -> .m4a when I used an iPod -> .flac once I stopped using an iPod and started using a J3), and with relative ease.
   
  -- Griffinhart


----------



## nanaholic

Quote: 





maverickronin said:


> Except that they all already support mp3 which doesn't have any of that either...


 
   
  I thought we were talking about FLACs?
  Also talking about mp3 is revisionist history.  The popularity of mp3 preceded the portable mp3 player - the mp3 (and mp2) format enabled the making of mp3 players so it had to be accepted for the new portable devices to be born, or the company would have to put in the R&D to develop another format, which some companies *did* try to make their own lock in formats eg wmv atrac+ and failed.  By the time Apple made iPods all the geeks and teenagers had chosen mp3 as the de facto format of sharing over the internet, mp3 had already gain too much traction for people to change, so the makers had to cave to what consumers wanted to sell their products.  FLAC is not nearly in the same situation..  
   
  Quote: 





griffinhart said:


> Incidentally, using MP3 would also mean having to pay licensing/patent fees because MP3 _isn't_ a free codec standard. And yet, people still use that, and can use it to move between software/hardware.
> 
> Also, your (nanaholic) argument is predicated on the idea that if you have something in a non-free format (e.g., ALAC), you are _locked into_ that format - which is obviously untrue, as it's entirely possible to move between formats (as I would know from first-hand experience, having gone from CD -> .m4a when I used an iPod -> .flac once I stopped using an iPod and started using a J3), and with relative ease.
> 
> -- Griffinhart


 

 I'm not talking about "hard" lock-ins where it is nearly impossible to change such as say, Sony's Memory Stick format which dictates you must get Sony made products, but "soft" lock-ins where the company intentionally makes it difficult for the average person to change (such as transcoding) - which are just as effective of a method to prevent customers from moving between different products.  Audiophiles are different because being a type of geek it is in our nature to want to tweak things to get better results, so the task of changing is not daunting thus this type of "soft" locks don't work on us.  Also if you have the CD of course moving to flac isn't a difficult task - but try telling someone with little interests in learning how to transcode apple lossless or get their DRMed iTunes songs on another brand of mp3 player they wouldn't bother.  This is the type of lock-ins I'm talking about.


----------



## Sylverant

Quote: 





griffinhart said:


> Incidentally, using MP3 would also mean having to pay licensing/patent fees because MP3 _isn't_ a free codec standard. And yet, people still use that, and can use it to move between software/hardware.
> 
> Also, your (nanaholic) argument is predicated on the idea that if you have something in a non-free format (e.g., ALAC), you are _locked into_ that format - which is obviously untrue, as it's entirely possible to move between formats (as I would know from first-hand experience, having gone from CD -> .m4a when I used an iPod -> .flac once I stopped using an iPod and started using a J3), and with relative ease.
> 
> -- Griffinhart


 

 I was only speaking about what I'd hope from apple at the most. I've been through nearly every format now as have all of us. Higher Ape levels are generally too much for good comps. I now put everything on my 2 terabyte drive in WAV+CUE from EAC while FLAC goes on my winmobile phone that I'm using portably. But FLAC is fine by itself. Its great. So I agree with you and am not that hopeful that they would give us any better than ALAC at their store. But if ALAC isn't capable of drm then I'm not even going to bother hoping that will happen now.
   
  Also, I don't understand why big name DAP's don't have native folder support. I can't tell you how much music of mine comes from Archive.org in the archaic shorten format. I couldn't imagine tagging it all to fit in a library, converting all my live Greatful Dead and estradasphere stuff to FLAC was as much as I'm willing to do. The big companies really shouldn't expect all our music to legally come from just CD's.


----------



## Griffinhart

Well, ALAC currently doesn't implement DRM, but since it uses an MP4 container, it's very likely that, if Apple wanted to, they could apply DRM to ALAC.
   
  (Not that they should, because most DRM schemata are terrible.)
   
  -- Griffinhart


----------



## grawk

When apple went with ALAC, FLAC was just one of many lossless choices.  ALAC uses the same wrapper as AAC, works everywhere FLAC does, and doesn't have the poison that is the GPL.   I've not seen a good argument for FLAC over ALAC other than people preferring GPL.


----------



## emmodad

Quote: 





wingman1659 said:


> This is exactly why I don't understand why apple wants 24bit downloads in their store. [ snip ]


 

  sigh, internet memes...... as much as i would be pleased by hi-res downloads within the apple ecosystem...... can anyone point to an actual verifiable source confirming that *apple* itself has presented info regarding intention to do 24-bit downloads?
   
  so far, all of the wild internet bloviating on this claim seems to trace back to one common source:  a CNN article which was speculation - not presentation of verified fact - based on a PR statement by record label exec Jimmy Iovine (very clearly a positioning statement for his own biz).  google on iovine apple 24 bit pipes and you will find it, relevant excerpt:  "We've gone back now at Universal, and we're changing our pipes to 24 bit. And Apple has been great," Iovine said. "We're working with them and other digital services -- download services -- to change to 24 bit. And some of their electronic devices are going to be changed as well. So we have a long road ahead of us."
   
   
  Quote: 





sylverant said:


> Its either all or nothing with apple huh? They can't just convert the entire store to CD quality ALAC; they have to offer the premium experience at high cost alongside a small catalogue?  [ snip ]


 
   
  Quote: 





griffinhart said:


> >ALAC
> >standard
> 
> /facepalm
> ...


 

 you might find this (re Apple, ALAC, FLAC, open source, licensing...) interesting in understanding why Apple provide no native FLAC support
   
   
  chuck


----------



## grawk

FLAC is free as in communism.  ALAC is free as in beer.


----------



## khaos974

emmodad said:


> You might find this (re Apple, ALAC, FLAC, open source, licensing...) interesting in understanding why Apple provide no native FLAC support
> 
> 
> chuck




Are you sure that integrating FLAC would be such a large issue due to the copyleft? To my knowledge, if you keep that piece of code distinct from the code of the main program, it wouldn't contaminate the the whole code. Besides there are already plenty of (proprietary code based) programs that encode and decode FLAC, wouldn't they be affected by the GPL license?


----------



## Griffinhart

Easier said than done; for something like foobar, where its FLAC playback capability can be added/removed via a component, that's easy. But iTunes isn't modular like foobar is, so if they stick FLAC playback capability in, they're pretty much having to integrate that capability into the entirety of the program.
   
   
  Quote: 





grawk said:


> FLAC is free as in communism.  ALAC is free as in beer.


 

 lol. You keep thinking that; meanwhile, I will continue to use awesome GPL-licensed software like the entire Cygwin suite (because Windows has a terrible dang CLI that needs to be taken out back and shot. Repeatedly) and Notepad++ (because Windows has terrible dang text editor that can't even be really called a "text editor" by the correct definition of the phrase).
   
  -- Griffinhart


----------



## grawk

It's not about how it works for the users, it's how it works for the companies implementing the software.


----------



## Griffinhart

There are DAPs that can do FLAC playback (Archos and Cowon brands come to mind) - what are those DAPs doing that iDevices/Zune aren't (or vice-versa)?
   
  -- Griffinhart


----------



## grawk

Quote: 





griffinhart said:


> There are DAPs that can do FLAC playback (Archos and Cowon brands come to mind) - what are those DAPs doing that iDevices/Zune aren't (or vice-versa)?
> 
> -- Griffinhart


 


  Publishing the source code to their entire device.


----------



## Sylverant

Quote: 





griffinhart said:


> There are DAPs that can do FLAC playback (Archos and Cowon brands come to mind) - what are those DAPs doing that iDevices/Zune aren't (or vice-versa)?
> 
> -- Griffinhart


 

 Are there even any archos devices worth FLAC in soundquality though? The Archos 5 android looks like it would be amazing for storage and playing dvd vob's but I don't remember how good its soundquality was.


----------



## Griffinhart

Quote: 





grawk said:


> Publishing the source code to their entire device.


 

 I'm calling BS; I have yet to see the source code for my Cowon J3.
   
  -- Griffinhart


----------



## xnor

IIRC this topic is about* audio bit depth* and not DAPs, lossless formats, licensing or something like that..


----------



## grawk

Whether they've done it or not, that's the legal requirement.  And if microsoft or apple included flac support, and didn't include the source for everything, you know they'd get sued.
  
  Quote: 





griffinhart said:


> I'm calling BS; I have yet to see the source code for my Cowon J3.
> 
> -- Griffinhart


----------



## grawk

Quote: 





xnor said:


> IIRC this topic is about* audio bit depth* and not DAPs, lossless formats, licensing or something like that..


 


  But nothing new has been said about bit depth in over a year.  It's just rehashing old arguments.  At least this is a new argument.


----------



## xnor

Quote: 





grawk said:


> But nothing new has been said about bit depth in over a year. It's just rehashing old arguments.  At least this is a new argument.


 

 Then so be it. No need to revive it with unrelated discussions. Instead start a new thread, right?


----------



## grawk

discussions happen where discussions happen.  As long as no one is breaking the terms of service, I don't see a problem letting things happen organically.


----------



## WobblyGoblin

GPL is poison? communism? 

Communism requires force to exist. You're not forced to open source your code if your code supports FLAC, only if your FLAC support is derived from open source code, and you're not forced to use open source code.

Please, get a clue and stop spreading lies.


----------



## grawk

My bad, I thought the codec was GPL'd, but I guess not.  So I don't know why Apple chose not to do FLAC.


----------



## EddieE

My guess would be the obvious - they have iTunes that most average consumers buy their music from - like any corporation they want to make the most money possible from that situation.
   
  Creating their own lossless codec means one of two things for them - other manufacturers pay their license fees to play it back on their devices or the consumers have to keep buying their ipods to play their music, forever.
   
  Pretty good racket.


----------



## grawk

Apple doesn't charge licensing fees for ALAC


----------



## xnor

It doesn't take a rocket scientist to see that *control *is very important to Apple. To put it bluntly, they are control-freaks and also suffering from NIH syndrome.


----------



## EddieE

Grawk,
So what's the deal, they just don't let other people use it full stop, or everyone else choses not to have it?


----------



## xnor

If anyone's still interested in the original topic of this thread, here's the spectrum of an unprocessed drum recording (recorded with a bit depth of 24 bits, average RMS amplitude: -32 dB !!!):
   

   
  The green line at the bottom shows added noise due to re-quantization to 15 bits + 1 bit simple dithering, no noise shaping was used.
   
  High end DACs like the WM8742 manage to reach a signal-to-noise ratio of about 120 dB under ideal conditions.
   
  To quote Arnold from HA: "While 24 bits is an insane overkill format, 16 bits is an overkill format."


----------



## grawk

for playback, 16 bits is plenty, for recording dynamic music, 24 bits lets you leave lots of headroom to prevent problems.
   
  As to why others don't use ALAC, I know some do and some don't.  I don't really care why, as I only use apple devices.


----------



## blackmetal88

[size=medium]Hi All, I see this conversation has kind of shifted directions. Nevertheless, I have written the below in response to the original post regarding 24-bit/16-bit music:

 The technical description by _GREGORIO_ is pretty good and accurate. However,  the value, or interpretation, of the facts is a little misguided. The number of bits does indeed define the dynamic range, or difference between softest sounds and loudest sounds, expressed in decibels. Each bit adds 6dB of dynamic range. Humans have the capability of hearing over a 100dB range (30dB to 130dB). 16-bit encoding has a 96dB dynamic range, and 24-bit has 144dB. So while you can argue that 24-bit is overkill, 16-bit is just adequate.

 The two points missing are:

 1.  Physical parts and circuits are not perfect. Digital-to-analogue converter chips never meet their theoretical capability due to resistance, capacitance, and inductance in the chip and in the circuit. 16-bit circuits were often 14-bit and sometimes as bad as 12-bit.  24-bit circuits are even more difficult and seldom are accurate beyond 20-bits. But almost everyone (audio engineers that is) thinks 20 bits is sufficient  - that is 120dB dynamic range.

 2. Tests performed at the dawn of digital technology proved that with analogue recordings, there was useful signal below the 'noise floor' that could be perceived by listeners. So even though a good LP only had a signal to noise ratio of only 70dB, there was perceivable signal 15dB below that level.
  
  With digital, when you run out of bits you're done. Any signal below the cut-off point is, well, cut-off/missing. This is the big rap against 'digital sound'.  A lot of information is missing never to return. This is why having the extra margin of 20 bit+ recording important.  It allows the recording engineer to have 'margin'. Because with live music the loudness can never be exactly predicted, recording engineers always leave a 6 dB (2-bits) available for something unexpectedly loud. They also use 'compressors' to be sure maximum '0dB' is never exceeded, but the compressor adds its own distortion (though not as bad as going above 0dB in a digital recorder).

 So what music is down at these very quiet levels? Mostly, the dying reverb of an instrument and environmental or ambient sound of the recording space.

 With many pop recordings there is no recording space as all instruments are plugged into the mixing board and vocals have artificial reverb added.  But acoustic music usually is recorded in a venue. The low level sounds tell the listener about that space – we can tell a large room from a small room by sound alone. So systems with wide dynamic range have 'air' and 'space' as well as all the instrumental nuance that define the difference between a Guarneri and Stradivarius violin or a Gibson and a Fender guitar. Sure, this is subtle, but it is the fine details that are the difference between good and excellent.

 The argument that recorded music has a very narrow dynamic range is again accurate but misleading. Sometimes it is narrow for artistic reasons, and sometimes it is narrow due to technical restrictions. A live symphony orchestra experience does cover a 100dB range from quiet to full orchestral crescendo. Artists and engineers will exploit the improved dynamic range capability of the medium to make more lifelike recordings in the future.
   ​[/size]


----------



## khaos974

blackmetal88 said:


> [size=medium]Hi All, I see this conversation has kind of shifted directions. Nevertheless, I have written the below in response to the original post regarding 24-bit/16-bit music:
> 
> The technical description by _GREGORIO_ is pretty good and accurate. However,  the value, or interpretation, of the facts is a little misguided. The number of bits does indeed define the dynamic range, or difference between softest sounds and loudest sounds, expressed in decibels. Each bit adds 6dB of dynamic range. Humans have the capability of hearing over a 100dB range (30dB to 130dB). 16-bit encoding has a 96dB dynamic range, and 24-bit has 144dB. So while you can argue that 24-bit is overkill, 16-bit is just adequate.
> 
> ...




You forget that no one advocates for recording at 16 bit, what we are taking about is playback in domestic venues and if 16 bit + dithering is enough. Personal opinion: YES!

And about FLAC, GPL and viral contamination? Guess what, the libraries and the codec specifications are under BSD, which means absolutely no contamination effect!


----------



## xnor

Quote: 





blackmetal88 said:


> 1.  Physical parts and circuits are not perfect. Digital-to-analogue converter chips never meet their theoretical capability due to resistance, capacitance, and inductance in the chip and in the circuit. [...]


 

 What you miss here is that the recording equipment, the entire chain from the mics to the A/D conversion, usually has a lower signal to noise ratio than what 16-bit D/A converters can achieve.
  To quote Ethan: "You'll be lucky to have 60 dB s/n and _really_ lucky to have 70." (mic + preamp + adc, even in a professional recording studio)
   
  Quote: 





> 2. Tests performed at the dawn of digital technology proved that with analogue recordings, there was useful signal below the 'noise floor' that could be perceived by listeners. So even though a good LP only had a signal to noise ratio of only 70dB, there was perceivable signal 15dB below that level.


 
   
  So approximately 85 dB (with mostly noise down there I guess) which is still inferior to 16-bit digital audio (96 dB). Even if you had a magnificent vinyl setup you'd only need about 14-bits in a digital system which would still be a lot more accurate in various other aspects (no clicks or pops, no large frequency deviations, no wear, etc).
   
   
  Quote: 





> With digital, when you run out of bits you're done. Any signal below the cut-off point is, well, cut-off/missing. This is the big rap against 'digital sound'.  A lot of information is missing never to return.


 
   
  And with analog there is no clipping nor a noise floor and you can store indefinitely silent signals, right? Sorry for the rant, but that's nonsense and, in fact, you get more headroom with digital.
  More information is missing (never to return!) with analogue recordings. And see the previous paragraph, you kinda contradict yourself here.
   
   
  Quote: 





> This is why having the extra margin of 20 bit+ recording important.  It allows the recording engineer to have 'margin'. Because with live music the loudness can never be exactly predicted, recording engineers always leave a 6 dB (2-bits) available for something unexpectedly loud.


 
   
  Even if that were true it wouldn't matter in a playback system with a 16-bit format.
   
   
  Quote: 





> The argument that recorded music has a very narrow dynamic range is again accurate but misleading. Sometimes it is narrow for artistic reasons, and sometimes it is narrow due to technical restrictions. A live symphony orchestra experience does cover a 100dB range from quiet to full orchestral crescendo. Artists and engineers will exploit the improved dynamic range capability of the medium to make more lifelike recordings in the future.


 
   
  Any references? The Handbook of Recording Engineering (John Eagle, 2005) says that the dynamic range in a concert hall normally doesn't exceed 80 dB.
  Also, please feel free to post 24-bit samples (10 to 30 seconds should do) of orchestral recordings that you think exploit the additional 8 bits. I'm very curious. I don't listen to classical music but I've taken a closer look at some free, supposedly audiophile-quality 24-bit recordings and found that the dynamic range didn't exceed that of what 16 bits are capable of storing.


----------



## nick_charles

Quote: 





xnor said:


> Any references? The Handbook of Recording Engineering (John Eagle, 2005) says that the dynamic range in a concert hall normally doesn't exceed 80 dB.
> Also, please feel free to post 24-bit samples (10 to 30 seconds should do) of orchestral recordings that you think exploit the additional 8 bits. I'm very curious. I don't listen to classical music but I've taken a closer look at some free, supposedly audiophile-quality 24-bit recordings and found that the dynamic range didn't exceed that of what 16 bits are capable of storing.


 

 I have about 700 Classical CDs , not one of them has a dynamic range anywhere close to CD limits - the CD I have with the biggest range Solti/CSO Mahler 1 (DDD) notches up about 62db  (measured)  it peaks at 0db in several places


----------



## xnor

Quote: 





nick_charles said:


> I have about 700 Classical CDs , not one of them has a dynamic range anywhere close to CD limits - the CD I have with the biggest range Solti/CSO Mahler 1 (DDD) notches up about 62db  (measured)  it peaks at 0db in several places


 

 Thanks nick_charles. 60-ish dB max also lines up with what I've analyzed.


----------



## gregorio

After more than 2 years, I thought I'd pop back here briefly to see how it was going and to thank those of you for the kind emails you sent. I started this thread to provide information to help you, the consumers, to avoid the huge amount of snake oil present in the audio and HiFi industry. I'm glad I was of help to many of you. I'm not so glad that the industry continues using deception as a marketing tool. If 24bit is a waste of time (and space) for playback and 96kHz sampling rate is already higher than the optimum rate for Analogue to Digital Converters, the latest ADCs boast 32bit and 384kHz! We are getting further and further away from the optimal design, therefore resulting in less accurate conversion, while the marketing is trying to convince you of exactly the opposite. This is a real shame, instead of developing better quality filters, clocking mechanisms and analogue components in 96kHz converters to give us better quality audio, the industry has decided it's cheaper and easier to develop poorer quality converters (at 192kHz and 384kHz) and take us for fools by marketing them as better. It's deceptive and unethical and makes me pretty mad as a consumer myself.
   
  The paper I linked to previously (http://www.lavryengineering.com/documents/Sampling_Theory.pdf) is as true today as it was the day it was written. Computing power has obviously dramatically increased since 2004 but there is a finite speed limit to how quickly a capacitor can be charged and amps can settle, no amount of computing power is going to change the laws of physics. I was pleased to see at least one other manufacturer come out recently and tell the truth about sampling rates. My estimation of them as a company has been greatly enhanced: Benchmark official statement 96kHz vs.192kHz
   
  I might pop back here occasionally (time permitting), to answer questions where I'm able and to help those others here like me trying to combat the marketing fraud being perpetrated. Accurate information is the best weapon in my opinion. If the highly offensive personal attacks from shills and those entrenched in their ignorance starts up again though I'll just call it a day again.
   
  Regards,
   
  G


----------



## Tronz

Quote: 





gregorio said:


> After more than 2 years, I thought I'd pop back here briefly to see how it was going and to thank those of you for the kind emails you sent. I started this thread to provide information to help you, the consumers, to avoid the huge amount of snake oil present in the audio and HiFi industry. I'm glad I was of help to many of you. I'm not so glad that the industry continues using deception as a marketing tool. If 24bit is a waste of time (and space) for playback and 96kHz sampling rate is already higher than the optimum rate for Analogue to Digital Converters, the latest ADCs boast 32bit and 384kHz! We are getting further and further away from the optimal design, therefore resulting in less accurate conversion, while the marketing is trying to convince you of exactly the opposite. This is a real shame, instead of developing better quality filters, clocking mechanisms and analogue components in 96kHz converters to give us better quality audio, the industry has decided it's cheaper and easier to develop poorer quality converters (at 192kHz and 384kHz) and take us for fools by marketing them as better. It's deceptive and unethical and makes me pretty mad as a consumer myself.
> 
> The paper I linked to previously (http://www.lavryengineering.com/documents/Sampling_Theory.pdf) is as true today as it was the day it was written. Computing power has obviously dramatically increased since 2004 but there is a finite speed limit to how quickly a capacitor can be charged and amps can settle, no amount of computing power is going to change the laws of physics. I was pleased to see at least one other manufacturer come out recently and tell the truth about sampling rates. My estimation of them as a company has been greatly enhanced: Benchmark official statement 96kHz vs.192kHz
> 
> ...


 
  Thanks for your honesty, we need more people like you.


----------



## pdupiano

There's only one thing that confuses me about the first post. 
   
  If what people hear from 16-bit systems is the recording + noise from Dither, then doesn't that mean that with larger bits, we would hear less noise and dithering would be unnecessary? 
   
  But Gregorio goes on to state that most recordings do not even play back a dynamic range of 60db and so even a 16bit system is overkill. And Nick_Charles stated earlier that the most he's been able to measure is roughly in the same range (and confirmed by Xnor). So does that mean that what we really need is a 12bit system? If that is true, then dithering is completely unnecessary because it is something used with 16bit systems for improvements, but clearly a 12 bit system is more than enough (based on classical music -generally agreed upon with having a greater dynamic range-having roughly a 60db dynamic range). So that would mean that Dithering should do absolutely nothing in a 16 bit system, and no one should be able to perceive it. Is that the case? Based on the first post, it appears that dithering is commonly used because it is beneficial and individuals can perceive its effects. 
   
  And so now the next line of questioning would be, if dithering does not matter then technically all audio products in the market are actually over kill. And all arguments against going insanely overkill are quite frankly moot. If I were to use a bat to kill a fly that would be overkill, and using a grenade to kill it would be.. well the same. Both equally stupid, and both equally acceptable as fly killers, especially when no one sells fly swatters. 
   
  If the answer is that dithering does matter, then we are capable of perceiving a noise level exceeding the capabilities of CD recording, in which case, it might be beneficial to go past 16 bit. It doesn't necessarily mean we must go to 24 bit, just a few bits more. Take for example HDCDs, which if I remember correctly are 20 bit rather than 24. 
   
  So i guess what I'm really asking is, does ditherning matter? Oh and does anyone make 12 bit audio gear?


----------



## grawk

dither matters when something actually uses the dynamic range.  If it's not using it, it'll be over 90 db below the level of the music, so you won't hear it.


----------



## gregorio

Quote:


pdupiano said:


> There's only one thing that confuses me about the first post.
> 
> If what people hear from 16-bit systems is the recording + noise from Dither, then doesn't that mean that with larger bits, we would hear less noise and dithering would be unnecessary?
> 
> ...


 

 Good questions! You are correct in your assumption that you can't hear the noise and 16bit is already overkill, which was the whole point of my original post, to dispel the market hype about 24bit. Even in a listening environment equivalent to a top class recording studio, the dither noise of 16bit will be below the noise floor of the playback equipment and/or the noise floor in the listening environment. The only error in your reasoning is the consequence of not using dither. Dither is a mathematical process which randomises all the quantisation error, the result is a perfectly encoded (and reconstructed) waveform plus some noise (which is the result of the randomised quantisation error). If we don't apply dither the consequence is potentially correlated (non-randomised) error, which is much more noticeable and unpleasant.
   
  The only thing I'd take issue with is that I personally would say that using a grenade to kill a fly is not "equally" but slightly more stupid than using a bat! 
	

	
	
		
		

		
		
	


	




 Generally a quite good analogy and understanding of the situation though. I would not say that 12bit was quite enough, although you theoretically only need 10 bits to encode a 60dB dynamic range. The overkill of 16bit ensures that the dither noise is below the noise of your playback environment and therefore you can forget all about it. Dither noise using 12bit would probably add enough to the noise of a good sound system to be perceivable even at normal hearing levels.
   
  Hope this helps,
   
  G


----------



## Kees

Computers work not with single bits but with 8-bit words. Hence every increase would be an 8 bit increase.


----------



## Griffinhart

Quote: 





kees said:


> Computers work not with single bits but with 8-bit words. Hence every increase would be an 8 bit increase.


 


 This is technically not a true statement; the size of a byte (your "8-bit word") is hardware-dependent, and there exists no standard that specifically defines a byte as 8 bits long. 8-bit bytes just make a lot of computations and the like _really_ easy (because powers of two) compared to the alternatives. Back in Ye Olden Dayes of the 50s and 60s, 6-bit words and 7-bit words existed. IBM made the 8-bit word hecka popular though, with its System/360 architecture and 8-bit microprocessors.
   
  When you want to unambiguously specify an 8-bit word, you say "octet" (because, like I said before, there's no actual standard that specifically defines a word/byte as 8 bits long).
   
  -- Griffinhart


----------



## pdupiano

Quote: 





gregorio said:


> Quote:
> 
> Good questions! You are correct in your assumption that you can't hear the noise and 16bit is already overkill, which was the whole point of my original post, to dispel the market hype about 24bit. Even in a listening environment equivalent to a top class recording studio, the dither noise of 16bit will be below the noise floor of the playback equipment and/or the noise floor in the listening environment. The only error in your reasoning is the consequence of not using dither. Dither is a mathematical process which randomises all the quantisation error, the result is a perfectly encoded (and reconstructed) waveform plus some noise (which is the result of the randomised quantisation error). If we don't apply dither the consequence is potentially correlated (non-randomised) error, which is much more noticeable and unpleasant.
> 
> ...


 

 Thanks for responding Gregorio, but the real question that I was trying to bring up is "sort of" answered by your last statement. Are you stating that people can perceive the difference between a 16bit recording with and without dither? If this is the case, then doesn't that imply that we do in fact perceive up to the full 16 bit dynamic range?
   
  On a side note, I wonder if other folks have started to use their 24 bit audio systems for data recording. I've used audio amplifiers and simple diy DACs for my own measurements in my lab, and now I'm very curious to see what kind of results I get with a 24 bit audio dac versus a 24 bit data acquisition dac, perhaps there are additional signal processes in the 24 bit audio dac not found in a DAQ card.


----------



## kiteki

I heard a 32 bit DAC will make 16 and 24 bit recordings sound better, does anyone have any deep technical insight on this? I'm talking about the latest (and long postponed/awaited) Fostex headphone amps in Japan, they use some Asahi Kasei DAC Chip.


----------



## Head Injury

Quote: 





kiteki said:


> I heard a 32 bit DAC will make 16 and 24 bit recordings sound better, does anyone have any deep technical insight on this? I'm talking about the latest (and long postponed/awaited) Fostex headphone amps in Japan, they use some Asahi Kasei DAC Chip.


 

 24 bit doesn't make 16 bit sound better. I don't see why 32 bit will make anything betterer.
   
  Unless you listen to your music at 192dB. If so, a 32 bit DAC might be for you!


----------



## kiteki

Yes, I see, however I'm curious why does the 32 bit chip exist in that case, and why are Fostex using it?


----------



## Head Injury

Quote: 





kiteki said:


> Yes, I see, however I'm curious why does the 32 bit chip exist in that case, and why are Fostex using it?


 

 Marketing purposes, probably. ASUS is using a 32 bit chip in their new DAC. There's a DAC somwhere which supports up to 384kHz sampling rate. Slap a bigger number on it and audiophiles will flip out, even if there's no recordings that would actually use those extra bits.


----------



## kiteki

Thanks, maybe I'll save money and not get the Fostex HP-A3 afterall, it does look nice though and is receiving positive reviews and has other nice components inside.


----------



## gregorio

Quote: 





pdupiano said:


> Thanks for responding Gregorio, but the real question that I was trying to bring up is "sort of" answered by your last statement. Are you stating that people can perceive the difference between a 16bit recording with and without dither? If this is the case, then doesn't that imply that we do in fact perceive up to the full 16 bit dynamic range?
> 
> On a side note, I wonder if other folks have started to use their 24 bit audio systems for data recording. I've used audio amplifiers and simple diy DACs for my own measurements in my lab, and now I'm very curious to see what kind of results I get with a 24 bit audio dac versus a 24 bit data acquisition dac, perhaps there are additional signal processes in the 24 bit audio dac not found in a DAQ card.


 

 Sorry, I thought I explained this when I said "If we don't apply dither the consequence is potentially correlated (non-randomised) error, which is much more noticeable and unpleasant." In other words, correlated error results in unwanted frequencies or tones in the music but probably not related to the musical content. It's probable that these correlated errors are higher in amplitude (louder) than the dither. Chances are that even with a very high end system correlated errors will be near the limits of hearing at normal listening levels but we use dither just to play it safe. Another fact: When we record music there is usually a great deal more noise in the recording chain than is created by dither, so any dither noise in the finished product should be masked and unrecognisable. This may not be true of correlated errors.
   
  I don't know what the difference is between a 24bit audio DAC and 24 bit data acquisition DAC, what is a 24bit data acquisition DAC?
   
  G


----------



## mukumi

Hi, concerning the Hi Definition audio that you can buy on certain websites, here is an interesting article : http://www.itrax.com/Pages/ArticleDetails.php?aID=32


----------



## bigshot

High bitrate is like uncompressed files in the sense that it isn't about what you can hear, it's about the peace of mind bigger filesizes give people with OCD. You could double the filesize again with pointless filler bits and there would still be people to buy it "just in case".


----------



## gregorio

kiteki said:


> I heard a 32 bit DAC will make 16 and 24 bit recordings sound better, does anyone have any deep technical insight on this? I'm talking about the latest (and long postponed/awaited) Fostex headphone amps in Japan, they use some Asahi Kasei DAC Chip.




A couple of things to note. First 32bit, now I don't know if they are 32bit integer or 32bit float, my guess would be the latter. If you didn't know, there is very little resolution difference between 32bit float and 24bit integer. 8 bits of a 32bit float "word" are reserved to specify the position of the floating point (exponent), leaving a 24bit mantissa which stores the actual value, hence, pretty much the same level of accuracy as a 24bit integer file. Second, when ever we perform a digital process on a 24bit file (EQ, compression, reverb, etc. etc.) we are performing a mathematical calculation which may not provide and integer result, this result is truncated or rounded, producing a small error. Not really a problem down at the -144dB level but when you are mixing music you may have a 100 or more such processes across 50 or more tracks. Summing all these errors together on mix down may cause problems, so these days professional processing is usually done at 32bit float, 48bit integer or sometimes even 64bit float. Any errors are so minute you can sum many hundreds together and hear nothing. So, these very high bit rates are useful professionally when mixing but for playback, 32bit is just as pointless 24bit.

One last point: 32bit integer should in theory provide 192dB dynamic range, have a look at the S/N ratio output of any so called 32bit DAC, what do the specs say; 110dB, 120dB maybe even 130dB? Hang on though, even at 130dB what's happened to the other 62dB? That represents over 10bits of data, just gone, what's happened to it? It's just noise! Even the very finest DACs out there cannot re-produce more than 22bits worth of dynamic range and this is not going to change until someone can come along and change the laws of physics. Literally, we have just about reached the absolute limit of the laws of physics and incidentally, exceeded what a human being can safely listen to by a factor of roughly 1000 (at 22bits)! Virtually no recording ever released exceeds a dynamic range of about 60-70dB.

So why a DAC with 32bits. 32bit float processors were the main format of the computing world for many years, I'm guessing here but it maybe that 32bit float processors are cheaper or no more expensive to manufacture than a 24bit integer processors, giving a marketing advantage at no or reduced cost. Whether it's cheaper to manufacture or not, 32bit DACs can be sold for higher prices because consumers can easily be convinced that 32bit is better than 24bit!

G


----------



## gregorio

head injury said:


> There's a DAC somwhere which supports up to 384kHz sampling rate. Slap a bigger number on it and audiophiles will flip out, even if there's no recordings that would actually use those extra bits.




It's worse than that I'm afraid, it not just a case of recordings being able to use those bits or extra bandwidth theoretically provided by 384kHz sampling rates. I only discussed sampling rates a little much earlier in this thread but we never really considered the whole thing together, sampling rates and bit depth. Electronic engineering and signal processing these days quite often comes up against the laws of physics and there is an axiom; the larger the bandwidth the lower the accuracy. In professional Analog to Digital Converters (ADCs) it's not uncommon to oversample by 256 or even 512 times, IE. A sample frequency of over 22mHz giving an audio bandwidth of over 11mHz but at this large bandwidth we are only capable of 6 or so bits of accuracy. It is possible to sample at over a gHz but the accuracy would obviously be proportionately lower. This isn't a problem which is going to go away. It's not a question of throwing more computing power at the problem, it's a limitation of the laws of physics. A great deal of ADC design is about trade offs!

So, what is the trade off or optimal bandwidth to accuracy ratio for music (as opposed to other signal processing, say in the telecom industry)? If we say that 24bits is the accuracy which we want (for recording) then the best evidence available at this point in time suggests a sampling rate of roughly 60-70kHz. Unfortunately, the industry has decided not to implement say a 65kHz sampling frequency so the best trade off would be 88.2kHz. 88.2kHz gives plenty of space for smooth and error free filtering (required by the Nyquist Theorem) and only a marginal and relatively inconsequential loss of accuracy. But this is not true of 192kHz and even less true of 384kHz. There is a price to pay for larger bandwidths and the larger the bandwidth the larger the price. The price is paid in non-linearity, in other words 192kHz isn't just about the futility of trying to record and reproduce irrelevant audio frequencies, it actually reduces the quality of those frequencies which you can hear! Yes, you are understanding correctly, 24/192 is actually poorer quality audio than say 24/96! For evidence of this please read the white paper published by Dan Lavry and confirmed last year by Benchmark (links on the previous page). The Lavry paper is quite technical but the Benchmark confirmation is written in a way everyone can understand. Lavry's paper was published in 2004, so it's not as if it's a recent discovery. That hasn't stopped the DAC manufacturers and some in the music industry using marketing to convince you that 24/192 is better than 24/94 and then charging you a premium for it! So, lower quality now seems to cost you a higher price, please don't get sucked in!!!!!

I want to make it clear, I am not saying that you won't hear a difference between a 24/96 and a 24/192 recording, it's possible you might, as there is more noise and distortion present in a 24/192 recording than there is in a 24/96 recording. It's obvious that many reviewers seem to prefer 192 sample rates, I can't say if this is because they are following the advertising revenue, believe 192 should sound better and therefore hear an improvement or whether they honestly prefer the sound of more noise/distortion. The bottom line though hasn't changed from when I started this thread. 16bit exceeds the resolution of playback systems (and your ears) and 70kHz sampling rate exceeds the bandwidth required to eliminate all artefacts. Unfortunately 16/70 format doesn't exist, neither does 16/88.2 or 16/96, so the best trade off that the industry allows us is 24/88.2 or 24/96. This represents the highest quality audio format which is currently available.

G


----------



## kiteki

Thanks for your response Gregorio, the Dac/Amp I'm talking about is the Fostex HP-A3 and I'll delve further and try to find out more, but your post was very insightful.


----------



## pdupiano

Quote: 





gregorio said:


> Sorry, I thought I explained this when I said "If we don't apply dither the consequence is potentially correlated (non-randomised) error, which is much more noticeable and unpleasant." In other words, correlated error results in unwanted frequencies or tones in the music but probably not related to the musical content. It's probable that these correlated errors are higher in amplitude (louder) than the dither. Chances are that even with a very high end system correlated errors will be near the limits of hearing at normal listening levels but we use dither just to play it safe. Another fact: When we record music there is usually a great deal more noise in the recording chain than is created by dither, so any dither noise in the finished product should be masked and unrecognisable. This may not be true of correlated errors.
> 
> I don't know what the difference is between a 24bit audio DAC and 24 bit data acquisition DAC, what is a 24bit data acquisition DAC?
> 
> G


 
   
  Ah ok, thanks for going over the possible perceived errors(artifacts) versus the randomization caused by dither to remove these artifacts.
   
  As far as data acquisition goes (DAQ), its basically the opposite of a DAC so instead of Digital => Analog, it would be Analog => Digital. Most of the gear I work with is limited to 16bits, so a 24 bit DAQ would be amazing. When processing signals, a DAC would still be very helpful because we do the intial A=>D conversion and then manipulate the data. As you stated there is a lot of noise in the chain to obtain the analog signals so we need to filter, modify, and use other techniques to get rid of parts of the data or change it to something meaningful.  We then output the data back to an analog signal so we still need to do a D=>A Conversion. At the moment that DA conversion is limited to 16 bits, going higher might be better. Since these signals are being perceived by instruments rather than human ears, the 24bit DAC would probably be put to good use.


----------



## gregorio

pdupiano said:


> As far as data acquisition goes (DAQ), its basically the opposite of a DAC so instead of Digital => Analog, it would be Analog => Digital. Most of the gear I work with is limited to 16bits, so a 24 bit DAQ would be amazing. When processing signals, a DAC would still be very helpful because we do the intial A=>D conversion and then manipulate the data. As you stated there is a lot of noise in the chain to obtain the analog signals so we need to filter, modify, and use other techniques to get rid of parts of the data or change it to something meaningful.  We then output the data back to an analog signal so we still need to do a D=>A Conversion. At the moment that DA conversion is limited to 16 bits, going higher might be better. Since these signals are being perceived by instruments rather than human ears, the 24bit DAC would probably be put to good use.




OK, thanks, I have no experience of DAQs, just ADCs and DACs. I have no idea how a 24bit DAQ works or what sort of signals you will be feeding it. But you have to be very careful with audio DAC specifications. If you understood the original post of this thread you will realise that the only thing that bit depth represents is the dynamic range, the range between the noise floor and the maximum output level (technically the signal to noise ratio, SNR). When they advertise themselves as a 24bit DAC, all they mean is the digital word length of data they are able to accept, not the resolution they are able to output! So, 16bit represents a maximum SNR of 96dB and 24bit represents a maximum SNR of 144dB. But no DAC on the market has an SNR of 144dB due to the limitations of the laws of physics in a circuit, the actual SNR varies from model to model, so it may actually be possible to find a 16bit DAC that has a higher SNR (resolution) than a 24bit DAC! In other words, it would be relatively easy in theory to make a 64bit DAC. Just allow for the input of a 64bit word length, hack off 48 bits and output the remaining 16bits worth of resolution (SNR = 96dB). In my opinion the marketing of DACs is highly misleading because the advertised bit depth gives no indication of the actual output resolution. The best you can do is look as the Signal to Noise specification of the DAC, divide that number by 6(dB) and that gives you the digital output resolution (bit depth). One other thing to look out for: The S/N specification for DACs is given in dB but some manufacturers are very crafty and use an A weighted scale, which is entirely inappropriate and will could indicate a significantly lower S/N than implied by the specification. It is possible that a DAC with a S/N spec of 110dB actually has a significantly larger dynamic range (output resolution) than a DAC with an S/N spec of 112dB (A). Sorry, just another one of the myriad marketing tricks employed by the audio industry. Try if you can to get the S/N spec in un-weighted dB, otherwise the 6dB per bit calculation will not be accurate.

G


----------



## Kost

I did some testing myself with my Audio-technica ATH-50 headphones and a fiio E5 amp. I did not notice any difference between 16/44.1 and 24/96. (Yes, my sound card does 24/96) Then again, I don't have 5k dollars worth of sound equipment either. I will say though that I am one of those people who can hear above the 20khz range, and quite frankly you wouldn't want to even if you could. It sounds like nothing but an annoying, super high pitched whine.
   
  Have you ever had a kid in come up to you in school and say "hey, you want to hear the most annoying sound in the world?", it sounds kind of like that. Whether those sounds can affect other sounds in such a way as to improve sound quality.. I dont know but i highly doubt it, and if so, definitely not with my setup.
   
  Thank you Gregorio, although im very new to higher grade sound. I found this thread very enlightening. Im surprised someone of your knowledge and skills would devote so much time to helping me better understand how my digital audio works. Thanks again!


----------



## spagetka

You are lucky. Not 5k on equipment, just less than 2k on Benchmark DAC1 PRE, AKG K701 and good usb cable.
  
  Quote: 





kost said:


> I did some testing myself with my Audio-technica ATH-50 headphones and a fiio E5 amp. I did not notice any difference between 16/44.1 and 24/96. (Yes, my sound card does 24/96) Then again, I don't have 5k dollars worth of sound equipment either. I will say though that I am one of those people who can hear above the 20khz range, and quite frankly you wouldn't want to even if you could. It sounds like nothing but an annoying, super high pitched whine.
> 
> Have you ever had a kid in come up to you in school and say "hey, you want to hear the most annoying sound in the world?", it sounds kind of like that. Whether those sounds can affect other sounds in such a way as to improve sound quality.. I dont know but i highly doubt it, and if so, definitely not with my setup.
> 
> Thank you Gregorio, although im very new to higher grade sound. I found this thread very enlightening. Im surprised someone of your knowledge and skills would devote so much time to helping me better understand how my digital audio works. Thanks again!


----------



## Coop

Quote: 





> Originally Posted by *Kost* /img/forum/go_quote.gif
> Have you ever had a kid in come up to you in school and say "hey, you want to hear the most annoying sound in the world?"


 


 I used to respond to that with "I just did, thanks" and just walk away 
   
  Very enlightning thread this!


----------



## kingpage

Is there any advantagge (or disadvantage) of using a DAC like E10 with 24bit/96Hz capability compared with a 16bit DAC, when playing 16bit 44Hz or 48Hz audio such as CD, MP3s or DVDs?
   
  I know audio upsampling is not exactly like image upscaling where pictures can look much better with Lanczos or SmartEdge interpolation algorithms, but it has been on my mind for a while now whether good audio upsampling would bring about better quality audio. For example, perhaps, a low bitrate mp3 which had gone through a low-pass filter can have some of the lost high frequencies recreated from upsampling?
   
  Any expert opinion would be very helpful. Thanks.


----------



## grawk

The advantage to various dacs, when dealing with 44/16 or 48/16 tracks is the sound, not whether they'll decode other sample rates.  If it sounds better, it's better.  Whether it can do something you don't need it to do doesn't end up mattering much.


----------



## nikongod

Quote: 





kingpage said:


> Is there any advantagge (or disadvantage) of using a DAC like E10 with 24bit/96Hz capability compared with a 16bit DAC, when playing 16bit 44Hz or 48Hz audio such as CD, MP3s or DVDs?
> 
> I know audio upsampling is not exactly like image upscaling where pictures can look much better with Lanczos or SmartEdge interpolation algorithms, but it has been on my mind for a while now whether good audio upsampling would bring about better quality audio. For example, perhaps, a low bitrate mp3 which had gone through a low-pass filter can have some of the lost high frequencies recreated from upsampling?
> 
> Any expert opinion would be very helpful. Thanks.


 

 MP3 does not use a low pass filter, it uses a model of human sound perception to selectively remove certain things under certain conditions.
   
  If you have any aspirations of a quality system why are you using low-bitrate MP3? At the very least run medium to high VBR MP3. Memory is cheap.


----------



## kingpage

MP3 compression tools use a low pass filter in many default settings. I only used that as an example. That's why I said CD (Lossless: FLAC, APE, etc), MP3 (Lossy: AAC LC, AAC HE, Ogg) and DVD (48Hz). What I wanted to know was what effects the 24bit DAC with 96Hz sampling rate will have on the 44Hz/48Hz recordings's sound quality in general regardless of compression.


----------



## bigshot

I had a high end SACD player that would upsample redbook for playback to two different higher bitrate levels. Try as I might, I could never hear any difference between when that little blue light was on and when it was off. I used to refer to it as the "placebo button".


----------



## brasled

So long as the data can be processed by the speakers, then a 24 bit higher hz amplitude recording will sound better under certain conditions.  Why?    Certain frequencies resonate with lower frequencies, causing them to to change volume.  When we listen to an acoustic guitar, we hear the fullness (resonance) of a good guitar, and we hear this change in volume of certain frequencies that occur.  It is part of the beauty we've come to appreciate when listening to analog music.  It is basic science of sound, and for some people, very noticeable.    You don't hear the frequencies directly, but you do here their influence upon the frequencies you do hear.


----------



## Head Injury

Quote: 





brasled said:


> So long as the data can be processed by the speakers, then a 24 bit higher hz amplitude recording will sound better under certain conditions.  Why?    Certain frequencies resonate with lower frequencies, causing them to to change volume.  When we listen to an acoustic guitar, we hear the fullness (resonance) of a good guitar, and we hear this change in volume of certain frequencies that occur.  It is part of the beauty we've come to appreciate when listening to analog music.  It is basic science of sound, and for some people, very noticeable.    You don't hear the frequencies directly, but you do here their influence upon the frequencies you do hear.


 

 Cool, but what exactly does that have to do with a dynamic range greater than 96 dB or frequencies we can't hear?


----------



## bigshot

Frequencies boUncing off each other isn't the reason that an acoustic guitar sounds the way it does. It's the way the sound bounces off the walls of the box inside the guitar, and the way the top vibrates. I don't know how that applies to high bitrate audio at all.


----------



## grokit

I've been operating under the impression that the main benefit of upsampling to a higher bit sample rate is that there will be no loss of resolution when attenuating the volume in the digital domain, or when doing any other DSP. Kind of a "digital headroom" thing.
   
_edited to change bit to sample_


----------



## xnor

Quote: 





grokit said:


> I've been operating under the impression that the main benefit of upsampling to a higher bit rate is that there will be no loss of resolution when attenuating the volume in the digital domain, or when doing any other DSP. Kind of a "digital headroom" thing.


 

 Upsampling has nothing to do with bit rate per se or changes of the word length. Upsampling is a process of increasing the sample rate.


----------



## grokit

I mistakenly said bit rate instead of sample rate (like from 44.1k to 88.2 or 96k), post corrected. I don't think I mentioned anything about word length.
   
  Besides the semantics, what is your view on DSP and upsampling?


----------



## stv014

Quote: 





grokit said:


> I've been operating under the impression that the main benefit of upsampling to a higher bit rate is that there will be no loss of resolution when attenuating the volume in the digital domain, or when doing any other DSP. Kind of a "digital headroom" thing.


 

 For digital volume control, you only need increased DAC resolution. Higher sample rate is useful mainly for other types of effects and synthesis, to avoid/reduce aliasing and/or interpolation artifacts.


----------



## grokit

So going from 16 bit to 24 bit would work better for digital volume, but changing the sample rate is what we want for reducing these artifacts?


----------



## killkli

True!
  Though the one who sells "higher" bits music often comes with better recording & mixing control.
  So it's still better than normal CDs.
  But if the record company's doing great job in their recording & mixing, than it's really no perceptible difference between 24bits & 16bits.
  Nor was it with 44kHz & 96kHz  .
   
  Also, I guess it's easier to achieve better recording result with vinyl, since it a simpler matter than digital.
  And that's the reason I think many people still preferred vinyl over CD.


----------



## grokit

I agree that the SACDs and vinyl can be mastered better, but unfortunately this isn't always the case!
  Vinyl vs. digital, there's a whole 'nother debate...


----------



## firev1

Quote: 





killkli said:


> Also, I guess it's easier to achieve better recording result with vinyl, since it a simpler matter than digital.
> And that's the reason I think many people still preferred vinyl over CD.


 

 This is not the case, its often because the recording engineer knows less about cd mastering vs vinyl mastering. Some of the earlier recordings made with cds had the problem where the engineer compensated for vinyl loss which is the main reason for the harshness in earlier recordings.


----------



## grokit

Don't forget about the use of compressors/limiters in the mastering process for many CD's, in order to make them sound better on one's iBuds.
   
  http://en.wikipedia.org/wiki/Loudness_war


----------



## maverickronin

Quote: 





grokit said:


> Don't forget about the use of compressors/limiters in the mastering process for many CD's, in order to make them sound better on one's iBuds.
> 
> http://en.wikipedia.org/wiki/Loudness_war


 

 I think its more about car radios than ibuds.  Or at least it started that way.  I know I can't listen to any classical at all in my car.  Its either ear piercing or I can't hear anything over the road noise.


----------



## bigshot

The reason so much compression is used is because of ipods. When you put music on shuffle play, compressed songs sound louder than wide dynamic range songs. Labels don't want their song to suddenly be quiet compared to other songs so they compress. Pretty soon it started feeding back on itself and they wanted their songs to be compressed even more so they would sound louder than compressed songs.


----------



## scottie584

I often skip loud songs on my phone, much more often than quieter ones. So I wouldn't necessarily say that more loudness and compression is more attractive to a consumer.


----------



## Griffinhart

ReplayGain exists for a reason.
   
  -- Griffinhart


----------



## scottie584

Which doesn't get rid of the original compression. It just balances out the volume.


----------



## Griffinhart

Nothing's going to get rid of the original compression short of getting your hands on the original recordings and doing your own masters. ReplayGain gets rid of the loudness.
   
  -- Griffinhart


----------



## daveDerek

Quote:


bigshot said:


> The reason so much compression is used is because of ipods. When you put music on shuffle play, compressed songs sound louder than wide dynamic range songs. Labels don't want their song to suddenly be quiet compared to other songs so they compress. Pretty soon it started feeding back on itself and they wanted their songs to be compressed even more so they would sound louder than compressed songs.


 

 actually routine overuse of compression goes back to commercial radio, well predating mp3 players, especially when played back in the relatively noisy environment of a car. this is also seen on tv, and is exaggerated during the commercials when the average volume level all of a sudden shoots up over the level of the program.


----------



## hlinssen

I thought I'd add my exasperated plea to good people like Carlos Santana and Bruce Springsteen, to name a couple:  "Please do not compress your disc releases so extremely.  I keep buying and trying, but they are totally unlistenable on my system".
     I used to use a dbxII compander to restore some dynamic range, but switching transients from my DAC zapped the left channel input, and now I have to listen to their recording as pressed...
   
  Harry Linssen


----------



## JefferyK

Deleted by author.


----------



## jaud

Quote: 





gregorio said:


> It seems to me that there is a lot of misunderstanding regarding what bit depth is and how it works in digital audio. This misunderstanding exists not only in the consumer and audiophile worlds but also in some education establishments and even some professionals. This misunderstanding comes from supposition of how digital audio works rather than how it actually works. It's easy to see in a photograph the difference between a low bit depth image and one with a higher bit depth, so it's logical to suppose that higher bit depths in audio also means better quality. This supposition is further enforced by the fact that the term 'resolution' is often applied to bit depth and obviously more resolution means higher quality. So 24bit is Hi-Rez audio and 24bit contains more data, therefore higher resolution and better quality. All completely logical supposition but I'm afraid this supposition is not entirely in line with the actual facts of how digital audio works. I'll try to explain:
> 
> When recording, an Analogue to Digital Converter (ADC) reads the incoming analogue waveform and measures it so many times a second (1*). In the case of CD there are 44,100 measurements made per second (the sampling frequency). These measurements are stored in the digital domain in the form of computer bits. The more bits we use, the more accurately we can measure the analogue waveform. This is because each bit can only store two values (0 or 1), to get more values we do the same with bits as we do in normal counting. IE. Once we get to 9, we have to add another column (the tens column) and we can keep adding columns add infinitum for 100s, 1000s, 10000s, etc. The exact same is true for bits but because we only have two values per bit (rather than 10) we need more columns, each column (or additional bit) doubles the number of values we have available. IE. 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024 .... If these numbers appear a little familiar it is because all computer technology is based on bits so these numbers crop up all over the place. In the case of 16bit we have roughly 65,000 different values available. The problem is that an analogue waveform is constantly varying. No matter how many times a second we measure the waveform or how many bits we use to store the measurement, there are always going to be errors. These errors in quantifying the value of a constantly changing waveform are called quantization errors. Quantization errors are bad, they cause distortion in the waveform when we convert back to analogue and listen to it.
> 
> ...


 

 I don't know if this has be said before (didn't want to read through all 50 odd pages of replies) but that's not entirely correct. The basic principles are correct, but the aspects of audio quality are not. 
   
  Let me explain, as we increase the bit depth we not only increase the dynamic range (which in my opinion is like a cherry on top) but we do increase the accuracy of the audio. It goes a little like this; as stated above the sample rate (measure in Hz) is the amount of times per second that the analog audio is indexed (or sampled) each sample 'remembers' (or expresses) only one thing, its amplitude which is expressed in a digital word ie binary. So a in CDDA (Compact Disc Digital Audio) we sample at 44.1kHz and we use a 16bit word length (or bit depth), each sample has a 16bit word telling the digital system what amplitude it is and weather it is positive or negative phase. What the samples are effectively doing is drawing a graph of the analog signal, just as you would join the dots to create a picture in a kids book. Are you with me so far? Alright, with a 16bit word length we can express ~65,000 different values, which means we express ~65,000 different levels of amplitude. I hear you ask, but what happens if the analog signal falls between one of the these ~65,000 values? Well, that's where 24bit world length come in, it can express over 16 million different levels of amplitude. 
   
  So as you can see as we increase the bit depth of the audio we increase the dynamic range but we also increase the accuracy of the digital waveform, if we increase the sample rate as well we also improve the accuracy of the digital audio; remember the sample rate is how many times the analog signal is being indexed, so if we increase the sample rate we get more 'snapshots' of the analog audio waveform. Oh, and another thing studios *don't* record in 48bit (I don't think that 48bit actually exists in a piece of hardware) most will record in 24bit and some will record in 32bit.
   
  Here's a pretty picture to illustrate what I'm talking about in regards to increasing the sample rate:
   

   
  And here's another pretty picture demonstrating an increase in the bit depth:
   

   
  Sources:
  Studied Sound Engineering at RMIT, Melbourne, Australia
  First image source: RMIT course content
  Second image source: http://www.echoaudio.com/Digital%20101.htm


----------



## Griffinhart

Quote: 





jaud said:


> Alright, with a 16bit word length we can express ~65,000 different values, which means we express ~65,000 different levels of amplitude. I hear you ask, but what happens if the analog signal falls between one of the these ~65,000 values? Well, that's where 24bit world length come in, it can express over 16 million different levels of amplitude.


 

 Specifically: 65536 and 16777216, respectively.
   
  32-bit gets you 4294967296 values.
   
  -- Griffinhart


----------



## emmodad

@ jaud:  in the 50+ pages of this (often technically-misinformative) thread, there have been numerous posts addressing the OP's misunderstanding (and sadly, stubborn defensiveness) wrt relationships between resolution, dynamic range, quantization error etc.  this is not the only thread on head-fi in which the OP - apparently lacking some fundamental technical understanding - has seemingly aggregated and referenced information from a variety of sources, but smoothly presents declarative conclusions of "fact" which are actually flawed; at minimum misrepresentative; or worse, false.
   
  This has been the case in posts from the OP regarding digital audio theory, digital signal processing fundamentals, audio engineering, and linear systems theory, to name only a few areas.  The OP has also previously eventually admitted that they understand neither the mathematics underlying digital audio theory, nor digital audio converter design.
   
  if you understand digital signal processing theory, you really should lean back with a good bottle of red wine and plow through this entire thread.  Makes for entertaining, and sometimes toe-curling, reading.
   
  chuck


----------



## jaud

@chuck
  I was planing on doing so but didn't have time at the time of posting, just wanted to clarify some finer details, and again sorry if im rehashing what has been previously stated in the thread, but i often find on HI-FI etc forums that most peoples knowledge on digital audio is sourced from lots of different places of varying quality. I find that most people get confused by the delicate interplay of sample rate and bit depth and what that means in terms of audio accuracy. When explain clearly (I hope that my post was clear and easy to understand, but if anyone wants some clarification or maybe some expansion please PM me or reply to this thread and I'll do my best to help 
	

	
	
		
		

		
			





 ) PCM digital audio is actually a very elegant and simple system for audio reproduction, I'm still trying to wrap my head around how DSD audio works though 
	

	
	
		
		

		
		
	


	



   
  cheers,
  jaud


----------



## xnor

jaud, an 8 bit signal will be output just as smooth as a 64 bit signal. The only thing that changes is the noise level assuming dither was used => dynamic range.
   
  What you describe as accuracy doesn't work the way you think it does. Otherwise you'd understand DSD.


----------



## stv014

Quote: 





jaud said:


> And here's another pretty picture demonstrating an increase in the bit depth:


 
   
  Although the OP does have some errors, it is correct that the "stair step" distortion on your picture is a common source of misconception, and it can be entirely eliminated by dithering (at the expense of some noise floor, hence the limited dynamic range). Here is what a dithered -90.3 dBFS sine wave looks like in 16 bit PCM format, and the result of removing the noise from the quantized signal with a comb filter. There should be no "steps" in the output of the DAC:

  This is not the same as quantizing first, and then adding the noise to "mask" the distortion, as the following picture shows:

   
  Quote: 





jaud said:


>


 
   
  The 48 kHz signal may look very different here, but that is because most of the energy of the analog input is in the ultrasonic range. So, what is removed is neither audible, nor typically present in music in large amounts. The audible frequency range is the same in all cases.


----------



## D. Lundberg

Quote: 





jaud said:


> What the samples are effectively doing is drawing a graph of the analog signal, just as you would join the dots to create a picture in a kids book. Are you with me so far?


 
  Huh? Waveforms are not reconstructed by drawing straight lines between samples. Where did you get that idea?
   
  Quote: 





> Alright, with a 16bit word length we can express ~65,000 different values, which means we express ~65,000 different levels of amplitude. I hear you ask, but what happens if the analog signal falls between one of the these ~65,000 values?


 
  It will get quantized (rounded) to the closest value. The errors from quantization will (in a properly dithered system) represent noise at a level determined by the word length.
    
  Quote:


> So as you can see as we increase the bit depth of the audio we increase the dynamic range but we also increase the accuracy of the digital waveform, if we increase the sample rate as well we also improve the accuracy of the digital audio; remember the sample rate is how many times the analog signal is being indexed, so if we increase the sample rate we get more 'snapshots' of the analog audio waveform.


 
  The bit depth *is* the accuracy (of quantization) and that accuracy determines the dynamic range and SNR.
  The sample rate determines the bandwidth, so a higher sample rate means you can represent higher frequencies.


----------



## jaud

Quote: 





d. lundberg said:


> Huh? Waveforms are not reconstructed by drawing straight lines between samples. Where did you get that idea?
> 
> It will get quantized (rounded) to the closest value. The errors from quantization will (in a properly dithered system) represent noise at a level determined by the word length.
> The bit depth *is* the accuracy (of quantization) and that accuracy determines the dynamic range and SNR.
> The sample rate determines the bandwidth, so a higher sample rate means you can represent higher frequencies.


 


  No waveforms are not reconstructed by drawing strait lines between samples, I'm using that as an example to illustrate what samples in PCM digital audio are doing, hence the use of the word "effectively", i feel that by using this example (and I've found that most people understand that when I say this I'm not talking literally but figuratively) that it is easier to understand role sample rate plays in PCM audio. Let  me put it another way, "samples only remember two things, their amplitude and their phase, each sample is effectively taking a snap shot of the analog signal at a certain point in time, then when played back in the same window of time a group of samples will 'render' a version of the analog signal"
   
  Yes, the value will get quantized if it fall between level of amplitude that can be captured by the digital system, but that is not a true representation of the amplitude.
   
  That's true, but with more (to use a similar wording as my example above) sample per second we gain more "snapshots" of the analog signal and can there for 'track' or 'render' more minute changes in the amplitude of the analog signal, pair that with a higher bit depth (which means we can record many more different levels of amplitude, with less distortion of the original analog signal when using quantization)


----------



## jaud

Quote: 





stv014 said:


> Although the OP does have some errors, it is correct that the "stair step" distortion on your picture is a common source of misconception, and it can be entirely eliminated by dithering (at the expense of some noise floor, hence the limited dynamic range). Here is what a dithered -90.3 dBFS sine wave looks like in 16 bit PCM format, and the result of removing the noise from the quantized signal with a comb filter. There should be no "steps" in the output of the DAC:
> 
> This is not the same as quantizing first, and then adding the noise to "mask" the distortion, as the following picture shows:
> 
> ...


 

 Yes, the stair step distortion picture isn't' the best example, but what it does show is the raw uncorrected performance improvements which can be obtained by recording and consequently reproducing PCM digital audio at higher bit depths.
   
  Also the images that you have used are working on a completely different time scale, with the lack of a scale indication, I would presume that these images represent a sine wave over a much long period of time than the image that i used, which I would presume was is a representation at a much smaller time frame, I emphasise the use of the word presume.


----------



## D. Lundberg

Quote: 





> Originally Posted by *jaud* /img/forum/go_quote.gif
> No waveforms are not reconstructed by drawing strait lines between samples, I'm using that as an example to illustrate what samples in PCM digital audio are doing, hence the use of the word "effectively", i feel that by using this example (and I've found that most people understand that when I say this I'm not talking literally but figuratively) that it is easier to understand role sample rate plays in PCM audio. Let  me put it another way, "samples only remember two things, their amplitude and their phase, each sample is effectively taking a snap shot of the analog signal at a certain point in time, then when played back in the same window of time a group of samples will 'render' a version of the analog signal"


 
  Alright, but the kids book analogy is a bit misleading since those books *are* about drawing straight lines between points and not about summing sinc-pulses (that would be a strange book 
	

	
	
		
		

		
		
	


	




) like digital audio.
  If you want people to understand digital audio you'll need to stop using that example and focus on explaining Nyquist/Shannon and Fourier.
   
  Quote: 





> Yes, the value will get quantized if it fall between level of amplitude that can be captured by the digital system, but that is not a true representation of the amplitude.


 
  It *is* a true representation of the amplitude, and it will be accurately represented whether you use 2 or 32 bit quantization. That's the beauty of the Nyquist theorem (and dither).
  The difference is in the amount of background noise you'll get and consequently the dynamic range.
   
  How do you think DSD or delta/sigma-converters would work if you couldn't get the correct amplitude using just a few bits?
   
  Here is a real example to show you that it actually works:
   
  A sinewave, 1kHz @ -0.9dB, 24 bit.

   
  Now what happens to the sinewave if you reduce the bit depth to 8 bit?
  You'll get the same 1kHz @ -0.9dB sinewave with the addition of quite a bit of background noise (from quantization):
   

   
  Quote: 





> That's true, but with more (to use a similar wording as my example above) sample per second we gain more "snapshots" of the analog signal and can there for 'track' or 'render' more minute changes in the amplitude of the analog signal, pair that with a higher bit depth (which means we can record many more different levels of amplitude, with less distortion of the original analog signal when using quantization)


 
  A waveform has four basic characteristics: frequency, amplitude, phase (relative to other waveforms) and dynamic range.
  If the sample rate is more than twice as fast as the highest frequency the signal contains, then you'll know three of those. The bit depth will give you the fourth one: the dynamic range.
   
  The real limits imposed by the sample rate and bit depth are in bandwidth and dynamic range. If the signal is within those constrains it can (if the implementation allows it) be accurately represented and reconstructed.


----------



## estreeter

I view this thread along the same lines as I view similar debates on jitter, and it largely comes down to this:
   
  - what would it cost me to buy the 'optimum' solution ?
   
  - assuming I could afford said solution, would these old ears even benefit from all that technology ?
   
  No, I'm not a Luddite and I dont have cloth ears, but I do realise how skilful many DAC makers are in the time-honored technique of baffling us with BS. I know Head-Fi is a 'bigger is automatically better' cave, but if something sounds good to me at 24/96 (or, gasp, 16/44.1 ...), I'm not going to spend 2K on a DAC simply because the numbers excite the tech geeks among us.


----------



## jaud

Quote: 





estreeter said:


> I view this thread along the same lines as I view similar debates on jitter, and it largely comes down to this:
> 
> - what would it cost me to buy the 'optimum' solution ?
> 
> ...


 
  Although my previous posts in this thread probably dint show it, but when I am researching and demoing gear etc. this for me is the ultimate decider on the purchase because i feel that the pursuit of Hi-Fi and audiophilia is the pursuit of gain pleasure form recorded music, not the pleasure of know that you have the best or most expensive gear/rig


----------



## estreeter

I would add the fascination many Head-Fiers have with asynchronous USB DACs over any other implementation, regardless of the reputation many of the older DACs had prior to Gordon Rankin sat down to write his asynch code. If it works, fantastic, but getting hung up on buzzwords for their own sake is just crazy, IMO.


----------



## khaos974

@jaud: An interesting experiment to show that* bit depth translates into noise level (which is another way to say dynamic range) rather than accuracy*, here is a* 4-bit file* with dither, and you should notice that the fact that it's 4 bit isn't really heard as inaccuraccy but a very high noise level.
   
  And before someone asks, it shows up as an 8-bit file, but it really is 4-bit,* the samples only take 16 values*, 0, 16, 32, 48, 64..., none of the intermediate values are used.
   
Download Extract.zip - 5.9 Mb
   
  PS: I cheated a little, some of the noise is far outside the audible range.


----------



## milosz

Stairstep waves?
   
  If you were listening to a signal recorded in the four least significant bits of a 16 bit recording, it would be VERY low in level; probably inaudible. My listening room has a noise floor about 70 dB below my max listening level.  4 LSB's would be a signal at -72 dB.  This is below the noise floor of my listening room, so really if sounds recorded at 4 bit levels are full of artifact, well, I will barely be able to hear the fundamental of a sound that is below my ambient noise level, and those harmonics caused by the artifact will be at least 6 dB below THAT- geez, you know, I'm just not going to worry about artifacts that are nearly 10 dB below the sound of my computer fan in the other room.
   
  I mean- just think about it.  If you have adjusted your listening level to be pretty LOUD- that would be 100 dB peak levels. These artifacts induced by low-bit-levels are going to most be right at or well below your ambient noise floor.  (OK, if you are listening at ROCK CONCERT levels of 115 dB or higher, _your ears are shot_ anyway and you're not going to be able to hear most of this high-end super-fi hi-rez stuff.)
   
  See http://www.engineeringtoolbox.com/nc-noise-criterion-d_725.html  background noise levels in residential settings are 30~45 dBA, so if you are listening at 100 dBA the signal-to-noise level is 55 dB to 70 dB.   Now tell me a signal at -80 dB with artifact harmonics 10 dB below THAT is going to really matter (the - 90 dB unwanted harmonics would be 10 dBA if you are listening at 100 dBA, or *AT LEAST 20 dB BELOW THE NOISE FLOOR OF A VERY QUIET HOME*)
   
  OK, so let's forget about recordings of music using the 4 LSB. Let's talk about real recordings, where we are someplace between maximum (0 dB) and maybe 60 dB down.
   
  After the signal comes out of the DAC process it is indeed "jumping" from one level to another instead of a smooth transition.  But now this "jaggy" signal train goes into a LOW PASS FILTER and part of this filter's job is performing integration over a specified timescale, factored by appropriately chosen coefficients- so when the signal comes out of the filter, those "jaggy" artifacts should mostly be gone, and if the sampling frequency is sufficiently high to allow employing a low pass filter that allows the entire audible spectrum to pass through, well, we have a smooth waveform as long as we started with a fair number of bits to begin with. Most of the content in recorded music is using_* at least*_ 8~10 of the available 16 bits, and after the low pass filter you're really not going to have much 'stairstep' artifact in there, the percentage of stairstep will be a ratio of least-significant-bit to number of bits in the signal, so for all but the quietest passages the artifact is way down there in % terms.  
   
  When we try to record a sine wave using just the LSB or maybe the two least significant bits, we are more likely to see quite some amount of artifact in the output of the "brickwall" filter, but the signal is  will be ~90 dB down (2 bits out of 16  anyway so a little extra harmonic content resulting from this artifact-laden waveform is going to be present, but that harmonic content will be lower than the level of the fundamental, those unwanted harmonics will be something like -100 dB I'd guestimate (ref to 0 dB, not ref to the fundamental)- so, even though a 24 bit signal has much less artifact in it's -90 dB presentation of waveforms, can you really HEAR that?  I mean, 90 dB down from full level is certainly well at least 20 dB below the level of ambient noise in my home- and unless you live in an anechoic chamber, it probably is in yours, too. 
   
  ( AN ASIDE: People are always talking about how great analog sounds in comparison to digital, because there's no digital artifact-  but if you can show me a vinyl system that actually has even a 60 dB s/n ratio from the ACTUAL LP SURFACE -not just the electronic s/n ratio of the phono stage- - well, I'd be impressed.  My own measurements show me that most LPs are lucky to offer 50 dB of actual s/n ratio.  Tape  can be a little better, but unless you are listening to a master tape.... well we could go on here but let's not.)
   
  So, yeah, a 24 bit recording is going to have measurably less artifact at very low levels. But at actual music recording levels? Gee, not so much. And at actual in-room listening levels?  Not much at all, almost none.
   
  And while all this engineering analysis is useful, we're in fact talking about human perception here, not just noise levels and harmonic power analysis. And when it comes to humans using their ears to judge sound, I don't think there has been *ANY* A/B/X study showing that anyone- from golden eared audiophiles to musicians to regular folks- who can reliably tell what is 24/96 and what is 16/44.1.  You could probably devise some kind of test tone that might make the difference apparent, but on music signals [size=10pt]I don't believe anyone has been found that can beat chance. [/size][size=10pt]Of course, just because no positive results for A/B/X 24 vs 16 bit identification testing has does not PROVE that *no one* can really hear the difference- you have to test EVERY HUMAN BEING to make that claim.  However, if a large sample of people including those with advanced training in terms of critical listening skills such as musicians and audiophiles has been shown to be unable to pick the hi-rez file in a blind test, I for one am willing to say that's "close enough for practical purposes" and will take it to mean that in all probability_* I*_ can't hear the difference.  And that's all that matters to me-  I'm the one who buys gear and recorded music for me, and if I can't hear the difference between redbook and 24/96 recordings, well then I see no need for me to spend my hard-earned cash on exotic DACs or hard-to-find 24/96 recordings. [/size]


----------



## estreeter

Quote: 





milosz said:


> [size=10pt]And that's all that matters to me-  I'm the one who buys gear and recorded music for me, and if I can't hear the difference between redbook and 24/96 recordings, well then I see no need for me to spend my hard-earned cash on exotic DACs or hard-to-find 24/96 recordings. [/size]


 

 I don't believe I'm quoting you out of context in grabbing your final couple of sentences - apologies if that is the case but this thread is a monster already. I agree with you 100% - good enough is most definitely *good enough*. If folk want to collect the remaining hi-rez discs and pay for downloads, all power to them, but most of us seem to have survived with nothing more than Redbook CDs for many years. I;d be happier if the recordings themselves were better in many cases, but I have absolutely no control over that. Inebitably, there will be people out there who claim they can tell the difference between Redbook and 24/96, but I'm not one of them. Choice is good, but that has to go both ways - religious wars rarely allow for that kind of leeway.


----------



## joe_cool

Quote: 





> Originally Posted by *jaud* /img/forum/go_quote.gif
> 
> 
> And here's another pretty picture demonstrating an increase in the bit depth:


 
   
  This is a misrepresentation of the differences in bit-depth as it relates to digital audio. The extra bits are not equally spread over the visible wave-form, they are centered around the zero-crossing area and are therefore practically invisible as well as inaudible. This sort of cognitive error along with common but misleading terms like "snapshots" lead to misunderstanding.


----------



## xnor

Quote: 





joe_cool said:


> The extra bits are not equally spread over the visible wave-form, they are centered around the zero-crossing area and are therefore practically invisible as well as inaudible.


 
  Huh?


----------



## ILikeMusic

Quote: 





xnor said:


> This is a misrepresentation of the differences in bit-depth as it relates to digital audio. The extra bits are not equally spread over the visible wave-form, they are centered around the zero-crossing area and are therefore practically invisible as well as inaudible. This sort of cognitive error along with common but misleading terms like "snapshots" lead to misunderstanding.


 
  Quote: 





xnor said:


> Huh?


 

  
  And there is the crux of almost all 192/24 vs. 44/16 threads...


----------



## Kees

Quote: 





ilikemusic said:


> And there is the crux of almost all 192/24 vs. 44/16 threads...


 

 Yes,this. 
	

	
	
		
		

		
		
	


	




.
  Only very few people here know what they are talking about.


----------



## firev1

Quote: 





jaud said:


> Here's a pretty picture to illustrate what I'm talking about in regards to increasing the sample rate:


 
   
  Since the noise question has been already been beaten to the death MULTIPLE times, I shall not go there. But it seems the sample rate one has yet "die" enough times so I shall go .
   
  Going by the sample rate impulse graph shown, 192khz seems like a terrible choice for a recording format. 24/96 combines the best of 2 worlds, DSD is good and all but its noise characteristics by heavy noise shaping makes it sucky for recording but great for playback or format delivery(mastering). 192khz though it reaches close to the impulse response of the analog sound, there is significant "spread" or ringing which to me, may be more audible than the maximum amplitude of the impulse response.


----------



## stv014

Quote: 





> Originally Posted by *firev1* /img/forum/go_quote.gif
> 
> 192khz though it reaches close to the impulse response of the analog sound, there is significant "spread" or ringing which to me, may be more audible than the maximum amplitude of the impulse response.


 

 The ringing is at 96 kHz, though, so I do not think it is audible. Also, by implementing a smooth roll-off already below the maximum frequency (e.g. from 24 kHz to 96 kHz), the ringing can be made much shorter (its length is inversely proportional to the bandwidth of the lowpass filter).


----------



## firev1

I see, but fact still remains that with the extremely high sample rates, we lose some accuracy, simply put, I don't think impulse response tell the whole picture.


----------



## stv014

In theory, no accuracy is lost with a higher sample rate. It just allows you to reproduce a wider frequency range, but it is up to you how the extra bandwidth is used. Here are some impulse and frequency response graphs for comparison:
   
  48 kHz with fast roll-off:
    
  48 kHz with slow roll-off above 20 kHz:
    
  96 kHz with slow roll-off above 24 kHz:
    
   
  The 96 kHz version has ruler flat frequency response (and also phase response as it is linear phase) up to 24 kHz, so nothing is lost there compared to the 48 kHz sample rate. Due to the increased bandwidth of the lowpass filter, it has far less ringing.
  A higher sample rate is technically better (assuming that it does not introduce problems related to non-ideal hardware), but of course it may not actually make an audible difference compared to 48 or even 44.1 kHz, so the extra bandwidth is still likely to be a waste of space in practice.


----------



## jcx

the fact is that with Analog Signals we are starting with limited "bits" - all real world signals have limited bandwidth, a noise floor (Johnson "thermal" noise and often other types)
   
  EarthWorks makes much in their marketing of their products super bandwidth - the fastest mic they sell for recording use has a 50 kHz 2nd order roll off, and due to its small size it has higher noise floor
   
  most mics used in recording studios have 20-25 kHz or less bandwidth
   
  plug the numbers into Shannon-Hartley Channel Capacity Theorem and good 20 bit (enob) 96/192 k ADC aren't missing much
   
   
  any idea of the best ever delivered by any analog mass market consumer music recording media - hard numbers?


----------



## firev1

My comment on accuracy mainly goes for 192khz sample rate, with 96khz there is almost no problem(at least for A/DC). Unless I misinterpret Lavry's paper incorrectly?


----------



## stv014

Quote: 





firev1 said:


> My comment on accuracy mainly goes for 192khz sample rate, with 96khz there is almost no problem(at least for A/DC). Unless I misinterpret Lavry's paper incorrectly?


 

 There is nothing inherently (i.e. not related to its current implementations) wrong with the 192 kHz sample rate either, although for music storage/distribution there is not much point using it.


----------



## skibum

Does anyone know where I can get a test track that is in 192KHz / 16 bit, 44.1KHz / 24 bit and 192kHz / 24 bit?
   
  I realize I can do this myself with dBpoweramp, but my understanding is that most software out there just truncates the extra bits off.  The process requires dithering and needs to be done properly for a fair comparison.
   
  I would like to play these tracks myself and see what I can actually hear.
   
  ?


----------



## stv014

If you have a link to a good 192/24 or 96/24 test file that can be freely used, I can create a blind test thread for the comparisons, including files that go through actual D/A-A/D conversion if that is relevant.


----------



## khaos974

-


----------



## Locknar

Quote: 





estreeter said:


> No, I'm not a Luddite and I dont have cloth ears, but I do realise how skilful many DAC makers are in the time-honored technique of baffling us with BS. I know Head-Fi is a 'bigger is automatically better' cave, but if something sounds good to me at 24/96 (or, gasp, 16/44.1 ...), I'm not going to spend 2K on a DAC simply because the numbers excite the tech geeks among us.


 
   
  Ha Ha! Hear, Hear!





  J


----------



## kiteki

gregorio said:
			
		

> (post #001) /img/forum/go_quote.gif
> 
> So, if you accept the facts, why does 24bit audio even exist, what's the point of it? There are some useful application for 24bit when recording and mixing music. In fact, when mixing it's pretty much the norm now to use 48bit resolution. The reason it's useful is due to summing artefacts, multiple processing in series and mainly headroom. In other words, 24bit is very useful when recording and mixing but pointless for playback. Remember, even a recording with 60dB dynamic range is only using 10bits of data, the other 6bits on a CD are just noise. So, the difference in the real world between 16bit and 24bit is an extra 8bits of noise.


 
   
   
  OK, so the thread starter says 24 bit is superior and very useful for recording.
   
  Let's see, it takes up one third (1/3) extra disk space over 16 bit.  Last time I checked, a 2 TB external hard-drive now costs a little over $100 at my local supermarket, last time I checked... you can fit 70 CD's on a dual-layer blu-ray disc.
   
  Keep it in 24 bit for playback too, then you don't need to discuss all this theory.
   
   
  It's true 16/44.1 is very very satisfactory, it's not true it's the ultimate.


----------



## stv014

Quote: 





> Originally Posted by *kiteki* /img/forum/go_quote.gif
> 
> Let's see, it takes up one third (1/3) extra disk space over 16 bit.


 
   
  With FLAC or other lossless compressed formats, it is actually more, because the additional 8 bits per sample contain mainly noise, and compress poorly.


----------



## grokit

I just downloaded a copy of the new Norah Jones Little Broken Hearts from HDtracks:

"Mastered at New York’s legendary Sterling Sound by Norah Jones’ longtime mastering engineer Greg Calbi, Little Broken Hearts is available in master-quality 24-bit audio from the original flat mix sources, the highest quality format available. Little Broken Hearts (*44.1kHz/24bit*)


----------



## kiteki

When I read the first post in this thread a second or third time, it makes it sound like 16 bit is unecessary and we should downsample recordings to 12, 8 or 4 bit.
   
  ...if the only difference is unecessary dynamic range since the most we'll ever listen to is ~60dB. (?)


----------



## khaos974

kiteki said:


> When I read the first post in this thread a second or third time, it makes it sound like 16 bit is unecessary and we should downsample recordings to 12, 8 or 4 bit.
> 
> ...if the only difference is unecessary dynamic range since the most we'll ever listen to is ~60dB. (?)




Background noise would be audible if we reduced bit depth, I have ABX'ed successfully 16 and 14 bit files.
(same origin, volume and time aligned)


----------



## xnor

@kiteki: See LossyWAV/FLAC for reduced bit depth where it most probably cannot be heard.


----------



## Baxide

Quote: 





khaos974 said:


> Background noise would be audible if we reduced bit depth, I have ABX'ed successfully 16 and 14 bit files.
> (same origin, volume and time aligned)


 

 Are you sure? My first CD player was a 12 bit design from about 1984. My next one was a 14 bit unit, which I then replaced with a 14 bit/4x oversampling Marantz. The 12 bit CDP sounded quite good actually as did the 14 bit one. I think one was a Fisher and the other a Hitachi.


----------



## khaos974

baxide said:


> khaos974 said:
> 
> 
> > Background noise would be audible if we reduced bit depth, I have ABX'ed successfully 16 and 14 bit files.
> ...




I didn't say it sounded bad, merely that I ABX'ed it,
Also, most delta sigma dac operate only on a depth of a few bit, using oversampling to reject noise out of the audible band, I'm not familiar with the CDP you talked about but how a DAC operates has little to do with the bit depth of the recording.


----------



## maverickronin

Quote: 





stv014 said:


> With FLAC or other lossless compressed formats, it is actually more, because the additional 8 bits per sample contain mainly noise, and compress poorly.


 
   
  That should be the proof right there that those extra bits only serve to more precisely define the noise floor.
   
  Then you realize that if they reject Shannon-Nyquist sampling theorem out of hand then they're likely to reject Shannon entropy too...


----------



## bigshot

kiteki said:


> Keep it in 24 bit for playback too, then you don't need to discuss all this [COLOR=0000FF]theory.[/COLOR] It's [COLOR=0000FF]true[/COLOR] 16/44.1 is very very satisfactory, it's [COLOR=800080]not true[/COLOR] it's the ultimate.




I like to mail letters in refrigerator boxes just to make sure they don't get any creases in the mail.


----------



## compoopers

Well, after reading this thread and many others, I've joined the camp that "24/192" stuff is unnecessary for listening to music. Thanks guys it was very informative.


----------



## joe_cool

Quote: 





grokit said:


> I just downloaded a copy of the new Norah Jones Little Broken Hearts from HDtracks:
> "Mastered at New York’s legendary Sterling Sound by Norah Jones’ longtime mastering engineer Greg Calbi, Little Broken Hearts is available in master-quality 24-bit audio from the original flat mix sources, the highest quality format available. Little Broken Hearts (*44.1kHz/24bit*)


 

 and......
  How about a review or some opinions?
  (I'm a big fan of Norah Jones. I have the SACD of her first album and look forward to more.)


----------



## grokit

It's very good, at least for my tastes. I sometimes wonder about her direction and influences but not here, this seems to be all Norah. It's a pop fusion of blues jazz & rock so not for the purist, but right in her creative wheelhouse IMHO. The SQ is quite good.
   
  Thanks for reminding me to go back and listen again. I need to get a file converter going because it takes a "special procedure" to play FLAC ATM with iTunes/Pure Music. I probably have one downloaded already, just haven't used it yet.


----------



## Cszcot

Wow, this is cool, thanks, always confused about bits, since I can't really tell the difference


----------



## priest

And I thought the Headphones forum made me feel stupid. Anyone know any good particle physics or cosmology forums?


----------



## Sandman65

Quote: 





bigshot said:


> I like to mail letters in refrigerator boxes just to make sure they don't get any creases in the mail.


 
   
   
  Thanks for Vernor's through my nose!
   
  Once again another highly informative thread illustrates how evil marketing types are trying to get you to reinvest in your music library.
   
   
  listening to Rued Langgaard's "Music of the Spheres"


----------



## spyrusthevirus

you know 50+ pages is a bit much for tonight, so I wont read it all.
  but here is what I am thinking.
  Let's suppose we want to sample a sine wave at 15khz. no problem, we can hear tha and mics can hear it too. so we will use 44.1khz sample rate.
  a 15khz sine is repeated 15k times in a second and we will take 44.1k samples in one second. which means that we will have 2.94 samples per period. lets make that 3.
  now if you have seen a sine wave, how is that remotely accurate?
  if we try that at 8khz, with the thought that no instrument makes a sound with a main frequency above that, it leaves us with 5.5 samples per period. It does not seem that accurate either.
  Just because the rule of thumb says that you use double the sampling rate of the maximum frequency you want to record, it does not mean the 96k and 192k are completely useless for say 15-20khz. if my basics in this are correct, you do capture more detail with a higher sample rate, the ideal being an infinite one.


----------



## xnor

Quote: 





spyrusthevirus said:


> but here is what I am thinking.
> Let's suppose we want to sample a sine wave at 15khz. no problem, we can hear tha and mics can hear it too. so we will use 44.1khz sample rate.
> a 15khz sine is repeated 15k times in a second and we will take 44.1k samples in one second. which means that we will have 2.94 samples per period. lets make that 3.
> now if you have seen a sine wave, how is that remotely accurate?
> ...


 
  The sine can - in theory - be perfectly reconstructed as long as there are > 2 samples. Since the lowpass filter in a DAC needs some room to work we cannot go up to 22.05 kHz, but somewhere around up to 21 kHz.
  Thinking in straight lines between sample points is a common fallacy, that's not how it works.
   
  No, your basics are wrong and a higher sample rate will not capture more detail in a given range of frequencies (say 0 to 21 kHz).


----------



## spyrusthevirus

Quote: 





xnor said:


> No, your basics are wrong and a higher sample rate will not capture more detail in a given range of frequencies (say 0 to 21 kHz).


 
  why does that happen? because a lower sample rate can already catch enough to reproduce those frequencies that interest us? of because of the way digitalising audio works?


----------



## xnor

Because > 2 samples are enough due to the sampling theorem.
   
  Clean 21 kHz sine wave (red dots are the samples) sampled at 44.1 kHz will be reconstructed to the blue line:

   
  This is *NOT* how it works:

   
   
   
  edit: better images


----------



## ctoth666

This thread inspired me to perform some very impromptu sonic experiments to determine whether or not I could tell a difference between bit depths. I have previously determined that I can hear the difference between 192khz and 44.1khz sample rates. Now were these blind experiments? No they were not, but that does not change the fact there IS an audible difference. My equipment includes a MacBook Pro 2011, an Echo AudioFire2 firewire interface, and modded D2000 cans. Also 21 year old ears.
   
  Here's what very unscientifically went down:
   
  -I set the playback sample rate on my interface to 96khz.
  -I recorded single C4 oscillating tone with Sylenth VST.
  -I exported three separate normalized .wav files at 16, 24, and 32 bit. All were sampled at 192khz.
  -I played back each of these recording in iTunes.
   
  I did the same experiment with the sample rate as the independent variable, and absolutely there is a clear difference at higher sample rates. I have played around with the playback sample rate on my interface with various file formats, and can discern a difference between 44.1khz and 96khz, and .wav and mp3, etc. The difference is not night and day, but still apparent on my equipment especially when I use fx processing in Ableton. Now, as for the different bit rates, I again can pick up subtle differences between 16 and 24 AND 24 and 32 bit. Yes, 32 bit does sound different on my equipment, and I stress "different" because it's not necessarily better and I don't what to attribute those differences to. Is it enough of a difference that I care? No it's not, except that now I know I can hear a difference so I'll have to get over that fact somehwere down the line.


----------



## bigshot

Try AAC 256 VBR


----------



## xnor

ctoth666, how did you export the files, with or without dither? Some sort of noise shaping?
   
  Why did you set your interface to 96 kHz but generated 192 kHz files? Why didn't you do an ABX test?
   
  What interface are you using and what audio API (ASIO, WASAPI, DirectSound ..)?
   
  What effects are you talking about? They might cause aliasing at lower sample rates, but applying the effects at higher sample rates/bit depth and doing a final conversion to 44.1/16 shouldn't sound any different. (The effect should do this upsampling automatically, else I'd say it's broken.)
   
  Why didn't you use a piece of music for the test?


----------



## stv014

Quote: 





ctoth666 said:


> Now were these blind experiments? No they were not,


 
   
  That alone prevents the results from being particularly useful. However, if you uploaded the files somewhere, and posted a link, it could be found out if there are any differences that are not the result of different sample rates and/or resolutions.


----------



## ctoth666

Quote: 





xnor said:


> ctoth666, how did you export the files, with or without dither? Some sort of noise shaping?
> 
> Why did you set your interface to 96 kHz but generated 192 kHz files? Why didn't you do an ABX test?
> 
> ...


 
  My knowledge of these things is very limited, but the only variable that changed was the bit rate. I exported the files all directly from Ableton Live with Triangular dither and did NOT do a final conversion to 44.1/16 if that's what you're asking. I set my interface to 96khz sampling but exported 192khz from Ableton because I figured it would contain the most data, but again all three files were 192khz. I do not know what an ABX is nor do I know how to perform one. The API was Core Audio and the audio interface was an Echo Audiofire2. The effects that I noted were very minor but not aliasing I don't think. Since my technical knowledge is so limited, I'll try to explain as best I can.
   
  -From 16 bit to 24 bit, the sound is brighter and there is more "oscillation" in the synth sound, like the tone has a different rhythm. On my headphones, higher frequencies are more discernible.
  -From 24 bit to 32 bit, there is one particular frequency sin wave that just sounds different if not more pronounced. When playing back the sample, I can hone in and "track" it with my ears. There is also a higher frequency oscillating tone that stands out more.
   
  That's about as well as I can describe the differences. And I didn't use a recording because I don't really have any recording equipment, and I didn't use any pre-sampled or pre-recorded music because I figured that it would defeat the purpose.


----------



## bigshot

ctoth666 said:


> From 16 bit to 24 bit, the sound is brighter and there is more "oscillation" in the synth sound, like the tone has a different rhythm.




I've run into that with weird down sampling rate mismatches. I'd bet your software has a setting that would do it correctly. Play around with it and see.


----------



## xnor

Afaik, CoreAudio resamples just like DirectSound to whatever is configured. ctoth666, if you want you can send me your test signal(s) in the highest possible format (192/32) and I'll convert them all to 96 kHz but one file will only contain frequencies up to 22.05 kHz (44.1 kHz sample rate), another one up to 24 kHz (48 kHz sample rate) and another one up to 48 kHz (96 kHz sample rate). If you don't have a spectrum analyzer enabled you can then try to pick out which is which "blind".


----------



## ctoth666

Quote: 





xnor said:


> Afaik, CoreAudio resamples just like DirectSound to whatever is configured. ctoth666, if you want you can send me your test signal(s) in the highest possible format (192/32) and I'll convert them all to 96 kHz but one file will only contain frequencies up to 22.05 kHz (44.1 kHz sample rate), another one up to 24 kHz (48 kHz sample rate) and another one up to 48 kHz (96 kHz sample rate). If you don't have a spectrum analyzer enabled you can then try to pick out which is which "blind".


 
   
  Well it was a very straightforward test. If you have the free Synth1 plugin then you can reproduce the tone that I exported from Ableton, but I suppose any plugin would do. I would like to learn something about audio sampling and bit rates from this. I don't quite understand: if I send you the test signal at 192/32, is your plan to downsample them and then upsample them to 96 kHz? I played back all of the recordings @ 96 khz through my headphones, but the actually files were all exported at the different bit rates and thus were different sizes. When I played them back, they were all resampled to 96 kHz, correct? All that I have demonstrated to myself is simply this: that there are sonic differences in the files that I exported. Again, I don't necessarily know why, but there are differences. I'm not saying that there is definitely an audible difference between 16 bit and 24 bit audio, but rather that I can hear a difference between the files that I exported.


----------



## xnor

Quote: 





> Originally Posted by *ctoth666* /img/forum/go_quote.gif
> 
> Well it was a very straightforward test. If you have the free Synth1 plugin then you can reproduce the tone that I exported from Ableton, but I suppose any plugin would do. I would like to learn something about audio sampling and bit rates from this. I don't quite understand: if I send you the test signal at 192/32, is your plan to downsample them and then upsample them to 96 kHz? I played back all of the recordings @ 96 khz through my headphones, but the actually files were all exported at the different bit rates and thus were different sizes. When I played them back, they were all resampled to 96 kHz, correct?


 
  Yes, I'd make sure the files look the same in terms of bitrate but actually only contain 44.1/16 content. That way you could play all the files without resampling from CoreAudio and wouldn't know which is which.
   
   
  Quote: 





> All that I have demonstrated to myself is simply this: that there are sonic differences in the files that I exported. Again, I don't necessarily know why, but there are differences. I'm not saying that there is definitely an audible difference between 16 bit and 24 bit audio, but rather that I can hear a difference between the files that I exported.


 
  I would be more careful with the conclusion. Do a proper ABX test (there should be apps for this available on OS X) and post the log here. Also, generate 96 kHz files if you set CoreAudio to 96 kHz. You have to eliminate all the extra variables...
   
  Btw, you also said you can hear a difference between 44.1 and 192 kHz. What was the test file for that? I'd be very surprised if you can hear above 21 or even 20 kHz.


----------



## ctoth666

Quote: 





xnor said:


> Yes, I'd make sure the files look the same in terms of bitrate but actually only contain 44.1/16 content. That way you could play all the files without resampling from CoreAudio and wouldn't know which is which.
> 
> 
> I would be more careful with the conclusion. Do a proper ABX test (there should be apps for this available on OS X) and post the log here. Also, generate 96 kHz files if you set CoreAudio to 96 kHz. You have to eliminate all the extra variables...
> ...


 
   
  All of the test files were identical. I will try a proper ABX test, but after recording some audio through the line input and doing similar "tests" I may have come to another conclusion, although I haven't verified it. My thinking is that digital audio recording is different from analog recording, and therefore maybe there really are differences in the files? To illustrate, let's say I generate a tone on a VST plugin in my DAW. I am creating that tone digitally and not sampling anything, and therefore when I export it at different sample rates and bit depths, the tone itself is actually different. If I export at 192khz, I'm actually creating twice as much data than at 96khz. This makes sense to me unless I have poorly explained it. With analog sampling there is no creation of data, but sampling of from an existing audio source whereas the digital tone does not "exist" before I export it. That was sufficiently confusing.


----------



## stv014

Why don't you just post the files, so that others can have a chance of finding out if there is any difference, rather than having to guess ?


----------



## Currawong

Quote: 





xnor said:


> Because > 2 samples are enough due to the sampling theorem.
> 
> Clean 21 kHz sine wave (red dots are the samples) sampled at 44.1 kHz will be reconstructed to the blue line:
> 
> ...


 
   
  The latter is not far off what you get if using a non-oversampling DAC. Sampling theory unfortunately doesn't match the reality perfectly. Even if the benefits are questionable, I don't really blame anyone for buying high-res files if they have a genuinely high-end system, as at least they will push both the dithering noise and other problems that result from reconstruction very far out of audibility. More practically, there is benefit when using high-res files with non-oversampling DACs. That removes two more steps between a digital master and the playback: The down-sampling to CD quality and the oversampling circuitry in the DAC. The benefits, of course, are debatable, as always.


----------



## xnor

Quote: 





currawong said:


> The latter is not far off what you get if using a non-oversampling DAC.


 
  Non-oversampling DACs fill a weird niche that lost its touch with reality, I wouldn't consider them high fidelity. The result is actually worse because those DACs use zero order hold so what you'll get is staircase-like.
   
  Quote: 





> Sampling theory unfortunately doesn't match the reality perfectly.


 
  Neither do we have ideal voltage sources, ideal op-amps, ideal anything. That doesn't mean it doesn't work, it works perfectly fine in reality.
   
  Quote: 





> Even if the benefits are questionable, I don't really blame anyone for buying high-res files if they have a genuinely high-end system, as at least they will push both the dithering noise and other problems that result from reconstruction very far out of audibility.


 
  I also don't blame anyone buying high-res files. Some might be mastered better than their regular CD counterparts, but the format itself doesn't really have advantages over CD audio for playback. Dithering noise is typically far below the level of the music.
  Whether a filter operates out of audibility or very far out of audibility doesn't really matter either.
   
  Quote: 





> More practically, there is benefit when using high-res files with non-oversampling DACs. That removes two more steps between a digital master and the playback: The down-sampling to CD quality and the oversampling circuitry in the DAC. The benefits, of course, are debatable, as always.


 
  Sure, but at 96 kHz the sine wave posted above still doesn't even come close to what you get with a proper DAC and a signal sampled at 44.1 kHz (the DAC actually operates in the 5 MHz region). What you'll get with non-oversampling DACs, which btw ignore the sampling theorem, is aliasing, intermodulation distortion, nonlinearity etc. (I'm not going to talk about the negative effects of analog filters employed in non-oversampling devices - these actually do have an effect on the audible range!)
   
  Short answer: non-oversampling D/A conversion and high fidelity are mutually exclusive.


----------



## bigshot

One of the universal truths of home audio is that you generally have to pay extra for messed up sound.


----------



## Sandman65

So, if I'm understanding this all correctly....properly mastered redbook will sound every "bit" as good as similarly mastered SACD, BlueRay and/or DVD-A?


----------



## bigshot

Yes, all things being equal.


----------



## ctoth666

Quote: 





stv014 said:


> Why don't you just post the files, so that others can have a chance of finding out if there is any difference, rather than having to guess ?


 
  Well this is a link to my SoundCloud http://soundcloud.com/ctoth666/sets/audio-test-files-headfi-org/


----------



## Sandman65

This Forum should be retitled "Welcome to Zion"


----------



## stv014

Quote: 





ctoth666 said:


> Well this is a link to my SoundCloud http://soundcloud.com/ctoth666/sets/audio-test-files-headfi-org/


 
   
  Unfortunately, there is no download link for these, and with the workaround I found to allow for downloading the files, the format seems to be 128 kbps MP3, which makes high source sample rates and resolutions irrelevant. But after opening two of the files, I already found that they are not level matched, and that the sound was generated with software synthesis that has some random variation in the algorithm (i.e. it always produces slightly different audio; I suspected that this might be the case). So, there are obvious differences that have nothing to do with "high resolution" formats. The best way to test the audibility of those is to generate a single track in the highest quality format, and then convert that to the low resolution/sample rate ones and back, and compare the original and degraded files using the foobar ABX plugin.


----------



## grokit

Quote: 





sandman65 said:


> So, if I'm understanding this all correctly....properly mastered redbook will sound every "bit" as good as similarly mastered SACD, BlueRay and/or DVD-A?


 
   
  Quote: 





bigshot said:


> Yes, all things being equal.


 
   
   
  I'm with you on that. But for some reason I still want the capability of 192k and everything in between.


----------



## Currawong

Quote: 





xnor said:


> Quote:
> 
> 
> 
> ...


 
   
  I agree about high res files and mastering being better. That is one primary reason I like them.
   
  Here are some graphs though that are the basis of what I am saying. I was using a Oscium iPad-connected scope.
   
  Metrum Octave 20 kHz 16/44.1 sine wave output, which is, as I said, "not far off what you get" in that graph.
   
   

   
  Reference 7 20 kHz 16/44.1 kHz sine wave (uses 8x oversampling, cut-off filter as conventional):
   
   

   
  Metrum Octave 20 kHz 24/96 sine wave output (note the horizontal scale is slightly different from the graph above):


----------



## xnor

Yes all three a pretty bad for what should be a clean 20 kHz *sine *(not triangle, square or something in between them) wave. After all, you can get 128x oversampling from < $1 chips and also clean results like this from onboard audio:
   

   
  I can only quote what bigshot posted:
  Quote: 





bigshot said:


> One of the universal truths of home audio is that you generally have to pay extra for messed up sound.


----------



## ctoth666

Quote: 





stv014 said:


> But after opening two of the files, I already found that they are not level matched, and that the sound was generated with software synthesis that has some random variation in the algorithm (i.e. it always produces slightly different audio; I suspected that this might be the case).


 
  Yes there is randomization in the algorithm, but the generated sound is still the same across all of the files is it not? I used the same Live project, and it was merely exported with different settings. Also how is it that they aren't level-matched? All of the files were normalized. I just don't know what any of this means tbh. I'm out of my depth. All I know is that I tweaked the export settings on Ableton Live project, and all of the files sound slightly different. That's all I know. And if I am to assume that there should be no audible difference between bit and/or sample rates, then pardon my daftness but that confuses the heck out of me. Anyway, I'm going to muck around with this a bit more and see what potentially erroneous conclusions I can come to.


----------



## stv014

Quote: 





ctoth666 said:


> Yes there is randomization in the algorithm, but the generated sound is still the same across all of the files is it not?


 
   
  It is different enough to significantly outweigh the small differences between 16 bit vs. 24 bit, and 44.1 kHz vs. 96 kHz.
   
  Quote: 





ctoth666 said:


> All of the files were normalized.


 
   
  Normalizing the peak amplitude does not guarantee equal loudness, especially with the above mentioned randomization that also adds randomness to the peak level. You would need to use something like ReplayGain for more accurate matching.
   
  However, the best approach is to generate only one file with the software synthesizer, at the highest possible quality (i.e. 192 kHz/32-bit), leaving a couple dB of headroom so that the sample rate conversions will not result in clipping, and converting that file to all the other (lower quality) formats to be tested. Finally, to make sure that the DAC does not introduce any differences, convert the files back to 192 kHz/32-bit, and compare those with the foobar2000 ABX comparator. If the software you use for the conversions is well written, you should not need to apply any level matching or synchronization.


----------



## ctoth666

Quote: 





stv014 said:


> It is different enough to significantly outweigh the small differences between 16 bit vs. 24 bit, and 44.1 kHz vs. 96 kHz.
> 
> 
> Normalizing the peak amplitude does not guarantee equal loudness, especially with the above mentioned randomization that also adds randomness to the peak level. You would need to use something like ReplayGain for more accurate matching.
> ...


 
  Ok, I went ahead and generated a new file at the optimized settings and converted it to 16/44.1 and back again. I also exported multiple files from the same Live project as I did before. I also ABX'ed. I have determined the following:
   
  -I cannot tell the difference between any of the derivate files of the original 192/32 master file. You were correct.
  -When using midi, none of the audio has been captured prior to exportation. The initial settings for exporting the master file greatly affect the audio quality. Exporting at 192khz vs 96khz and so on produces a marked difference in quality, which likewise applies to the bit rate. These different export settings can ABX'ed at 100% accuracy. This has been the source of my confusion. However, interconverting the files post-export does not seem to result in any noticeable difference.


----------



## XxDobermanxX

*"When 44.1 and 96kHz are compared it gets real subjective"*


----------



## Benjamin6264

I thought that what mattered was the sample rate, and that the bit depth was only to allow for higher sample rates.


----------



## mikeaj

Quote: 





benjamin6264 said:


> I thought that what mattered was the sample rate, and that the bit depth was only to allow for higher sample rates.


 
   
  These are two separate things.  As for whether or not sample rate past a certain point matters, practical implementations, ultrasonics, etc., there is plenty of information (also a lot of misinformation) on that.  See the last page or two here, for instance.
   
  Anyway, the bit depth is the number of bits (commonly 16, 24... also 32) used to represent each sample.  The higher the bit depth, the more values (amplitude levels) can be represented.  There are only a finite number of possible values:  2^16, 2^24, etc., depending on the bit depth.  Note that the difference between two consecutive steps at 24-bit depth or higher is much smaller than the electrical noise in any real-world DAC, and that most studio setups / recordings / etc. (also listening environments) have higher noise than two consecutive steps for 16-bit depth.  The sample rate is how many samples there are per second.  With a higher sample rate, more frequencies can be represented—up to half the sampling rate.
   
  Two tracks (stereo) of 16-bit depth at 44.1 kHz sampling rate is 2 x 16 bits/sample x 44100 samples/second = 1411200 bits/second = 1411.2 kbps.  Increasing the sample rate and increasing the bit depth both increase the bitrate (number of bits / time).  Increasing the bit depth is not necessary to increase the sample rate.


----------



## firev1

Quote: 





ctoth666 said:


> Yes there is randomization in the algorithm, but the generated sound is still the same across all of the files is it not? I used the same Live project, and it was merely exported with different settings. Also how is it that they aren't level-matched? All of the files were normalized. I just don't know what any of this means tbh. I'm out of my depth. All I know is that I tweaked the export settings on Ableton Live project, and all of the files sound slightly different. That's all I know. And if I am to assume that there should be no audible difference between bit and/or sample rates, then pardon my daftness but that confuses the heck out of me. Anyway, I'm going to muck around with this a bit more and see what potentially erroneous conclusions I can come to.


 
  Ableton is not very transparent with its downsampling algorithm, use SoX libraries for that job. Or if you are not well verse in programming(I'm pretty dumb myself), Adobe Audition 5.5/6 has very transparent sampling algorithms.
   
  Anyone would like to have a 96/88.2khz downsampling test, I will downsample one of the Master's of Their Day tracks and upload them later.


----------



## Currawong

Quote: 





xnor said:


> Yes all three a pretty bad for what should be a clean 20 kHz *sine *(not triangle, square or something in between them) wave. After all, you can get 128x oversampling from < $1 chips and also clean results like this from onboard audio:


 
   
  Sorry, I should have added that the scope (an Oscium thing that attaches to the iPad) has errors which show up whatever I measure, whether it be an iPod, my MacBook Pro or a DAC. The point I wanted to make wasn't in regards to the linearity of the DACs in particular, as I don't have the proper equipment to measure that. The point I've explained in the posts I made. If you want to find some excuse to invalidate my comments, please address where you believe I'm wrong in what I'm commenting on, which was to illustrate a real example of the behaviour of a NOS DAC and how it behaves similarly to an oversampling DAC with high-res files.


----------



## xnor

Quote: 





currawong said:


> Sorry, I should have added that the scope (an Oscium thing that attaches to the iPad) has errors which show up whatever I measure, whether it be an iPod, my MacBook Pro or a DAC. The point I wanted to make wasn't in regards to the linearity of the DACs in particular, as I don't have the proper equipment to measure that. The point I've explained in the posts I made. If you want to find some excuse to invalidate my comments, please address where you believe I'm wrong in what I'm commenting on, which was to illustrate a real example of the behaviour of a NOS DAC and how it behaves similarly to an oversampling DAC with high-res files.


 
  No no I don't want to invalidate your comments. I was just commenting on the performance.
  I agreed with the mastering point you made, but for a non-oversampling DAC to behave similarly to an oversampling DAC you'd need files sampled at a much higher rate. But since there's a compromise between speed and accuracy there are no AD converters that work at such high sampling rates. So you'd have to oversample on your computer or in a pre-filtering stage which kinda defeats the purpose of non-oversampling DACs. There are many more problems but I think I've made my point pretty clear.


----------



## zakazak

So in a nutshell, those settings should be perfectly fine for me:
   
  Audio-GD NFB 12.1 -> OS: 4, Filter: 2 (altough this means 4x oversampling?)
  Windows sound properties: 16 bit, 48kHz
  foobar2000 + WASAPI (16 bit output) + resampler (48kHz)
   
  While using Sennheiser HD600.
   
  Although it seems like 24 bit won't do any harm ?


----------



## Currawong

Quote: 





xnor said:


> Quote:
> 
> 
> 
> ...


 
   
  Many DACs use 2x or 4x oversampling (though most AFAIK use 8x for 44.1kHz data), which is in the range of available high-res files (88.2 or 176.4 kHz). I think DSD and DXD recording covers the point you make, though I don't know what equipment is available to make such recordings. Whether there are actual benefits, of course, is another matter altogether. It's an interesting topic, anyway.


----------



## xnor

Quote: 





currawong said:


> Many DACs use 2x or 4x oversampling (though most AFAIK use 8x for 44.1kHz data), which is in the range of available high-res files (88.2 or 176.4 kHz). I think DSD and DXD recording covers the point you make, though I don't know what equipment is available to make such recordings. Whether there are actual benefits, of course, is another matter altogether. It's an interesting topic, anyway.


 
  Nope, DACs using delta-sigma modulation typically work at 128 times oversampling / the sampling rate. 1-bit DACs wouldn't even work at just 4x or 8x oversampling. The minimum should be somewhere around 32x iirc.
   
  DSD also is a single bit signal sampled at 64 times the CD audio sampling rate. DXD has 24 bit resolution and is sampled at 8 times the CD audio sampling rate, but does your NOS DAC accept such a signal?


----------



## firev1

https://www.dropbox.com/s/n8tad28hnb1pgef/A.aif
https://www.dropbox.com/s/fmcqtmjknj6kz8i/B.aif
   
  These are from Masters From Their Day episode one. Let me know what you guys think, whether A or B is the resampled one. I expect the tracks to be played on Foobar with A/BX plugin and replaygain on.


----------



## Makiah S

Wow nice read, I do need that WASAPI plug in, and although I did not read the entire thread [only pages 1 and this one]
   
  While technically there may not be a differance, [and I have 24 and 16bit recording of which there is minimal or no differeance] but the placebo of "extra depth" does make your music sound better... why because we humans are emotional, and the joy of finding that 24bit flac of your favorite song just makes it all the better <3
   
  Still though good read and it makes a lot of sense, good stuff to know!


----------



## MrMateoHead

In theory nothing really matters. In reality however, I've never heard two different albums sound exactly the same. I've never heard two speakers sound exactly the same. People want to screw around with hardware to try to tweak the best sound, its understandable. I even use EQs! Yipes!


----------



## MrMateoHead

Also can anyone explain how or why so many companies choose to rely on a 1-bit DAC?


----------



## xnor

Quote: 





mrmateohead said:


> Also can anyone explain how or why so many companies choose to rely on a 1-bit DAC?


 
  High linearity and low cost. Btw, sigma-delta modulation can be mixed with multi-bit DACs.


----------



## Makiah S

Quote: 





xnor said:


> High linearity and low cost. Btw, sigma-delta modulation can be mixed with multi-bit DACs.


 
  whoa explain plox


----------



## stv014

Quote: 





mshenay said:


> whoa explain plox


 
   
  A 1-bit DAC is the easiest and cheapest to implement, as it has only two possible output levels. With only two levels, there is - ignoring other factors - in theory no distortion from the D/A conversion itself. With the current technology, increasing the clock frequency to the MHz range is also cheap, as is the DSP necessary to implement the oversampling and noise shaping. By contrast, a high resolution and accurate resistor ladder (that could match the ability of a delta-sigma DAC to convert 24 bit PCM data at less than 0.001% distortion) is more difficult and expensive. Basically, the complexity is moved from the analog domain to the digital one, where sound quality becomes a function of speed and transistor count, which can be increased at low cost, unlike analog accuracy. The use of oversampling also allows for a digital reconstruction filter, making the analog lowpass filter simpler and cheaper.
   
  The reason why multi-bit DACs are still used is that in a 1-bit format, it is impossible to implement a proper dither that makes the quantization error uncorrelated to the input signal (the sum of the dither noise and the signal will get clipped). A low resolution (e.g. 4-bit) multi-bit DAC avoids the dithering problem, in addition to reducing the total amount of quantization noise, and is a good compromise overall.


----------



## Makiah S

Quote: 





stv014 said:


> A 1-bit DAC is the easiest and cheapest to implement, as it has only two possible output levels. With only two levels, there is - ignoring other factors - in theory no distortion from the D/A conversion itself. With the current technology, increasing the clock frequency to the MHz range is also cheap, as is the DSP necessary to implement the oversampling and noise shaping. By contrast, a high resolution and accurate resistor ladder (that could match the ability of a delta-sigma DAC to convert 24 bit PCM data at less than 0.001% distortion) is more difficult and expensive. Basically, the complexity is moved from the analog domain to the digital one, where sound quality becomes a function of speed and transistor count, which can be increased at low cost, unlike analog accuracy. The use of oversampling also allows for a digital reconstruction filter, making the analog lowpass filter simpler and cheaper.
> 
> The reason why multi-bit DACs are still used is that in a 1-bit format, it is impossible to implement a proper dither that makes the quantization error uncorrelated to the input signal (the sum of the dither noise and the signal will get clipped). A low resolution (e.g. 4-bit) multi-bit DAC avoids the dithering problem, in addition to reducing the total amount of quantization noise, and is a good compromise overall.


 

 Well I'm never a fan of cheap quick fixes... so what are some examples of these multi bit DACS?


----------



## xnor

Quote: 





mshenay said:


> Well I'm never a fan of cheap quick fixes... so what are some examples of these multi bit DACS?


 
  These DACs are far from "cheap quick fixes"...
   
  An example for a multi-bit sigma delta DAC chip would be the WM8740.


----------



## Makiah S

Quote: 





stv014 said:


> A 1-bit DAC is the easiest and cheapest to implement, as it has only two possible output levels. With only two levels, there is - ignoring other factors - in theory no distortion from the D/A conversion itself. With the current technology, increasing the clock frequency to the MHz range is also cheap, as is the DSP necessary to implement the oversampling and noise shaping.
> By contrast, a high resolution and accurate resistor ladder (that could match the ability of a delta-sigma DAC to convert 24 bit PCM data at less than 0.001% distortion) is more difficult and expensive.
> 
> The use of oversampling also allows for a digital reconstruction filter, making the analog lowpass filter simpler and cheaper.


 
  Basically it sounds like the 1-bit DAC's are "simpler and Cheaper" fixes, like for example when your making brownies you can buy a pre made mix and OMG taste GOOD and it was SO simple to make and SO cheap
   
  or
  From Scratch Brownies
  You can CREAM the Butter and Sugar your self then ect... ect... Tastes OMG LIKE SMEX in my mouth, but it took a lot longer and cost me more time due to complexity...
   
  I'd like my DAC to be like those from scratch brownies... AWESOME, because "good" is not GOOD enough
   
  Still though I wish I understood more about electronics so I can avoid putting my foot in my mouth! <3


----------



## xnor

Quote: 





mshenay said:


> Basically it sounds like the 1-bit DAC's are "simpler and Cheaper" fixes, like for example when your making brownies you can buy a pre made mix and OMG taste GOOD and it was SO simple to make and SO cheap


 
  So DSD (SACD) is also a "simple and cheap" format? Seriously, 1-bit SDM is ingenious.


----------



## jcx

table from: http://web.archive.org/web/20070118012711/http://www.iet.ntnu.no/~ivarlo/files/School/PhD/Report_audiodac.pdf
   
  a 2005 graduate level report on audio DAC internals - if you really want to know more about multi-bit delta-sigma


----------



## Makiah S

Quote: 





xnor said:


> So DSD (SACD) is also a "simple and cheap" format? Seriously, 1-bit SDM is ingenious.


 
  See, there my foot is now in my mouth <3
   
  *sigh* I DONT WANT TO BE IGNORANT! *reads article*


----------



## BernieW

Hi, just joined this forum because this topic has interested me for a while. I work in pro audio, mostly live. I've been reading a bit about this debate, including xiph's and Lavry's digital audio articles. I recently bought a computer audio system, and I try to get 24/96 or other high data rate tracks whenever possible.
   
  I accept that 16/44.1 CD format is adequate, in theory, for almost everybody. I'm also willing to believe that some people perceive (not HEAR, but PERCEIVE) frequencies above 20kHz. I DON'T believe "hi-res" media's purpose is to reproduce frequencies above 20kHz. The filter in most DAC's would account for that anyway (so flame away, but I don't care. I have a thick skin, I've been online for years).
   
  So here's a question I haven't seen addressed yet. (The topic was actually about bit depth, not sample rate, but this thread has covered a lot of ground). Sampling theory says 44.1kFs/s is enough to fully encode an analogue waveform with content up to 20kHz... in theory. But that's where I see the problem. Surely that represents an ideal, and all practical systems are going to fall short of that ideal. The output wave will differ from the original thanks to that short-fall, meaning distortion. So wouldn't a stream of more samples per second mean reproduction will deviate less from the ideal? (because it's being corrected/referenced more often).
   
  The problem I've always had with CD audio is that there's enough information to perfectly recreate the original waveform, but only just. Won't economics mean there's so much compromise in most systems that deviation from that ideal is pretty high? After all, those 20 buck portable players DO sound awful. I'm suggesting imperfect (real world) hi-res playback should have an advantage over equivalent standard rate playback. Very little equipment, only the most expensive, is near-ideal. So do economics and the real world equal a case for high-accuracy formats?


----------



## xnor

Quote: 





berniew said:


> So here's a question I haven't seen addressed yet. (The topic was actually about bit depth, not sample rate, but this thread has covered a lot of ground). Sampling theory says 44.1kFs/s is enough to fully encode an analogue waveform with content up to 20kHz... in theory.


 
  Nope, it can contain frequencies up to just below 22.05 kHz. Giving filters some room to work we can say that 20 to 21 kHz is a practical upper limit.
   
  Quote: 





> But that's where I see the problem. Surely that represents an ideal, and all practical systems are going to fall short of that ideal. The output wave will differ from the original thanks to that short-fall, meaning distortion.


 
  Cue: oversampling DACs.
   
  Quote: 





> So wouldn't a stream of more samples per second mean reproduction will deviate less from the ideal? (because it's being corrected/referenced more often).


 
  More samples would just be redundant. And too many (e.g. 192 k) samples per second will cause real-world performance to suffer.
   
   
  Quote: 





> The problem I've always had with CD audio is that there's enough information to perfectly recreate the original waveform, but only just. Won't economics mean there's so much compromise in most systems that deviation from that ideal is pretty high? After all, those 20 buck portable players DO sound awful. I'm suggesting imperfect (real world) hi-res playback should have an advantage over equivalent standard rate playback. Very little equipment, only the most expensive, is near-ideal. So do economics and the real world equal a case for high-accuracy formats?


 
  We've got a lot closer to the ideal over time, still do.
  A $ 20 portable player sounding awful has little to do with the CD format.


----------



## stv014

Quote:  





> The problem I've always had with CD audio is that there's enough information to perfectly recreate the original waveform, but only just. Won't economics mean there's so much compromise in most systems that deviation from that ideal is pretty high?


 
   
  DAC chips that cost only a few dollars are close enough to the ideal now, it is not that hard for an oversampling DAC to reconstruct a signal sampled at 44.1 kHz without audible artifacts.


----------



## AstralStorm

/me = slowpoke. The thread has 60 pages. I'm almost certainly sure the points have been mentioned multiple times.
   
   
  Quote: 





stv014 said:


> DAC chips that cost only a few dollars are close enough to the ideal now, it is not that hard for an oversampling DAC to reconstruct a signal sampled at 44.1 kHz without audible artifacts.


 
   
  Heck, even 0.5$ DAC can be pretty good. The real issues are usually in the analog part further down, that is, amp. I can make a pretty decent DAC+amp for $20 in parts, definitely good enough for 16-bits. The trick is to put these parts in correct topology and be careful about grounds. Granted, it won't be exactly Objective O2 class amp, closer to, say, FiiO E5. (sans the bass roll and THD from undersize cap as well as with better channel separation and bit less noise in general - those are sacrificed for battery life and size.)


----------



## stv014

Quote:  





> Heck, even 0.5$ DAC can be pretty good. The real issues are usually in the analog part further down, that is, amp.


 
   
  That is not really relevant to the question of whether CD format is good enough with a typical reconstruction filter, though, since the digital filter is in the DAC chip itself.


----------



## xnor

Quote: 





astralstorm said:


> A) Dither does not eliminate quantization errors, only decorrelates the error from the signal, making it less detectable/more pleasant, but not any less measurable. Decorrelated noise put into an integrator (averaging system) will be greatly reduced at the output, how much depends on the integration length. Correlated noise causes frequency response spikes similar to comb filtering.


 
  That's why he wrote "eliminate quantization *distortion*" and: "Essentially during the conversion process a very small amount of white noise is added to the signal, this has the effect of completely randomising the quantisation errors."
   
  Quote: 





> B) [...] Nyquist also deals with continuous signals, but temporal resolution of DACs/amps is also great.  Any issue there is nothing higher digital bit depth can fix - unless the DAC happens to use a different reconstruction filter for bit depths - that is more common with different sample rates, but usually if one of the filters is broken, they all are.
> (Hello, Hifiman HM-801; cheap chinese noisy amps. Also high output impedance effects.)


 
  I don't understand what you're saying here. Could you rephrase this please or expand a bit?
   
  Quote: 





> C) Higher digital SNR allows you to reduce volume in digital domain (subtract) quite a lot without truncation loss or noise. However, remember that 50% loudness is just about 6 dB cut. (@ 1kHz)  This difference means that you will likely increase loudness in analog domain, which might or might not introduce more noise than that.


 
  Sure, but you can play a 16-bit track using a 24-bit or 32-bit DAC. Doesn't change that 16-bit seems to be enough to store audio.
   
   
  Quote: 





> This mostly comes into play when you're doing any DSP, such as equalization, where that ~20 dB extra dynamic range might be useful, otherwise the truncation might be just barely audible. (since there are 70 dB SNR remaining or so) Even if the recording is louder, you've added noise by boosting the reference volume, the noise might or might not be perceptually masked.  That's why good equalizers work with at least 24-bit precision and dither back if required. Certain lousy equalizers work with the same bit depth as input and/or output though.


 
  When doing equalization for playback you usually correct the FR of the headphones/speakers. If there's a 10 dB dip in the FR you boost by 10 dB. The signal can again be played through a higher bit DAC. The end result should be the same, as if played on a flat headphone/speaker without eq.
   
   
  Quote: 





> D) You can sometimes hear the difference between 16-bit and 24-bit version of the same track on different medium, because the 24-bit one doesn't use any dynamic range compression and/or has different mastering. It's like with any other remaster.


 
  Yeah, that sucks. They should release properly masters tracks on CD too... in some genres we're down to 3 dB dynamic range.


----------



## dbbloke

I don't have time to read this thread. It's bollocks.
  24 bit is an improvement over 16bit. There is more air around things, there is more description of the reverb of the room, things sound more lifelike and slightly more subtle. I know this 100% for sure, it's a fact. No matter of flawed science can prove it otherwise.
   
  Whether or not it makes THAT much difference or that much MORE of a music experience though ... probably not much. If anything like a good DAC, it can ruin with extra vision like 1080p sometimes does.


----------



## Puranti

Quote: 





dbbloke said:


> *I don't have time to read this thread. It's bollocks.*
> *24 bit is an improvement over 16bit.* There is more air around things, there is more description of the reverb of the room, things sound more lifelike and slightly more subtle. I know this 100% for sure, *it's a fact.* *No matter of flawed science can prove it otherwise.*
> 
> Whether or not it makes THAT much difference or that much MORE of a music experience though ... probably not much. If anything like a good DAC, it can ruin with extra vision like 1080p sometimes does.


 
   
  Why even bother ?


----------



## stv014

Quote: 





dbbloke said:


> I don't have time to read this thread. It's bollocks.
> 24 bit is an improvement over 16bit. There is more air around things, there is more description of the reverb of the room, things sound more lifelike and slightly more subtle. I know this 100% for sure, it's a fact. No matter of flawed science can prove it otherwise.


 
   
  So you did something like this:
  - take a 24-bit track and convert it to 16-bit with dithering
  - optionally convert the 16-bit file back to 24-bit again
  - compare the original and degraded 24-bit file with the foobar2000 ABX comparator plugin (or other ABX software of your choice)
  And you can tell the files apart with at least 95% confidence, at a sane volume level ?


----------



## skamp

dbbloke said:


> I know this 100% for sure, it's a fact.




Well, you can't beat FACTS. I'm convinced!



stv014 said:


> compare the original and degraded 24-bit file with the foobar2000 ABX comparator plugin




Don't you know that ABX tests are fundamentally FLAWED? They confuse your brain into thinking that everything sounds the same! They're EVIL! We've got people claiming that they can't ABX even 128 kbps MP3! Non-sense!!!!!1!!ONE


----------



## xnor

Quote: 





> 24 bit is an improvement over 16bit. There is more air around things, there is more description of the reverb of the room, things sound more lifelike and slightly more subtle. I know this 100% for sure, it's a fact. No matter of flawed science can prove it otherwise.


 
  You just made a claim. Prove it. If not: silly troll attempt.
   
  Nope, you just saying so doesn't mean anything. See my signature.


----------



## AstralStorm

Quote: 





xnor said:


> That's why he wrote "eliminate quantization *distortion*" and: "Essentially during the conversion process a very small amount of white noise is added to the signal, this has the effect of completely randomising the quantisation errors."
> 
> I don't understand what you're saying here. Could you rephrase this please or expand a bit?


 
  Specifically, dither improves specific inharmonic distortion.
   
  About the other point: Nyquist-Shannon theorem says things about stationary signals (continuous and repeating). Typical signals though have a beginning and the end - it's there where the reconstruction filter response matters. This is typically the windowed sinc filter, as (straight) sinc filters are covered by the second part of the theorem.
   
  Almost all devices have very good reconstruction filters, dropping at most 1 dB at the highest end and adding neglible distortion. There are rare mistakes, like the 18 dB rolloff in the audible band of HM-801 - but these won't be fixed by bit depth. There are other limitations at the lowest end due to (at least stray) inductance forming a highpass filter.
   
  There is a limit on reconstruction filters too: https://en.wikipedia.org/wiki/Cheung%E2%80%93Marks_theorem
  And here's the dual for time (transient) resolution: https://en.wikipedia.org/wiki/Balian%E2%80%93Low_theorem
  The issue is similar to Schroedinger's equation in wave form.
   
  Excess bit depth will slightly improve interpolation filter's quality - the result should be slightly better distortion figures as well as noise. (with or without dithering or AWGN added)


----------



## dbbloke

Convert from 24-16-24 ??? Whats wrong with doing a recording straight from a desk 24bit and then 16 bit? Yeah, it's not going to sound identical cause it's hard to play completely the same piece twice, but will it sound different? Lot simpler. I guess you need good mikes etc..
2 adcs simultaneously might not be the go on account of different electronics, but with a master clock, high grade low tolerance components should be close. I suppose if you had 2 adcs you could do 16 bit on each and see if they sound the same. then 16 vs 24 and see if it;s the same as well  ?

Who says science is correct. It isn't, not even close in a number of areas. I mean just look at a similar topic global warming.

Seems there are 24bit activists / terrorists as well now. It's just plan funny a thread got this long on something that's obvious.
It's like arguing you don't need speakers that reach below frequencies under the bottom note of a bass guitar or something. Super tweeters are interesting as well.

I guess I should have made more of a name for myself as a hifi reviewer, any audio people who know me and trust me would just take my word.
To actually prove to you is a bit more tricky, it would involve sitting in front of a great hifi setup (not headphones for sure) and playing back identical source material at different bits etc. The new graceland 25th anniversary release is nice, remastered in 24 and 16 bit. I should try the test myself. Or maybe just try and get my favourite record label to record 24 bit just for me 
http://www.computeraudiophile.com/f14-music-analysis-objective-and-subjective/graceland-24-96-a-14025/

Not heard my decent hifi in 24bit yet it's in a container for the last 3 yrs, did all my 24 bit tests a while ago.
So I can only suggest people go to a good hifi shop like I did (few and far between) and get a 24bit demo room booked for the afternoon.
Take along your own files, go to they gym before, drink lots of water etc.. Cardio workout does more wonders for hifi imho.

On the topic of files, I guess you can hide behind mastering and release formats as to why it's more detailed using the same DAC, laptop, amps, speakers, cables, just a different file. But if 24 bit files are more detailed in general over many files (its what i found, instruments consistently more real) then there is a good enough reason to go big files.

So ok, argue it out in science, this thread really is pointless like WAV vs FLAC or foobar vs jplay, I2s etc.. Enjoy  I bother to reply because I have read a few places 24 bit isn't any different where the science isn't there yet and it's just plain annoying and defeating to progress. 24 bit had enough issues.
BTW, Linn are offering free 24 bit downloads ATM. http://christmas.linn.co.uk/


----------



## xnor

Quote: 





dbbloke said:


> Convert from 24-16-24 ??? Whats wrong with doing a recording straight from a desk 24bit and then 16 bit? Yeah, it's not going to sound identical cause it's hard to play completely the same piece twice, but will it sound different? Lot simpler.


 
  Wait a second, what's wrong with doing a simple conversion? There's no point in comparing two different recordings. Even if you had identical mics, they cannot be in the same place.
   
  Quote: 





> Who says science is correct. It isn't, not even close in a number of areas. I mean just look at a similar topic global warming.


 
  24/16-bit audio and global warming .. similar topics? Does not compute!
  And who says you are not wrong?
   
  Quote: 





> I guess I should have made more of a name for myself as a hifi reviewer, any audio people who know me and trust me would just take my word.


 
  So hifi reviewers are less wrong / more trustworthy. You're kidding right? Especially considering the affiliations ...
  The beauty of science is that we do not have to take somebody at his/her word.
   
  Arguments from authority .. irrelevant, anecdotal evidence ... irrelevant.
   
  Quote: 





> To actually prove to you is a bit more tricky, it would involve sitting in front of a great hifi setup (not headphones for sure) and playing back identical source material at different bits etc. The new graceland 25th anniversary release is nice, remastered in 24 and 16 bit. I should try the test myself.


 
  Ensure the 16-bit file wasn't mastered differently.
   
  Quote: 





> On the topic of files, I guess you can hide behind mastering and release formats as to why it's more detailed using the same DAC, laptop, amps, speakers, cables, just a different file. But if 24 bit files are more detailed in general over many files (its what i found, instruments consistently more real) then there is a good enough reason to go big files.


 
  Hide behind? These are variables that have to be controlled in order to get any meaningful result. But maybe that simple fact is above some hifi reviewers heads..
   
  Quote: 





> So ok, argue it out in science, this thread really is pointless like WAV vs FLAC or foobar vs jplay, I2s etc.. Enjoy


 
  The thread is not pointless, but irrelevant replies are. 
	

	
	
		
		

		
		
	


	



   
  Quote: 





> I bother to reply because I have read a few places 24 bit isn't any different where the science isn't there yet and it's just plain annoying and defeating to progress. 24 bit had enough issues.


 
  But misinformation aids progress?
   
  Quote: 





> BTW, Linn are offering free 24 bit downloads ATM.


 
  Oh, do they also offer 16-bit files for "comparison" that are secretly mastered differently .. like they did a while ago on this forum, I guess, to promote their more expensive 24-bit files?
   
   
  edit:
  Quote: 





dbbloke said:


> My testing was done an a 250k system.


 
  And what is the point of that?


----------



## Puranti

It's not because you can't understand science that it is false...


----------



## bigshot

Why is it that people who don't understand how audio works seem to own $250K systems. There's a correllation there somewhere.


----------



## Strangelove424

Quote: 





dbbloke said:


> Who says science is correct. I mean just look at a similar topic global warming.


 
   
  Just when I thought I found the sane and rational Head-Fi sub-forum.


----------



## bigshot

Quote: 





strangelove424 said:


> Just when I thought I found the sane and rational Head-Fi sub-forum.


 
   
  It's OK. He's just visiting.


----------



## dbbloke

Quote: 





xnor said:


> Quote:
> 
> 
> 
> ...


 
   
  I agree, just when you think you find a decent forum, people respond to posts only from their perspective. And don't give others any credit, why is the forum world so often closed minded, judgemental, pretentious.


----------



## chewy4

Quote: 





dbbloke said:


> *why is the forum world so often closed minded, judgemental, pretentious.*


 
   
  That describes the exact nature of your posts in this thread.
   




   
   
  If you see a flaw in something specific about the science being discussed in this thread, feel free to point it out. But to say that all science is flawed is a little ridiculous. If you think that's true, I would avoid any doctors, medicine, etc... and just step back from your computer because it was made using theories from science too. And you don't want none of that.


----------



## mikeaj

dbbloke said:


> I agree, just when you think you find a decent forum, people respond to posts only from their perspective. And don't give others any credit, why is the forum world so often closed minded, judgemental, pretentious.




We hope that in a sound science forum, nobody takes arguments personally (because they're not intended that way), and if you have a different theory or evidence to support a different conclusion, you please contribute that rather than launching ad hominem attacks. If you feel certain about something that is easy to test, then test it, if you have the time and will. If there is not much justification for any claims made, then expect skeptics to be skeptical. If that bothers you, prove them wrong. If it doesn't, then we can all just move on.


----------



## xnor

Quote: 





dbbloke said:


> So you can't tell the difference between a les paul and a stratocaster.
> Instruments all have a unique sound (stradivarius etc) and as such, if you record one 16 bit and another 24 bit you will notice the difference in the detail and the fingerprint of the instrument. So you don't need to identical recordings at all.


 
  I'm repeating my question: what's wrong with a simple conversion?
   
  You don't seem to understand even something as simple as an ABX test. If we just want to test 16 vs. 24 bits why on earth would you use different recordings, instruments ...
   
   
  Quote: 





> I know I'm not wrong, best way to go about life, be open to everythig. You can say I am I don't care so much about one person really.


 
  Oh, how convenient. It doesn't matter whether you are delusional or if 10, 100 or 1000 people say you are wrong - it doesn't change reality one bit.
   
  Being open is good, but are you also open to the idea of eliminating biases? Blind tests? Science in general? If not, you're clearly closed-minded.
   
   
  Quote: 





> The problem with science is we don't understand the system yet! So it's pointless to try other than to prove the science is flawed.


 
  Are you saying you don't understand science? You're certainly right about that.
   
   
  Quote: 





> Come on, I agree in general most people aren't so good at things, but you can't generalise about everyone.


 
  Look up the terms arguments from authority and anecdotal evidence if you don't know what they mean.
   
   
  Quote: 





> I was trying to say that some people here clearly haven't heard a difference between 24 and 16 bit files. They think their science is right and the understanding of how people perceive music complete. It's just strange when there is an obvious difference, people use science and all kinds of arguments not to accept what is there. Perhaps they just need to listen on a decent non-headphone system.


 
  And I was trying to say that some people *think* they have heard a difference between 24 and 16 bits. They think that science is wrong and there's magic to how people perceive music. (Btw, I've seen nobody claim that they understand how people perceive music completely...)
  It's just strange when there is no obvious difference, people use anecdotes and arguments from authority etc. to try to convince others of something that's not even there. Perhaps they just need to do a proper test.
   
   
  Quote: 





> It's pointless because no matter what science you throw at something, if the science isn't at a level of describing the real world then it's pointless trying to use science.


 
  You really don't understand science. :/
   
   
  Quote: 





> So you think I'm not so newb who is using a pair of ****ty AKG headphones linked up to an fiio or something. I dont have a 250k stereo, it was in a shop for half a day.


 
  No, I was asking you: what is the point of that? You complain about science being unrealistic but we should believe some random, anonymous guy who claims to have heard differences on a 250k system without a shred of evidence?! LOL.


----------



## mark_h

Soooo, if, as others have mentioned in different threads, they cannot hear a difference between 320 mp3 and .wav and now in this thread people are saying there is no discernible difference between 16 or 24 bit then it stands to reason there is no difference in sound quality between 320mp3 and 24bit studio quality?


----------



## skamp

dbbloke said:


> Instruments all have a unique sound (stradivarius etc) and as such, if you record one 16 bit and another 24 bit you will notice the difference in the detail and the fingerprint of the instrument. So you don't need to identical recordings at all.




The question is whether 24 bit is audibly different from 16 bit. In order to answer that question, one needs isolate that difference from all others, otherwise there would be no way to tell which difference was really detected. The only way to do that is, first, to take one 24 bit recording and convert it to 16 bit, and second, to remove all sighted bias (what's the point of testing the strict audibility of something if other senses can instantly reveal the difference?), by the way of an ABX (i.e. double blind) test. Only when those conditions are met and all other differences are eliminated, will you be able to tell with certainty whether 24 bit makes an audible difference or not. And that's not so much science as it is logic and common sense.


----------



## bigshot

mark_h said:


> Soooo, if, as others have mentioned in different threads, they cannot hear a difference between 320 mp3 and .wav and now in this thread people are saying there is no discernible difference between 16 or 24 bit then it stands to reason there is no difference in sound quality between 320mp3 and 24bit studio quality?




Not at normal listening volumes. High bit rates are important for mixing, when you might need to boost the volume a great deal to bring up a detail in the mix. With 24 bit it comes up clean, but boosting an MP3 that far brings up noise that you can't hear at normal listening levels.


----------



## compsalot

in response to the endless debate about 192k samples, is it better?
   
  I suppose that it is pointless to reply to a thread that is this old.  But someone on the internet is wrong!!  http://xkcd.com/386/   and so I just had to say something.  
   
  People have persistantly misunderstood the significance of the Nyquist frequency.  He never said that 2x sampe rate gets you a good quality reproduction of the signal...  he said it was the *absolute minimum* that was needed to be able to capture the signal at all. 
   
  But at a 2x sample rate what you are doing is turning a sine wave into a square wave.  There is a huge qualitative difference between them.  If you haven't already had that experience then you should go get your self a tone generator and set it to sine and then to square waves at the same frequency and listen to the difference between them, the square waves sound like krap.
   
  It is like the difference between DPI (dots per inch) and Pixels.  If you want to print a high quaility representation of a photo then you want a very high dots per inch printer, say 1200, even if the photo itself is only 200 pixels per inch.  That is because it takes lots of dots to do a good job of smoothly representing the pixels.
   
  Audio is the same way...  you can argure all you want about the limits of human perception, which does vary greatly from person to person, and is further subject to various biases.  But when it comes to actual wave-form capture, more samples is better.  You will get less distortion in the portrayal of that wave-form.   The only reason they can get away with a ~40k sample rate is they put a low-pass filter that rounds the corners of the square waves (removes higer frequency harmonics).  This is all very fine and well, but it can lead to a reduction over-all in the high end response, it is hard to make a filter that is not also rounding the frequiencies that you do want.
   
  bottom line: from a purely technical standpoint, 192k sample rate gives you a better wave form, no if's and's or but's about it.  You don't have to take my word for it, just borrow an o'scope and look for yourself. 
   
  But what you do with that wave form (the rest of the components in the audio system) is a totally seperate matter;  as is the question of if your ears are actually sensitive enough to preceive the difference. 
   
  Nobody says you have to record higher frequencies with a high sample rate, that is just nonesense, a total strawman.  You record the same frequencies <20k  but at a higher resolution, that's how you get a smoother wave-form with less distortion.
   
  I'm really surprised that so many "experts" have got this one wrong.


----------



## xnor

Quote: 





compsalot said:


> People have persistantly misunderstood the significance of the Nyquist frequency.  He never said that 2x sampe rate gets you a good quality reproduction of the signal...  he said it was the *absolute minimum* that was needed to be able to capture the signal at all.


 
*No*, you need a sampling frequency greater than two times the maximum frequency to be able to reconstruct the original signal. The reconstruction is *perfect *in theory. It seems you don't understand the theorem.
   
   
  Quote: 





> But at a 2x sample rate what you are doing is turning a sine wave into a square wave.


 
*No*, that's wrong. See reconstruction.
   
   
  Quote: 





> It is like the difference between DPI (dots per inch) and Pixels.  If you want to print a high quaility representation of a photo then you want a very high dots per inch printer, say 1200, even if the photo itself is only 200 pixels per inch.  That is because it takes lots of dots to do a good job of smoothly representing the pixels.
> 
> Audio is the same way...


 
*No*, it's not. You don't understand the sampling theorem.
   
   
  Quote: 





> you can argure all you want about the limits of human perception, which does vary greatly from person to person, and is further subject to various biases.  But when it comes to actual wave-form capture, more samples is better.  You will get less distortion in the portrayal of that wave-form.   The only reason they can get away with a ~40k sample rate is they put a low-pass filter that rounds the corners of the square waves (removes higer frequency harmonics).  This is all very fine and well, but it can lead to a reduction over-all in the high end response, it is hard to make a filter that is not also rounding the frequiencies that you do want.


 
  It seems you're talking about *ancient *non-oversampling DACs.
   
   
  Quote: 





> bottom line: from a purely technical standpoint, 192k sample rate gives you a better wave form, no if's and's or but's about it.  You don't have to take my word for it, just borrow an o'scope and look for yourself.


 
  All DACs (ignoring the ancient nos crap) I know of perform worse at 176.4 or 192 kHz sampling rate.
   
   
  Quote: 





> Nobody says you have to record higher frequencies with a high sample rate, that is just nonesense, a total strawman.  You record the same frequencies <20k  but at a higher resolution, that's how you get a smoother wave-form with less distortion.
> I'm really surprised that so many "experts" have got this one wrong.


 
  LOL, the irony! You don't get increased resolution. Please read up on the sampling theorem and oversampling DACs. Also see #846.


----------



## bigshot

Xnor is right. A simple spin through the relevant wikipedia pages will explain.


----------



## compsalot

There is and continues to be a lot of confusion on this thread about certain concepts, but I also do appreciate that there are also some people here with a huge depth of knowledge far in excess of my own.

  
  Let me translate the above conversation...  see if this makes more sense to you...

  
  I said:  from a *purely technical standpoint*  a car that goes 200 mph is faster than a car that can only go 100 mph and this is a fact which is beyond dispute.

  
  I also said that just because a car can go 200 mph does not mean that you would or should drive it at that speed.  People seem to be confused by the idea that if a car can go 200 mph they are obligated to drive it at 200 mph, that is a major misunderstanding -- and is basically what prompted me to attempt to add some clarity to this thread. (I did not realize my reply would go at the very end of the conversation instead of following the post to which I was specifically replying; why does this forum have a reply button on each message if the reply does not get associated with that message).

  
  I attempted to point out that some people buy cars that go 200 mph because even at much lower speeds they find that the car is more responsive and peppy with smoother acceleration even when operating at the lower speeds.

  
  ----------

  
  Xnor responded to my comments by saying that his car preforms very well when going 100 mph, but if he tries to go 200 mph his tires are not up to the task and he ends up in a ditch.  He then concluded that nobody should drive at 200 mph because it will actually be slower than driving at 100 mph.  He then tried to extend this reasoning to all cars. He further argued that since you are not allowed to go that fast it is pointless to use a car that can.

  
  I concede this point to xnor, that if you end up in a ditch and have to wait for the tow truck then certainly you will not see a net gain in performance by driving at 200 mph because certainly waiting for the tow truck to come, will take a lot more time than if you had proceeded at 100 mph.  e.g. the distortion will kill you.

  
  Further I think he makes a very good point that a lot of the low quality results people complain about is the result of equipment that does not live up to it's specs and does not actually deliver quality performance at 200 mph.  But I also contend that some cars are designed better and those cars do deliver quality at 200 mph.

  

  
  However, we all agree that the actual legal speed limit is 60 mph, so nobody is actually going to be driving faster than 60 mph, although there is a lot of debate about the fact that in some places such as the Nevada freeway it is legal to drive at 80 mph (or so I've heard). 

  
  But this does not change the fact that a car which has more capacity, is going to be more responsive even when driven at a slower speed.  and this might just be enough of a difference that some people will want to pay for that extra measure of quality even though it does not change the fact that you are still only going to get there at 60 mph, the ride just feels a little bit smoother...  and some people appreciate this difference. 

  
  I would even agree that the law of diminishing returns is such that the additional performance difference which results from the extra capacity of a 200 mph car versus the extra capacity of a 100 mph car is likely to be quite small - when both are being driven at 60 mph; but I do not accept that the difference is zero, which is what many people contend.

  
  However, I concede that by the time you down sample to 44.1 CD, it's probable that the *effective* difference is zero.  On the other hand if your target is Blu-ray you do not have to degrade your final mix.  Whether or not someone's stereo+ears can actually discern the difference is indeterminate.  But one should not be too quick to dismiss the psychological satisfaction that people get from having the *best* quality specs, because that satisfaction can also be a part of the total experience, otherwise why would people put so much effort into case design?  which has no impact on the sound at all, but does contribute greatly to the total experience.

  
  A similar technical argument applies to the 24 bit versus 16 bit.  At 24 bits your step size is smaller, your transitions are smoother.  Is this a difference that anyone can actually hear?  that remains an open question, but one thing is certain, at 16 bits it takes effort to avoid clipping, but at 24 bits, it is pretty much a no-brainer that you can both avoid clipping and retain smooth gradations.

  
  For an experiment, find an example photo to view, and set your color mode for 16 bit, now set your color mode for 24 bit, can you tell the difference?  Answer: Only if you have a high quality monitor and good visual acuity and the photo itself encompasses a wide gamut.  Many monitors are not of high enough quality that they can display the difference.  Many peoples eyes are not sufficiently skilled (yes color perception can be learned) that they can discern the difference (which is why monitor manufactures get away with lower quality displays while claiming to be 24 bit).

  
  One final thought, I have also seen a lot of comments expressing concern for the amount of disk space required...  once again I think there is some confusion about this issue. We now live in an era of one terabyte disk drives which can be purchased for less than $100.  This means that a gigabyte now costs ten cents.  One gigabyte will hold about ten hours of 24 bit stereo at 192k samples per sec of uncompressed audio (this figure is conservative, it will actually hold more). Or in other words, the recording will cost you about one penny per hour for storage (five cents if you keep proper backups). Anybody who feels that this is too expensive.... er, well, I rest my case.
   
  (update: I was half-asleep when I wrote this and got the calculation wrong, the correct amount for 192k is 60 cents per hour if you save it to DVD with 2 backups, see the correction in the message below.  At 96k your cost savings would be 50% or 30 cents per hour.  If either of those amounts is a worry to you then this cost is the least of your concerns).

 -------------

  
  I probably never should have waded into this conversation, I see that it has already gone on for years and nearly 1000 comments (I've read about half), with nothing much appearing to be resolved. It was just that I saw some major confusion happening in a lot of the comments and mistakenly thought that I might be able to add something of value here.  But I see that my notion was foolish and am now going to bow out as gracefully as I can manage.

  
  For optimal productivity, I highly recommend reading http://xkcd.com   it makes more sense than does further discourse on this thread.


----------



## skamp

compsalot said:


> One gigabyte will hold about ten hours of 24 bit stereo at 192k samples per sec of uncompressed audio (this figure is conservative, it will actually hold more)




You can't even count.

24 bits * 192,000 samples per second * 2 channels = 9,216,000 bits per second = 1,152,000 bytes per second

1 gigabyte (GB) = 10^9 = 1,000,000,000 bytes
1 gibibyte (GiB) = 2^30 = 1,073,741,824 bytes

1 GB will hold: (1,000,000,000 / 1,152,000) = 868 seconds = 14 min 28 sec
1 GiB will hold: (1,073,741,824 / 1,152,000) = 932 seconds = 15 min 32 sec

1 terabyte (TB) = 10^12 = 1,000,000,000,000 bytes
1 tebibyte (TiB) = 2^40 = 1,099,511,627,776 bytes

1 TB will hold: (1,000,000,000,000 / 1,152,000) = 868,055 seconds = 241 h 7 min 35 sec
1 TiB will hold: (1,099,511,627,776 / 1,152,000) = 954,437 seconds = 265 h 7 min 17 sec

BTW, you totally lost me with the car analogy.

Also, this debate is kinda pointless, because ultimately, proof for audibility has to come from ABX tests (what else is there?), which audiophiles generally don't go through, and when they do and fail miserably, they just resort to denying the validity of such tests. They haven't come up with anything better, either. So this can go round and round until everyone's too exhausted to continue.


----------



## compsalot

oops, I was peacefully soaking the the tub and realized I had miscalculated the storage cost... that's what I get for doing it on the fly.
  so here is how it is done for anyone who is interested... and hasn't already figured it out.
   
  1 terabyte = 1000 gigabytes for a cost of $100. So $100/1000 gives us ten cents per gigabyte for our storage cost.
   
  rounding up to account for overhead, we get 200k samples per second * 3 bytes per sample (24 bits) = 0.6 megabytes per second
  since we are assuming stereo we go 2 * 0.6 = 1.2 megabytes per second.
   
  1.2 megabytes per second * 60 seconds per minute * 60 minutes per hour = 4.3 gigabytes
  4.3 gigabytes * 10 cents per gigabyte = 43 cents per hour
   
  And 43 cents per hour is dirt cheap by any measure. Also if you save it to DVD, they cost about 20 cents these days for about 4.5 gigs of storage
   
  these are ballpark numbers which are good enough for budgetary purposes.  specific values depend on the file system used to format the disk drive etc.
   
   
  P.S.  I won't be making any further responses to this thread, as skamp says, it it pointless to continue to discuss this.


----------



## xnor

Quote: 





> I probably never should have waded into this conversation, I see that it has already gone on for years and nearly 1000 comments (I've read about half), with nothing much appearing to be resolved. It was just that I saw some major confusion happening in a lot of the comments and mistakenly thought that I might be able to add something of value here.  But I see that my notion was foolish and am now going to bow out as gracefully as I can manage.


 
  I understand you're trying to help but tbh you just add to the confusion.
   
  Quote: 





compsalot said:


> I said:  from a *purely technical standpoint*  a car that goes 200 mph is faster than a car that can only go 100 mph and this is a fact which is beyond dispute.
> 
> I also said that just because a car can go 200 mph does not mean that you would or should drive it at that speed.  People seem to be confused by the idea that if a car can go 200 mph they are obligated to drive it at 200 mph, that is a major misunderstanding -- and is basically what prompted me to attempt to add some clarity to this thread. (I did not realize my reply would go at the very end of the conversation instead of following the post to which I was specifically replying; why does this forum have a reply button on each message if the reply does not get associated with that message).


 
  I don't think that people are confused about that. If you want you can store a signal that is limited to fmax = 10 Hz using a 192 kHz sampling rate. It's an utter waste of space, but of course it is possible.
   
   
  Quote: 





> I attempted to point out that some people buy cars that go 200 mph because even at much lower speeds they find that the car is more responsive and peppy with smoother acceleration even when operating at the lower speeds.


 
  The analogy doesn't work. To make proper analogies you first have to understand the theorem.
  Think of two cars that are *identical*, but one is limited to 200 mph and the other one to 100 mph. If you drive at 100 mph your ride will be smoother and quieter. The analogy is still flawed though.
   
  I have no idea what you're trying to say with the further points like the ditch and tow truck.
   
  Quote: 





> A similar technical argument applies to the 24 bit versus 16 bit.  At 24 bits your step size is smaller, your transitions are smoother.  Is this a difference that anyone can actually hear?  that remains an open question, but one thing is certain, at 16 bits it takes effort to avoid clipping, but at 24 bits, it is pretty much a no-brainer that you can both avoid clipping and retain smooth gradations.


 
  Using terms like "transitions" and "gradations", have you even read the first post?
   
   
  Quote: 





> For an experiment, find an example photo to view, and set your color mode for 16 bit, now set your color mode for 24 bit, can you tell the difference?  Answer: Only if you have a high quality monitor and good visual acuity and the photo itself encompasses a wide gamut.  Many monitors are not of high enough quality that they can display the difference.  Many peoples eyes are not sufficiently skilled (yes color perception can be learned) that they can discern the difference (which is why monitor manufactures get away with lower quality displays while claiming to be 24 bit).


 
  Another flawed analogy. 16 vs. 24 bit colors show a clearly visible difference even on an average monitor. After all, the human eye can discern about 10,000,000 colors. Audio is different though.
  There's no simple comparison of digital images and audio. You can see the color of a single pixel but a single sample always just sounds like a click. In order to hear a tone you need several evenly-spaced samples, which is also why small quantization errors do not matter...


----------



## bigshot

I'm sorry compsalot, but you are flat out wrong. I suggested it before, and I'll suggest it again. Reading a bit on the relevant wikipedia pages will clear up a lot of your misconceptions.

The difference between 16 bit and 24 bit is resolution at extremely low volume levels. The dynamic range improvement extends downward. Up in normal listening levels, 16 and 24 sound identical.


----------



## AstralStorm

Well, unless this isn't PCM. In A-law, u-law or similar companding formats, this would increase precision, but nobody who cares about sound quality really uses those.
  Heck, 16-bit A-law would probably be better than 16-bit PCM, bit for bit, especially in current loudness war campaings.
   
  That said, properly implemented 16-bit PCM is more than enough, the quantization error is quiet enough too.


----------



## compsalot

There are 3 DIFFERENT things...
   
  1) What is the SPEC?
   
  2) What is the performance of a SPECIFIC DEVICE that ATTEMPTS to implement that spec?
   
  3) What is the human ear capable of perceiving?

  
   
  What I observed throughout this entire 900+ thread is that people keep mixing those 3 things up as if they were interchangeable...  they aren't!  And that is the confused muddle that I was trying to address.
   
  To claim that 192k sounds worse than 96k is ludicrous as long as you are talking about the SPEC.  It's the same as saying that a car which goes 200 mph is SLOWER than a car that goes 100 mph.  It's a totally absurd claim.  To insist that you must record ultrasonics simply because you can record ultrasonics is equally silly.
   
  Now on the other hand if you want to say that specific device XYZ sounds like krap when you try to use it at 192k but sounds good at 96k, you will get no disagreement from me.  But to debate that the SPEC for 192k delivers worse sound then does the SPEC for 96k, this is nonsense.
   
  I'm willing to accept your claim about the ear being unable to tell the difference. I know nothing about you, but shall presume that your experience in this field grants you the ability to make that claim. 
   
  But I am also inclined to defer to the wisdom of the engineers who did extensive research when they designed the Blu-ray spec. They seemed to think it can make a difference otherwise they would not have bothered to include support for 192k in their spec.
   
  -------
  As far as 24 bits vs 16 bits, it is pointless, nay it is impossible to discuss this as long as people insist that more bits can ONLY mean LOUDER bits rather than being set to produce an equal volume range at a finer gradation; or some combination of both louder and finer (say 4 bits for each); which is a design decision of a specific implementation of the output stage.
   
  ----------
  As far as color perception goes, perhaps I can offer you a couple of data points... Because, yes, color perception is a reasonable parallel to audio perception.
   
  My current laptop has a krap monitor...  it's terribly disappointing considering that the company that makes it, made their reputation on the quality of the monitors on their top of the line laptops.  In all fairness though, I bought the ~budget~ model (which is still twice as expensive as other brands with similar specs), in the expectation that some of that color monitor goodness of the top end would percolate down to their lower end, it did not.... 
   
  If you take the time to do the research you will find that most laptop screens do in fact cheat...  they are physically incapable of delivering 24 bit color, 16 to 18 bit is probably a more realistic physical maximum.  (desktop monitors are generally better than laptops, but go find yourself an old CRT and see how many millions of colors it cannot deliver.  while you are at it don't forget to consider the quality of the video card itself -- we are dealing with systems consisting of multiple components of which the human is only one of those components, the quality/ability of each component in that system will affect the outcome).
  
  ------------------
  Next data point:
   
  It's all very fine and well to assume that ~everybody~ can easily perceive millions of colors, but my experience would strongly indicate otherwise.
   
  Back in the era before digital, I used to be a semi-pro photographer.  I got fed up with the commercial labs inability to deliver the results I wanted, so I built my own color darkroom and taught myself how to use it. 
   
  Color darkroom exposure uses a measurement called a CC = Color Correction unit.    When I started I could barely tell the difference between 30 CCs of exposure.  I had read that people who were really good at it could tell the difference between 10 CCs of exposure.  After many hundreds of hours spent in the darkroom making prints, I eventually found that I could tell the difference between 5 CCs of exposure. So, in my experience, perception can be a learned skill -- I'm sure this applies to audio as well.... also wine tasting, etc..
   
  Now here is the catch...  I would spend hours in the darkroom agonizing over the smallest changes.  But when I exhibited my work, I found that the average person could not even perceive the subtle differences, that I had felt to be so important, and spent so much time perfecting.
   
  So, if people want to argue that most ears can't tell the difference in the sound, then fine, knock yourself out, you are probably right.  If people want to claim that device XYZ does a lousy job of recording at 192k, I won't disagree with that either. But if people want to argue that the SPEC for 192k delivers WORSE sound than the SPEC for 96k.....   well, that is where I was foolish enough to stick my hand into this chainsaw of a forum and attempt -- apparently poorly -- to convey something different.

  
  I now choose to withdraw my hand from this churning tar-pit of inequity


----------



## xnor

Quote: 





compsalot said:


> To claim that 192k sounds worse than 96k is ludicrous as long as you are talking about the SPEC.


 
  That's why I was specifically talking about DAC chips.
   
  Quote: 





> As far as 24 bits vs 16 bits, it is pointless, nay it is impossible to discuss this as long as people insist that more bits can ONLY mean LOUDER bits rather than being set to produce an equal volume range at a finer gradation; or some combination of both louder and finer (say 4 bits for each); which is a design decision of a specific implementation of the output stage.


 
*It is impossible to discuss this with people that do not understand the sampling theorem, quantization and dithering. Please read the first post.*
   
  You're talking about "equal volume range". What's that range? What's the math behind the number you come up with?
   
  Hint: more bits = higher dynamic range
   
  Quote: 





> My current laptop has a krap monitor...


 
  Set a wallpaper with smooth gradients on your desktop and switch from 32 bit to 16 bit colors. Do you really not see the difference?
   
  Quote: 





> But if people want to argue that the SPEC for 192k delivers WORSE sound than the SPEC for 96k.....   well, that is where I was foolish enough to stick my hand into this chainsaw of a forum and attempt -- apparently poorly -- to convey something different.


 
  Afaik nobody argued that.


----------



## bigshot

You can assume that whenever I use the phrase "sounds like", I am referring to human hearing with ears. They're the only things I have to hear sounds with.
   
  When I talk about high bitrate audio, I am talking about a recording format with a higher dynamic range. Resolution at normal listening volumes is identical, both to human ears and to spec. All of the added resolution is down below the range your ears can hear. Handy for mixes when you need to bring up something very quiet, but for listening, it's useless as teats on a bull hog.


----------



## skamp

compsalot said:


> But I am also inclined to defer to the wisdom of the engineers who did extensive research when they designed the Blu-ray spec. They seemed to think it can make a difference otherwise they would not have bothered to include support for 192k in their spec.




Did it never occur to you that they only went for higher numbers for marketing reasons? Admitting that 16/48 was already good enough would have been detrimental to sales. They had to make people believe that higher numbers really were that much better.



compsalot said:


> As far as 24 bits vs 16 bits, it is pointless, nay it is impossible to discuss this as long as people insist that more bits can ONLY mean LOUDER bits rather than being set to produce an equal volume range at a finer gradation




*Quieter*, actually, since the point of reference is always 0 dBFS, which is equally loud at 24 and 16 bits.


----------



## skamp

compsalot said:


> it is impossible to discuss this as long as people insist that more bits can ONLY mean LOUDER bits rather than being set to produce an equal volume range at *a finer gradation*




Here's this "finer gradation" you're talking about.

Let's take the loudest signal possible, which is 0 dBFS. It is encoded as a signed 16 bit integer and a signed 24 bit integer like this:

```
16 bit: 01111111 11111111
24 bit: 01111111 11111111 11111111
```

The 16 bit value is missing all the values in between that can be encoded thanks to the lower 8 bits in the 24 bit format. So the range of values that's missing is between the following two numbers:

```
01111111 11111111 11111111
01111111 11111111 00000000
```

Now, how much difference is that? What is the _range_ of "finesse", "nuance", "precision" that is gained with 24 bit sampling? First, we have to convert those binary numbers to decimal. Then, the formula to get an equivalent value in dBFS is this:

```
20 * log10(n / (2^23 - 1))
```

Here are the converted values:

```
01111111 11111111 11111111 = 0 dBFS
01111111 11111111 00000000 = -0.0003 dBFS
```

That's a difference of 0.0003 dB! It's very, very, very small, well beyond audibility. Now, let's see how much range we're gaining at various levels:


```
01111111 11111111 11111111: 0 dBFS,
01111111 11111111 00000000: -0.0003 dBFS (0.0003 dB more range)

00111111 11111111 11111111: -6.0206 dBFS,
00111111 11111111 00000000: -6.0211 dBFS (0.0005 dB more range)

00011111 11111111 11111111: -12.0412 dBFS,
00011111 11111111 00000000: -12.0423 dBFS (0.0011 dB more range)

00001111 11111111 11111111: -18.0618 dBFS,
00001111 11111111 00000000: -18.0639 dBFS (0.0021 dB more range)

00000111 11111111 11111111: -24.0824 dBFS,
00000111 11111111 00000000: -24.0866 dBFS (0.0042 dB more range)

00000011 11111111 11111111: -30.103 dBFS,
00000011 11111111 00000000: -30.1115 dBFS (0.0085 dB more range)

00000001 11111111 11111111: -36.1237 dBFS,
00000001 11111111 00000000: -36.1406 dBFS (0.0169 dB more range)

00000000 11111111 11111111: -42.1443 dBFS,
00000000 11111111 00000000: -42.1782 dBFS (0.0339 dB more range)

00000000 01111111 11111111: -48.1651 dBFS,
00000000 01111111 00000000: -48.2329 dBFS (0.0679 dB more range)

00000000 00111111 11111111: -54.1859 dBFS,
00000000 00111111 00000000: -54.3222 dBFS (0.1363 dB more range)

00000000 00011111 11111111: -60.2071 dBFS,
00000000 00011111 00000000: -60.4818 dBFS (0.2747 dB more range)

00000000 00001111 11111111: -66.2287 dBFS,
00000000 00001111 00000000: -66.7872 dBFS (0.5585 dB more range)

00000000 00000111 11111111: -72.2514 dBFS,
00000000 00000111 00000000: -73.407 dBFS (1.1556 dB more range)

00000000 00000011 11111111: -78.2763 dBFS,
00000000 00000011 00000000: -80.7666 dBFS (2.4903 dB more range)

00000000 00000001 11111111: -84.3054 dBFS,
00000000 00000001 00000000: -90.309 dBFS (6.0036 dB more range)

00000000 00000000 11111111: -90.343 dBFS,
00000000 00000000 00000001: -138.4738 dBFS (48.1308 dB more range)
```

As you can see, the ranges are extremely small until the signal gets quieter and quieter, most (if not all) way beyond audibility.


----------



## bigshot

Blurays often have as many as 8 channels. They need a higher bitrate. The rest is marketing. Home theater folks can be as nuts about numbers as audiophiles.


----------



## dbbloke

Quote: 





xnor said:


> *No*, you need a sampling frequency greater than two times the maximum frequency to be able to reconstruct the original signal. The reconstruction is *perfect *in theory. It seems you don't understand the theorem.
> 
> 
> *No*, that's wrong. See reconstruction.
> ...


 
   
  Quote: 





xnor said:


> That's why I was specifically talking about DAC chips.
> 
> *It is impossible to discuss this with people that do not understand the sampling theorem, quantization and dithering. Please read the first post.*
> 
> ...


 

*It's impossible to discuss with people who don't listen to live acoustic music often, who haven't heard a well recorded 24/96 on a really good 2ch speaker hifi system but base their assumptions in science. :>*
   
  And to note, probably the simplest and best way to do this is to take a live instrument (like a harpsichord/harp imho), record 16-44/48 then 24-96/192 bit. Play both files back and see if the instrument sounds different in any way.
  Does one sound more detailed than the other, does the 3d staging and room dynamics sound different, does the instrument have more air about it, is it more solidly placed in the room, does one seem wider and larger, can you hear the reactions between the strings or the fingers plucking sound more finger like as it would in real life.
  Assuming same recording and playback hardware (preferably underground in a faraday cage  ) forget the track is different, the instrument sounds slightly more alive and real. At least this is what I heard, blind, back to back, levels normalised. Every time it was easy to write down what I thought and compare after doing said test a dozen or more times and find I was correct. Lucky biased guessing? I guess 1 in 4000 for a dozen comparisons is possible. What's the science on probability say? Hey, some science is mostly right, just some is more flawed then others, not all as I clearly say. I actually dropped a coin and it landed on it's side just the other day.  Think it was a zlotie though.
   
  Using some conversion software pulls so many things into the equation it makes the argument open to criticism.


----------



## chewy4

So using conversion software pulls too many things into the equation but... using two different recordings doesn't?


----------



## xnor

Quote: 





dbbloke said:


> *It's impossible to discuss with people who don't listen to live acoustic music often, who haven't heard a well recorded 24/96 on a really good 2ch speaker hifi system but base their assumptions in science. :>*


 
  You cannot know that, but regardless, it doesn't change a bit that you make claims/assumptions/statements (or whatever you like to call it) about something you don't understand and say others got it wrong.
   
   
  Quote: 





> And to note, probably the simplest and best way to do this is to take a live instrument (like a harpsichord/harp imho), record 16-44/48 then 24-96/192 bit.


 
  The problem with this has been pointed out before, but you added another variable: different sampling frequencies.
   
  Btw, tests have been done before ...
   
   
  Quote: 





> Assuming same recording and playback hardware (preferably underground in a faraday cage  ) forget the track is different, the instrument sounds slightly more alive and real. At least this is what I heard, blind, back to back, levels normalised. Every time it was easy to write down what I thought and compare after doing said test a dozen or more times and find I was correct. Lucky biased guessing?


 
  Please share the recordings used for the test and the ABX logs. Then we can try to figure out what really happened and if it was lucky guessing, some error elsewhere, or not.
   
   
  Quote: 





> I guess 1 in 4000 for a dozen comparisons is possible. What's the science on probability say?


 
  How many comparisons did you do? How many did you get right? Again, please share the ABX log if possible.
   
   
  Quote: 





> Using some conversion software pulls so many things into the equation it makes the argument open to criticism.


 
*No*, just no. /wallbash


----------



## skamp




----------



## bigshot

dbbloke said:


> *It's impossible to discuss with people who don't listen to live acoustic music often, who haven't heard a well recorded 24/96 on a really good 2ch speaker hifi system but base their assumptions in science. :>*




I listen to classical music most of the time, and I've not only heard 24/96, I've supervised recording sessions and mixes that were produced in 24/96. The difference between 24 and 16 is dynamic range, not resolution.


----------



## AstralStorm

The main limitation is the microphone equipment, those have ~85 dB SNR with 1 Pa = ~95 dB. The best ones achieve 90 dB SNR.
  Note that orchestra can almost reach 80 dB SNR in a full attack.
  The difference between 16-bit and 20-bit summed to signal (AWGN) won't move the dynamic range even by 1 dB.


----------



## XxDobermanxX




----------



## bigshot

astralstorm said:


> The main limitation is the microphone equipment, those have ~85 dB SNR with 1 Pa = ~95 dB. The best ones achieve 90 dB SNR.




Some stuff plugs in too. It's helpful to be able to boost a channel's level in the mix without pulling up noise along with it. The more dynamic range the merrier,


----------



## AstralStorm

Indeed, but that has nothing to do with bit depth, since amplification is typically analog.


----------



## bigshot

No, you usually normalize tracks up in the digital domain.


----------



## kiteki

berniew said:


> /
> I accept that 16/44.1 CD format is adequate, in theory, for almost everybody. I'm also willing to believe that some people perceive (not HEAR, but PERCEIVE) frequencies above 20kHz.


 
   
  According to one study, people can hear up to 25kHz at extreme volumes.  At normal volumes, one theory is we can perceive UHF via some other pathway, like eyes or skin, yes, which means that it wouldn't work with headphones.
   
  The original Oohashi experiment did note it didn't work with headphones, and the study has never_ really_ been perfectly replicated and refuted... that is, with fMRI, gamelan music, super-tweeters, whateverz... so the findings still stand in that sense.
   
  I suppose there are some perceptions which are too complicated to ask "yes / no" and tick boxes, you need to look inside their head hahaha.  It's like subliminal advertising... "did you just see a subliminal advertisement? Yes / No" ...............................
   
   
  Perhaps... if a room is painted blue, versus red, and you're trying to find out if the colour of a room affects peoples mood, without telling them what the study is about, you usher 20 people into different coloured rooms and make them tick boxes, you see that's difficult as well, since it's a study on emotion there are no clear-cut limits, unless you want to measure levels of cortisol in their blood after being in a red room for hours.
   
   
  Anyway there is one modern paper which 'proves' that we can perceive higher-rez material than 44.1 kHz, but the statistics aren't very good in it, I'm not sure if it's valid, in my opinion (if it's invalid, I suppose that says something about the validity of all other papers listed at the AES) --> http://www.aes.org/e-lib/browse.cfm?elib=15398
   
  What I did like about it, is they didn't use the common fast switch, time-aligned ABX, which imho is very overrated.  Someone really needs to write a new blind-test program for Foobar, without the immediate switch (like a 10 second break between A... A... A, versus A... B... B..., and not time-aligned, the sample is played from 0 - 20s, etc.).


----------



## bigshot

Quote: 





kiteki said:


> According to one study, people can hear up to 25kHz at extreme volumes.


 
   
  As pain.


----------



## Speedskater

Please note that the above paper was read at a convention.  It is not a refereed paper published in the journal.


----------



## kiteki

Yeah, it's flawed, but the Meyer & Moran study was pretty flawed too, which is also on their site.
   
  See "debunking Meyer & Moran", Moran joins the thread and gets pretty hostile at everyone.


----------



## AstralStorm

Quote: 





bigshot said:


> No, you usually normalize tracks up in the digital domain.


 

 You don't, since microphones tend to have higher dynamic range when amplified more. That is especially true of condenser mikes and other with phantom power.
   
  It's not a large difference, but it's there. So assuming you have a half decent amp and are using a condenser mike, do amplify. The idea is to have maximum gain with enough headroom.


----------



## jupitreas

dbbloke said:


> And to note, probably the simplest and best way to do this is to take a live instrument (like a harpsichord/harp imho), record 16-44/48 then 24-96/192 bit. Play both files back and see if the instrument sounds different in any way.




Nobody ever claimed that high-rez formats aren't better suited for recording, in fact, it is considered good practice to work with 24/96 (or higher) when recording, adding effects, mixing and mastering. This is because the noise floor is lower and it is possible to alter the sound more with a high-rez format without introducing audible artifacts than would be the case with a 16/44 file. 

The point everybody is trying to make here is that there is no audible improvement to using 24/96 *for playback*, not for recording. After the sound has been mastered, there is no audible difference between exporting it as a 24/96 file or a 16/44 file, which is why the high-rez format is nothing but a waste of space.


----------



## kiteki

> /
> The point everybody is trying to make here is that there is no audible improvement to using 24/96 *for playback*, not for recording. After the sound has been mastered, there is no audible difference between exporting it as a 24/96 file or a 16/44 file, which is why the high-rez format is nothing but a _waste of space._


 
   
  It used to be, it's not a waste of space on blu-ray, or 6TB external drives.


----------



## xnor

Then how about that: _waste of money_.


----------



## kiteki

Yes, since you need to buy expensive super-tweeters to avoid IMD. =p
   
  If 24-bit / 48 kHz becomes a standard for live concert blu-ray recordings that's fine with me though.


----------



## jupitreas

kiteki said:


> It used to be, it's not a waste of space on blu-ray, or 6TB external drives.




It is still a waste of space, since the files sound the same. Current hard disc sizes have nothing to do with this.


----------



## xnor

@kiteki: Instead of a higher format, that doesn't change anything, I'd rather see proper recording, mixing and mastering. Because in some genres we're down to 3 dB dynamic range. We could actually go for 8 bit files. -_-


----------



## Happy Camper

xnor said:


> @kiteki:* Instead of a higher format, that doesn't change anything, I'd rather see proper recording, mixing and mastering.* Because in some genres we're down to 3 dB dynamic range. We could actually go for 8 bit files. -_-


Yep.


----------



## bigshot

astralstorm said:


> You don't, since microphones tend to have higher dynamic range when amplified more. That is especially true of condenser mikes and other with phantom power. It's not a large difference, but it's there. So assuming you have a half decent amp and are using a condenser mike, do amplify. The idea is to have maximum gain with enough headroom.




Maybe I've been doing it wrong all these years. But I guess I wouldn't have a choice because the recording was usually done out of house where they had the best mikes and mike pres, and I would do the editing and ruff mix in house on protools.


----------



## kiteki

jupitreas said:


> kiteki said:
> 
> 
> > It used to be, it's not a waste of space on blu-ray, or 6TB external drives.
> ...


 
   
  They have nothing to do with this, in your opinion, but in reality the larger hard-disk sizes and internet speeds of today are making 24 / 192 much more popular, since people have the space they think "difference or not, nothing to lose".
   
  I guess it's like calibrating the wheels on your car at a mechanic, can they feel the difference, nope, will it make a difference nope but may as well calibrate the wheels anyway lol.


----------



## justanut

Obviously 64/256 formats are much better then! Let's open a company promoting this new format that gives you the highest possible audio quality to match those 4k tvs coming out! Nvm that each song fills a single conventional CD. Ppl will pay good money cos its higher quality than 24 /192 and has to sound better.. Esp on speakers that allow their bodies to perceive the resonance in the air! >..<


----------



## jupitreas

kiteki said:


> They have nothing to do with this, *in your opinion*, but in reality the larger hard-disk sizes and internet speeds of today are making 24 / 192 much more popular, since people have the space they think "difference or not, nothing to lose".
> 
> I guess it's like calibrating the wheels on your car at a mechanic, can they feel the difference, nope, will it make a difference nope but may as well calibrate the wheels anyway lol.




Its not an opinion, it is simple logic that states that when two files sound the same, the bigger one is a waste of space. The car analogy is also flawed as there is a rational reason to calibrate the wheels ie. maintenance of the car. Meanwhile, it is irrational to pay more for high-rez files and for larger storage to hold these files when it is scientifically evident that there is no audible improvement in sound quality.


----------



## bigshot

kiteki said:


> They have nothing to do with this, in your opinion, but in reality the larger hard-disk sizes and internet speeds of today are making 24 / 192 much more popular, since people have the space they think "difference or not, nothing to lose".




Imagine you were a millionaire and had a sixteen bedroom mansion. Wouldn't it be a waste of space if you used one of those rooms to store the packing peanuts from each and every box you got from Amazon?


----------



## chewy4

Quote: 





bigshot said:


> Imagine you were a millionaire and had a sixteen bedroom mansion. Wouldn't it be a waste of space if you used one of those rooms to store the packing peanuts from each and every box you got from Amazon?


 
  I live in a 725 sq ft apartment and I do that with a closet of mine. I would _definitely _use a room for that if I had a 16 bedroom mansion. Maybe two.
   
  Space well used IMO.


----------



## bigshot

Do you sometimes get naked and leap into your closet like a ball pit at a Chuck E Cheese?


----------



## chewy4

Quote: 





bigshot said:


> Do you sometimes get naked and leap into your closet like a ball pit at a Chuck E Cheese?


 
  Not too often.
   
  I also keep the boxes in there, and the truth is Amazon doesn't normally use packing peanuts but rather bagged air and/or paper. It's not very comfortable to jump into naked.


----------



## rtaylor76

By your logic, I should not be able to tell the difference between an analoge source, a 24-bit file, or a 16-bit file. Well, sir, I have heard these, and it is quite revealing. 
   
  Bits do matter, and nothing can sound like analoge. Closest thing I have heard is SACD, and that resolution is way more than 16 or 24 bit.
   
  So while most do not have the equipment or the ears to hear the different, there IS a difference. I would say it is not a difference of "WOW," but the difference is there.


----------



## bigshot

Analogue is easy to distinguish. Just listen for the noise or distortion.


----------



## xnor

Quote: 





rtaylor76 said:


> By your logic, I should not be able to tell the difference between an analoge source, a 24-bit file, or a 16-bit file. Well, sir, I have heard these, and it is quite revealing.


 
  There is no "_your _logic" ... but anyway. Did you compare equally mastered files? This is usually not the case when people hear differences.
   
   
  Quote: 





> Bits do matter, and nothing can sound like analoge. Closest thing I have heard is SACD, and that resolution is way more than 16 or 24 bit.


 
  SACD has a dynamic range of about 120 dB. Analogue like vinyl doesn't even come _close _to digital formats.
   
  Here's a 1 kHz sine wave (format: *44.1/16*), dithered:

   
  What music/tracks are you listening to? I bet it doesn't have a minimum RMS level below -80 dBFS.
   
   
  Quote: 





> So while most do not have the equipment or the ears to hear the different, there IS a difference. I would say it is not a difference of "WOW," but the difference is there.


 
  <rant>While most people have not seen aliens themselves there ARE aliens. You can talk to the people that got abducted. Their stories truly are WOW. </rant>
   
  Have you heard of the scientific method? It's used to distinguish fact from fiction.


----------



## ultrabike

Quote: 





xnor said:


> <rant>While most people have not seen aliens themselves *there ARE aliens*. You can talk to the people that got abducted. Their stories truly are WOW. </rant>


----------



## rtaylor76

I participated in a ADC shootout with a 1/2" master. All was done at 24 bit, then 16 bit. There was a clear winner in the converters.
   
  After it was done, the engineer played the 1/2" master straight through, no converters. It was not a question of noise, dynamic range, etc. It was a question of clarity, depth, detail, warmth, and all those other fuzzy words to describe ultimate audio. My jaw literally dropped. It had so much more life and truly sounded like the band was in the next room. This of course was with a grammy nominated engineer as well. 
   
  So I know what tech specs say, but I also know what my ears tell me. And technically there should be little difference in the 16 bit file to the analoge version, but not always. I will say that it does take a good engineer, good musicianship, and great equipment to make such music to tell the difference. It also takes better trained ears to tell the difference. I doubt my wife could tell, and I doubt a badly recorded track I could tell. JMO.
   
  And yes, different masters can make a difference. Although I do remember a similar unscientific test a friend and I did with Dave Brubeck's _Time Out_. There are several versions and we compared a CD, to a newer 180 vinyl, to the new SACD. There was little difference in the SACD to the vinyl, but the CD was clearly from a different master and was hard to tell. So therefore, I cannot definitively say that SACD is equal to vinyl. We all know vinyl has it's own issues with mono bass below 140Hz, channel separation, RIAA curves, quality of equipment, etc. But yes, there are differences in masters.


----------



## ultrabike

Quote: 





rtaylor76 said:


> I participated in a ADC shootout with a 1/2" master. All was done at 24 bit, then 16 bit. There was a clear winner in the converters.
> 
> After it was done, the engineer played the 1/2" master straight through, no converters. It was not a question of noise, dynamic range, etc. It was a question of clarity, depth, detail, warmth, and all those other fuzzy words to describe ultimate audio. My jaw literally dropped. It had so much more life and truly sounded like the band was in the next room. This of course was with a grammy nominated engineer as well.


 
   
  It could be that the equipment used to play back the recordings somehow introduced uncontrolled coloration. I wouldn't rule out the 1/2" tape equipment. Rolling off the high frequencies can result in perceived warmth, but at the expense of fidelity.
   
  It could also be that the 24 bit and 16 bit recordings where poorly or improperly re-mastered. I believe I've heard this can happen whether the engineer is/was grammy nominated or not. Dynamic range issues are more often than not a result of abusing compression.


----------



## rtaylor76

The 1/2" Ampex tape and source never changed. ADC's were plugged and unplugged with the same cables accordingly. A-B comparisons were done from the computer after encoding from the digital stream. Then after tests were done, source was pulgged through same input to console straight. There might have been some coloration in the DAC, but we used the best converter for that. The RME ADI-2.
   
  There are things going on with the analog 1/2", but that WAS the source being converted. So it should have sounded exactly the same, but it did not.
   
  And I say again, JMO. Do some tests yourself and see if you can tell the difference. If not, then be happy with 16 bit. After that session, I was forever transformed.


----------



## ultrabike

What Ampex model was used? AFAIK tape players are not necessarily free of coloration. These guys seem to have measured the frequency response of an ATR-102 (scroll to post #127): http://www.gearslutz.com/board/gear-shoot-outs-sound-file-comparisons-audio-tests/654876-ampex-atr-102-anamod-ats-1-uad-waves-processed-files-5.html
  Without the EQ, it seems to roll of the highs a little and there seems to be some low to mid bass emphasis.
   
  Might want to read the whole tread. They have some wav files comparing the HW with simulated: http://www.gearslutz.com/board/gear-shoot-outs-sound-file-comparisons-audio-tests/654876-ampex-atr-102-anamod-ats-1-uad-waves-processed-files.html
   
  When making the conversions, was there any compression, equalization, or compensation algorithm when using the RME?


----------



## rtaylor76

ultrabike said:


> What Ampex model was used? AFAIK tape players are not necessarily free of coloration. These guys seem to have measured the frequency response of an ATR-102 (scroll to post #127): http://www.gearslutz.com/board/gear-shoot-outs-sound-file-comparisons-audio-tests/654876-ampex-atr-102-anamod-ats-1-uad-waves-processed-files-5.html
> Without the EQ, it seems to roll of the highs a little and there seems to be some low to mid bass emphasis.


 
  It likely was the ATR-102, however, none of that stuff matters. The source for conversion was the output of the tape from the tape machine. So any EQ, emphasis, or otherwise mojo of that machine/tape, would have been encoded digitally. And yes, it was calibrated.
   
  The differences were not something EQ or roll-off or spectrum filtering. It was more 3 dimensional and instruments had such amazing separation and detail.
   



> When making the conversions, was there any compression, equalization, or compensation algorithm when using the RME?


 
   
  No. None of that. It was the raw mix that was bounced to 1/2" then straight off the tape into the converter. Source did not even go to a console or patch bay. It was directly plugged in and monitored off the output of the ADC, not the input.
   
  I am not trying to discredit anyone. I am just letting you know my experience. It was not a perfect or the most scientific study, but it was enough for me to realize that even 24-bit did not sound as good as the 1/2" source. When I spoke to the engineer of my dis-belief, he just says, "Now you know."


----------



## bigshot

I worked with 24 track masters and fullcoat film back around the transition to digital, and the difference between digital and analogue is in the peaks. We occasionally burned a peak into the tape and the sound didn't suffer. When we took the tape to digital, we had to drop the level or it would clip very badly. Comparing digital to analogue would require careful level matching. I would bet that the tape master was a little hotter.

If you were patched direct, the analogue was definitely hotter.


----------



## rtaylor76

Yeah, you can push tape above 0 db on the meter and get nice soft compression. Very popular and preferred for the "tape" sound and high saturation. Especially with pop and rock music.
   
  I don't know what the true levels were. It was too long ago. And yes, I understand even a slight change in volume can make a difference.


----------



## ultrabike

Compression, clipping, level matching... That doesn't sound that unscientific at all. More like someone might not have adjusted certain parameters well in the tests.


----------



## xinhang

Nice write up! So in layman term, there is no difference in quality between 16bit and 24bit? Just like all this pixel density beyond "retina" is just marketing gimmick?


----------



## jcx

lets say that it is a strong working hypothesis with no widely accepted peer reviewed counter examples - for listening to music at reasonable levels
   
  but as a negative proposition that "no one can hear a difference" - it is impossible to "prove"
   
  you can ceratinly turn up 16bit audio until noise, some dithers are audible - but then 0 dB fs SPL would drive you out of the room


----------



## jupitreas

rtaylor76 said:


> The 1/2" Ampex tape and source never changed. ADC's were plugged and unplugged with the same cables accordingly. A-B comparisons were done from the computer after encoding from the digital stream. Then after tests were done, source was pulgged through same input to console straight. There might have been some coloration in the DAC, but we used the best converter for that. The RME ADI-2.
> 
> There are things going on with the analog 1/2", but that WAS the source being converted. So it should have sounded exactly the same, but it did not.
> 
> And I say again, JMO. Do some tests yourself and see if you can tell the difference. If not, then be happy with 16 bit. After that session, I was forever transformed.




Buy you didn't do the test blind, did you? Try comparing properly converted files blind and see if the difference is noticeable.

I'm not saying it won't be, it is entirely possible that the conversion was not done properly.


----------



## rtaylor76

Quote: 





jupitreas said:


> Buy you didn't do the test blind, did you? Try comparing properly converted files blind and see if the difference is noticeable.
> I'm not saying it won't be, it is entirely possible that the conversion was not done properly.


 
   
  The files were converted properly. And yes, it was not a blind test. However, when I first heard the 1/2" without conversion, just straight in, I thought it was another converter. And I said, "What converter is that? Holy Cow!" Turns out the engineer without me knowing plugged in the tape machine direct.
   
  The comparison was to test converters, not analoge vs. digital. We did the digital converter test blind, and the RME ADI-2 was the cheapest out of the bunch. I think they were compared against a Lynx and another...maybe a Benchmark or a Mytek or a Lavry. I do know that an Apogee was not tested, however in another test I did hear Lavry vs. Mytek vs. Apogee, and Lavry and Apogee were strikingly similar and Mytek was pretty great.
   
  I would love to go back and do a blind test to see if I could still tell. That would be rather revealing.


----------



## ultrabike

The problem I see with the assessment there is that the analog tape was transferred to digital. A lousy job in the transfer and playback due to a poor choice of parameters: level matching, dynamic range loading, compression and so on, could have been responsible for rtaylor76 impressions.
  Quote: 





rtaylor76 said:


> The differences were not something EQ or roll-off or spectrum filtering. It was more 3 dimensional and instruments had such amazing separation and detail.


 
  Equalization and roll-off have an impact in instrument separation and localization.


----------



## ultrabike

Quote: 





rtaylor76 said:


> It likely was the ATR-102, however, none of that stuff matters. The source for conversion was the output of the tape from the tape machine. *So any EQ, emphasis, or otherwise mojo of that machine/tape, would have been encoded digitally. And yes, it was calibrated.*


 
   
  Actually I'm a bit confused. Was the tape player "calibrated" and an EQ applied? Did you hear the output of the player directly out of the tape player, or out of the EQ? Was the digital recording compensated using the same EQ?
   
  Quote: 





bigshot said:


> I worked with 24 track masters and fullcoat film back around the transition to digital, and the difference between digital and analogue is in the peaks. We occasionally burned a peak into the tape and the sound didn't suffer. When we took the tape to digital, we had to drop the level or it would clip very badly. Comparing digital to analogue would require careful level matching. I would bet that the tape master was a little hotter.
> *If you were patched direct, the analogue was definitely hotter*.


 
   
  Quote: 





rtaylor76 said:


> *Yeah, you can push tape above 0 db on the meter and get nice soft compression*. Very popular and *preferred for the "tape" sound and high saturation*. Especially with pop and rock music.
> 
> I don't know what the true levels were. It was too long ago. *And yes, I understand even a slight change in volume can make a difference.*


 
   
  So it seems things might have been a little hotter, levels might not have been properly accounted for, and the analog might have gone into saturation (which seems to be preferred)? and quite possibly things in the digital clipped? No wonder things sounded differently


----------



## bigshot

You don't need to EQ a four track tape to be flat. It already is.
   
  This one rings true to me. I'm positive the engineer did everything right. My guess is that he dropped the level a hair when he dubbed to digital to avoid clipping, but when he played it back, he used the same patch for both sources, so the tape was a bit louder.


----------



## ultrabike

Thanks dude. I'm not an audio engineer so I'm a little lost here. I'm still not clear what it's meant when someone says "it was calibrated" in this context.


----------



## bigshot

A couple of different things... on a tape deck it's the angle that the tape goes past the heads (that's called "azymuth" or something like that.) The bias can also be calibrated to the particular tape stock being used. The last thing you calibrate is the level. You run a test tone and use that to line the deck up to the level of the master. I believe you set the test tone lower on digital than you do analogue to protect from clipping.
   
  Response is calibrated on the other end of the chain... just before the speakers.


----------



## ultrabike

So the difference in perception here is more than likely due to volume level?


----------



## bigshot

That would be my guess. Hard to say without actually being there, but if I was an engineer patching two masters and switching between them, I'd probably patch them direct into the same pot. Even if he lined up to the reference tone on the head of the tape and patched them through two separate channels, it still might have been different because the digital master would have been transferred lower to avoid clipping. (85% of peak?)
   
  A louder volume would have made it sound punchier and more present. The bass would have been a tiny bit stronger too. The difference probably wouldn't be huge, but it would be noticeable.
   
  This is nice, because I can tell rtaylor76 is actually describing something that he actually heard. So many times I hear people describing tests like this that have the aroma of figments of the imagination concocted to make their point. (If you know what I mean.)


----------



## rtaylor76

Thanks for the vote of confidence.
   
  I doubt if it was a question of level. The ADC's used have no input level control. Basic plug and play. Many ADC's, especially high end ones, have soft or even hard limiters at 0db, so it dosen't go into digital distortion territory.
   
  And there was no EQ. Nothing plugged up between tape machine and ADC. Only XLR cables. Tape machine -> ADC -> console to stereo input. For the 1/2", the XLR cables going into the converter were then plugged straight into the cables going to the console, bypassing the converter. What I meant by EQ was tonal balance.
   
  One thing I will will bring up that you can feel free to discuss is the theory I have of resolution, detail, and volume level. We all know that digital is "stair steps," and that condition gets worse as the volume level is lower. So technically a 24-bit file could possibly have the same resolution of 16-bit because it is too low, and not taking advantage of the extra dynamic range. In this case, imagine the 24-bit file being just a tad quieter on the recording, but then level matched during playback. The 24-bit file in this case could also take advantage of that extra dynamic range in the peaks, but really it almost the same resolution of the 24-bit file. Make sense?
   
  Now on this same concept, detail is in the quieter parts. This is the area that gets more "stair steppy" and thus distorted, but then masked by dither. I have always thought that digital to me loses it in the finer details. Not because of sampling rate, nyquist theories, filtering, 20Hz cut-off, but more due to the details, the quieter parts, have less resolution. Now the same can be argued against any analog medium, say tape, that it also has noise and loses resolution in quieter passages. And I would say that is true, but it does not introduce distortion and masking the way digital does.
   
  Now back to 24 vs 16 bit per the OP - my question is, does it matter? In stereo files that are congested, maybe not to the extent that we think. Maybe we can't always tell and spot the difference. I do know that almost every recording done today is at 24 bit, but they are tracking everything and need as much detail to fit down to two stereo tracks. Does that track need to be 24 bits? Do we have the system to tell? Do we have the ears to hear it? All questions we must ask ourselves. I know I have several recordings I love that would not benefit me at all in higher resolution. It takes true talent to get and demonstrate something more out of a particular recording. And it is not just one thing, either the medium, format, tracking engineer, producer, mixing, mastering, talent, but all of it. Just as there are many more recordings I would love to hear in a higher resolution format. To me it is not just a comfort thing.
   
  I know I am new here, but as someone involved in audio for awhile, I find it hard to think many break things down to just audio spectrum or high frequencies. It is much more than that. Dimension, space, detail, impact, bandwidth, all come well before say any such high frequency information is there. A good recording should sound 3D and have depth for days. And not just wide, like deep space.


----------



## ultrabike

Quote: 





rtaylor76 said:


> Thanks for the vote of confidence.
> 
> I doubt if it was a question of level. The ADC's used have no input level control. Basic plug and play. Many ADC's, especially high end ones, have soft or even hard limiters at 0db, so it dosen't go into digital distortion territory.
> 
> ...


----------



## rtaylor76

Quote: 





ultrabike said:


> *A properly implemented DAC does not result in stair steps. Also digital is better conceptualized as digital impulses, instead of "stair steps." Digital samples, that ideally should be taken at more than twice the signal bandwidth, should be properly interpolated through a low pass filter. The resulting signal from the DAC should be very close to the original. Quantization error due to the resolution of the digital signal is a different matter (i.e. not sampling rate.) There is quite a bit of headroom in 16-bits let alone 24-bits. There is however some dependencies on the dynamic range of the ADC when digitizing a waveform.*


 
  You are talking Nyquist here, understood. However, there is still so many samples per complicated part in quieter passages.
   
*Tape and analog in general also have bandwidth limitations imposed by the pick up head and so forth. If the original sound signal does not have energy above 20kHz then nothing is lost given sufficient sampling rate. Same could be said about a 40kHz signal if sampled at say 96kHz or above. Note that sampling rate, Nyquist frequency and so on are not the same thing as quantization noise. One could assign 48-bits of resolution through an ADC that is sampling at 44.1kHz, or 16-bits of resolution through an ACD that is sampling at 192kHz. Background and thermal noise can be dominant over quantization noise.*
   
  The bandwidth limitations of the pick-up, record head, pre-emphasis eq, bias, etc. are all imposed by the very nature, not forced through filters just because. I am not worried about anything above 20k. I doubt I can hear anything above 15k. But how accurate is that high end? How fast is it? If RMAA says it can go to 30k, then to me at lower frequencies, it is more accurate. Not always the case, but most likely. And yes, system noise from background or thermal noise can be higher than quantization or dither noise. Even our own listening environments have high noise floors. However, the noise is encoded in the file and always there to mask the distortions.
   
*I think it's more a question of how the signal was processed. One can compress the life out of a recording and store it at 192 bits per sample. Compare the results with a properly recorded and dynamic range preserved recording stored at 16 bits per sample.*
   
  Agreed. I have no issues with this.
   
*When playing back a recording, dimension, space, detail, impact and bandwidth are influenced strongly by frequency response of the playback components. I however feel that these qualities are more a function of how the original recording was produced.*
   
  Here is where we can differ, and that is fine. I can see that frequency response can affect dimension, space, detail, impact and bandwidth. However, with two different speakers with the same measured frequency responses, might sound worlds different in these areas, but be actually representing the full frequency spectrum. One might sound flat, dead, and honky, and the other deep and infinite. Or one could sound in your face and full, but have no depth. Certain gear can do this as well, not just speakers.
   
  Now I do agree that how the original recording was produced also influences these factors - very heavily. That is why to always have good tracks to use to evaluate our gear with. Otherwise, what is the point?


----------



## xnor

Quote: 





> Originally Posted by *rtaylor76* /img/forum/go_quote.gif
> 
> One thing I will will bring up that you can feel free to discuss is the theory I have of resolution, detail, and volume level. We all know that digital is "stair steps," and that condition gets worse as the volume level is lower. So technically a 24-bit file could possibly have the same resolution of 16-bit because it is too low, and not taking advantage of the extra dynamic range. In this case, imagine the 24-bit file being just a tad quieter on the recording, but then level matched during playback. The 24-bit file in this case could also take advantage of that extra dynamic range in the peaks, but really it almost the same resolution of the 24-bit file. Make sense?


 
  Sorry, not to me.
   
  Quote: 





> Now on this same concept, detail is in the quieter parts. This is the area that gets more "stair steppy" and thus distorted, but then masked by dither.


 
  That's now how it works. Dither ensures that there is no quantization distortion. The level of dither, as posted just a page back, is extremely low.
  Even if you record at 24 bits you've most probably recorded noise that is magnitudes higher in level than dither.
   
  Quote: 





> I have always thought that digital to me loses it in the finer details. Not because of sampling rate, nyquist theories, filtering, 20Hz cut-off, but more due to the details, the quieter parts, have less resolution.


 
  That's why we have dither.
   
  Quote: 





> Now the same can be argued against any analog medium, say tape, that it also has noise and loses resolution in quieter passages. And I would say that is true, but it does not introduce distortion and masking the way digital does.


 
  Yeah, most likely a lot more noise, harmonic and intermodulation distortion.
   
   
  Quote: 





> Now back to 24 vs 16 bit per the OP - my question is, does it matter? In stereo files that are congested, maybe not to the extent that we think. Maybe we can't always tell and spot the difference. I do know that almost every recording done today is at 24 bit, but they are tracking everything and need as much detail to fit down to two stereo tracks. Does that track need to be 24 bits? Do we have the system to tell? Do we have the ears to hear it? All questions we must ask ourselves. I know I have several recordings I love that would not benefit me at all in higher resolution. It takes true talent to get and demonstrate something more out of a particular recording. And it is not just one thing, either the medium, format, tracking engineer, producer, mixing, mastering, talent, but all of it. Just as there are many more recordings I would love to hear in a higher resolution format. To me it is not just a comfort thing.


 
  If you take a 24 track, convert it to 16 bit and subtract one from the other and finally do a spectrum analysis you should see _*only noise *_at about -125 to -135 dBFS.
   
   
  Quote: 





> I know I am new here, but as someone involved in audio for awhile, I find it hard to think many break things down to just audio spectrum or high frequencies. It is much more than that. Dimension, space, detail, impact, bandwidth, all come well before say any such high frequency information is there. A good recording should sound 3D and have depth for days. And not just wide, like deep space.


 
  Digital audio are just a bunch of samples. If we compare sample by sample and see that the differences are at an extremely low level ... Btw, the things you mention have more to do with recording, not the format (used for playback, which this thread is about).


----------



## bigshot

Quote: 





rtaylor76 said:


> I doubt if it was a question of level. The ADC's used have no input level control. Basic plug and play. Many ADC's, especially high end ones, have soft or even hard limiters at 0db, so it dosen't go into digital distortion territory.


 
   
  That's why I think it's a level issue. A 4 track puts out a slightly hotter signal than a digital one. The reason is because a 4 track can go into the red safely and digital can't. It's standard practice when dubbing from an analogue master to digital to lower the volume slightly so the stuff that goes into the red safely on tape doesn't clip in digital.
   
  The difference in resolution between 16 and 24 at low volume levels is so far down, it's not going to be audible at normal listening levels.


----------



## ultrabike

Quote: 





rtaylor76 said:


> You are talking Nyquist here, understood. However, there is still so many samples per complicated part in quieter passages.
> 
> *Those complicated and quieter passages should be reproduced accurately through a properly implemented digital system.*
> 
> ...


----------



## jupitreas

Quote: 





rtaylor76 said:


> Thanks for the vote of confidence.
> 
> I doubt if it was a question of level. The ADC's used have no input level control. Basic plug and play. Many ADC's, especially high end ones, have soft or even hard limiters at 0db, so it dosen't go into digital distortion territory.
> 
> ...


 
  24 bits is not an increase in detail over 16 bit, it is an increase in dynamic range. The extra 8 bits are useful when recording and producing but 16 bits is more than adequate for playback. The "stair steps" analogy as used by so many people is completely wrong as it comes from a misunderstanding of signal theory.


----------



## rtaylor76

Quote: 





jupitreas said:


> 24 bits is not an increase in detail over 16 bit, it is an increase in dynamic range. The extra 8 bits are useful when recording and producing but 16 bits is more than adequate for playback. The "stair steps" analogy as used by so many people is completely wrong as it comes from a misunderstanding of signal theory.


 

 True. It has more to do with the encoding process than the playback or decoding process. You are absolutely correct.
   
  Really, a higher sampling rate is more encoded resolution. Although, 24-bits does mean higher precision accuracy and dynamic range.
   
*Two speakers than measure the same at any one location, may measure completely different at another, even the same model and brand. I'm not even including non-linear differences. Regardless of whether these speakers are "full-range" or not, they will sound different because they will have different frequency response. If you are more familiar with the term tonal balance, tonal balance is a characteristic of frequency response.*
   
  Okay. What I am trying to say is that how many bookshelf speakers measure 60-20k Hz? Nearly all of them. But we all know they sound different and some have more natural or pleasing sound. Now two speakers with the same frequency response curve, might also not sound the same. Same goes for amplifiers or any other type of equipment. That is all I am trying to say. Just because frequency response is the same means they will react in the same way.


----------



## ultrabike

Quote: 





rtaylor76 said:


> Okay. What I am trying to say is that how many bookshelf speakers measure 60-20k Hz? Nearly all of them. But we all know they sound different and some have more natural or pleasing sound. Now two speakers with the same frequency response curve, might also not sound the same. Same goes for amplifiers or any other type of equipment. That is all I am trying to say. Just because frequency response is the same means they will react in the same way.


 
   
  We can hear bellow 60Hz, and we can feel sub-bass which adds to the experience. Two speakers with the same frequency range will sound different still because they have different frequency response. Two same model and brand speakers might sound different because of manufacturing tolerances and variations. Even if we were talking about the same speaker, put it in one room and it will measure and sound different if you put it in another one.
   
  Now lets say you have the same speaker in the same room, all conditions the same. It may still sound different because one day I decided to drink coffee vs. the other day when I decided to down a full six pack of beer.


----------



## bigshot

Quote: 





rtaylor76 said:


> What I am trying to say is that how many bookshelf speakers measure 60-20k Hz? Nearly all of them. But we all know they sound different and some have more natural or pleasing sound. Now two speakers with the same frequency response curve, might also not sound the same. Same goes for amplifiers or any other type of equipment. That is all I am trying to say. Just because frequency response is the same means they will react in the same way.


 
   
  Frequency response isn't just a range. It's a balance level. The devil is in the +/- number that comes after the range. If two speakers have the same range +/- 5 dB they might not sound at all the same. If they have +/- .5 dB or less, they will sound very, very similar, if not identical.


----------



## GrindingThud

Cross thread post (did not see it referenced in this thread...apologies if I missed it), found a very enlightening article on both sample rate and bit depth:
http://people.xiph.org/~xiphmont/demo/neil-young.html
  This is worth the read and also worth listening to the files.  I'm in the 24bit matters camp because it seems to mitigate poor mastering/conversion to 16bit.   
   
  thanks to phlashbios for posting it.
http://www.head-fi.org/t/626950/24-192-audio-pointless


----------



## bigshot

Bookmarked. thanks


----------



## rtaylor76

grindingthud said:


> Cross thread post (did not see it referenced in this thread...apologies if I missed it), found a very enlightening article on both sample rate and bit depth:
> http://people.xiph.org/~xiphmont/demo/neil-young.html
> This is worth the read and also worth listening to the files.  I'm in the 24bit matters camp because it seems to mitigate poor mastering/conversion to 16bit.
> 
> ...




That first link was good for a laugh. 

Differences in speakers is a different thread, and a different forum. It is is nit all frequency response. There are many other factors to consider. Speaker specs are measured in anechoic chamber.


----------



## bigshot

My confidence is waning.


----------



## ultrabike

Quote: 





rtaylor76 said:


> That first link was good for a laugh.
> Differences in speakers is a different thread, and a different forum. It is is nit all frequency response. There are many other factors to consider. Speaker specs are measured in anechoic chamber.


 
   
  I believe you brought up speakers when you were claimed "two different speakers with the same measured frequency responses, might sound worlds different in these areas..." in regards to "dimension, space, detail, impact and bandwidth." When measuring speakers in an anechoic chamber vs a home theater guess what will be different? Yup, SPL frequency response... Seems room interactions do that to sound waves.
   
  Now, what part(s) of that article made you laugh? TBH I though it was very well put together.


----------



## zspuckl

Thank you very much, I guess that just because the number is higher doesn't mean the quality is


----------



## xnor

Quote: 





ultrabike said:


> Now, what part(s) of that article made you laugh? TBH I though it was very well put together.


 
  Seems like just another case of cognitive dissonance.


----------



## ultrabike

I guess, some folks strongly believe in "nothing can sound like analoge." However, I would love if some of these folks were a little more flexible and give other ideas, equipment, and parameters a chance. IMO there is fun in learning and understanding why things are the way they are.


----------



## bigshot

The audio industry has pretty much decided on the issue. It isn't easy to find a studio that maintains a 24 track 2 inch tape machine for anything oter than dubbing.


----------



## rtaylor76

To set the record straight, to me, just because something was recorded in analog does not mean it is superior. It is the sum of all parts that makes it so that we have already discussed and agreed upon. Some things would quite benefit.
   
  But the closer you can get to the original source is nearly always better.


----------



## ultrabike

The original source is not always analog these days.
   
  However, if we were to record an orchestra using among other things a high quality A/D and into a hard-drive, do you think we would be missing anything compared to using a magnetic head and into tape?


----------



## rtaylor76

Quote: 





ultrabike said:


> The original source is not always analog these days.
> 
> However, if we were to record an orchestra using among other things a high quality A/D and into a hard-drive, do you think we would be missing anything compared to using a magnetic head and into tape?


 

 Good question. Of course it might depend on quality of converters vs tape machine and/or tape used. There are even different ways to bias and emphasis eq the same tape formula, but lets not discuss this. You are asking about the very essence or basics of analog vs. digital mediums in the recording and playback format.
   
  I have a hard time saying no, but when you say yes, then you have to beg the question "what would be missing?" I think there would at least be some loss I think, but it is still extremely subtle.
   
  Of course you also have to figure in the coloration that the tape adds. Some say those that don't like digital and cling to analog prefer the imperfect coloration of analog. To me, there are advantages to both.
   
  Don't get me wrong, I don't thing digital is the devil and it is so terrible and analog reigns supreme. I do however find issues with those that claim there is no difference at all. That human perception can't tell. That SACD's and CD's or MP3's sound the same. Or that a $50 audio card can sound as good as a professional A/D  converter.
   
  I do know there is much smoke in mirrors in the audio world. More than any other industry. It comes from misunderstanding and placebo perceptions. I am aware. Not all of us have the same taste buds, not all of us have the same exact ears.
   
  The whole reason 24-bit or higher sampling rates were even invented is for scientific reasons of more accurate representation of the input signal vs. output without any fancy tricks with D/A conversion. But I do understand the basis of the article. "Can we actually perceive these differences sonically and without bias?" Hard to say. I am in the it does matter camp. You may not and that is fine. I do however believe it does not matter as much as some say, but a difference is a difference no matter how small.


----------



## jcx

sure, hi rez ADC of mic feed misses lots compared to tape: hiss, 3rd order saturation distortion, stick-slip/judder mechanical tape motion causing FM distortion, high and low frequency amplitude errors "head bump", aperture and azimuth error, print through, "bits" falling off the tape gumming up the machine, wearing down the head...
   
  ... then play back the tape on a different machine than used for the recording, or just some time later on the same machine - have to recalibrate the playback for speed, head azimuth, could choose wrong bias


----------



## ultrabike

I do believe there are trade-offs. I see this more clearly in digital than analog because that's what I know best. But from the limited knowledge I have in analog, I know there are very real limitations there too.


----------



## xnor

rtaylor76, care to answer what parts of the article on xiph.org made you laugh? And could you please name 24-bit tracks where you heard a difference to the 16-bit versions? Thanks.


----------



## rtaylor76

Quote: 





jcx said:


> sure, hi rez ADC of mic feed misses lots compared to tape: hiss, 3rd order saturation distortion, stick-slip/judder mechanical tape motion causing FM distortion, high and low frequency amplitude errors "head bump", aperture and azimuth error, print through, "bits" falling off the tape gumming up the machine, wearing down the head...
> 
> ... then play back the tape on a different machine than used for the recording, or just some time later on the same machine - have to recalibrate the playback for speed, head azimuth, could choose wrong bias


 

 Don't forget wow and flutter, head alignment, alignment in general, saturation and hysteresis, head magnetization, print through, adjacent track bleed, quality of tape and domains...yeah, lots of problems.


----------



## bigshot

rtaylor76 said:


> I do however find issues with those that claim there is no difference at all. That human perception can't tell. That SACD's and CD's or MP3's sound the same. Or that a $50 audio card can sound as good as a professional A/D  converter.




Have you done a controlled test to determine whether any of that is the case or not? (I have.)

I believe the most important specification is the limits of human perception. There are too many audible issues to deal with to waste time on inaudible ones. Recording is a business like any other. You want to produce the best product you can with the least wasted effort. Every minute that passes in studio costs money.


----------



## bigshot

ultrabike said:


> However, if we were to record an orchestra using among other things a high quality A/D and into a hard-drive, do you think we would be missing anything compared to using a magnetic head and into tape?




The main thing you'd be missing would be tape hiss.


----------



## rtaylor76

Yes. More than once. Sometimes there was little differences in converters, sometimes it was obvious (like in the case with the RME). I still think spending twice ore three times that might get a better converter, but not twice the quality. The differences in format or converter are extremely subtle. You have to listen closely, but you can pick up on it. It is not quite as obvious as different mics or different mic pre amps.


----------



## bigshot

Have you compared different levels of consumer playback equipment?


----------



## AstralStorm

With the 4-track tape (granted, I've only a very short listen for comparison, but very good locally made equipment, Soviet era), the readily noticeable artifact was reduced stereo separation. This cannot be avoided, as there's bleed through tape substrate between the tracks. It's quite audible, somewhat more than between various amplifiers.
   
  This alone is enough to render any listen with the tape non-blind.


----------



## bigshot

That's probably head mis-alignment.


----------



## justanut

Reminds me of the time I was contemplating a new road bike... The sales person was telling me how aero dynamic one of the models was (which incidentally cost almost double of a similarly specced other model). I was yeah... it looks good, probably will help me get to work / home that bit faster... Until my wife whispered to me "But YOU're not aero-dynamic" >..<
   
  The point is, why bother stopping people with the money and the ears that can hear the difference? Good for you if you can. As for me, I'm putting my money to better use elsewhere... on things that make obvious differences without me having to close my eyes and meditate like a monk in order to discern the difference in quality. Like getting myself into a more aero-dynamic shape


----------



## rtaylor76

Quote: 





xnor said:


> rtaylor76, care to answer what parts of the article on xiph.org made you laugh? And could you please name 24-bit tracks where you heard a difference to the 16-bit versions? Thanks.


 

 Ok ok. Fair. I need some more samples, but I cannot say that I was always over 50%. I will say that with a better headphone amp my accuracy improved dramatically to 60-70%. (I am not surprised of this as it took this same cmoy headphone amp with good headphones to really hear how bad some of my AAC files from iTunes really sounded when I was completely happy with equipment I had before). And that was straight out of my computer with headphones. It is also more difficult on a mac since it sets a global sample rate. I need to do it on a PC.
   
  I would describe the sound as a more narrow soundstage and more lacking in things like reverb cues. Also, 16-bit versions were lacking in impact. Likely due to a loss in dynamic range. All that said, it was much more subtle than I thought. I had to really concentrate. It will drive you crazy. I can see better the reasons for these articles. What your brain is not focusing on does not matter and we accept what it has given to is and most of the time 16/44.1 is damn good enough for those things. Especially with the limitations of our equipment. However, I still can't subscribe that 16/44.1 should always be used or 24/96 is pointless. There was a bit more there. Mind you, it was not in proportion to the file size, but there was a bit more. The tests I used were orchestral recordings and some single instruments. I need to do more tests with some other material and with a system that can change sample rates instead of upsampling. 
   
  I know that people like Ethan Winer and others have brought things to the table that need to be discussed. There are many myths out there that need to be busted. Things like power cords, computer cords, etc. There is a lot of voodoo in cables by itself. There are common myths all the way to other planet. Most of the myths have little to no merit and very little research and support. I however do take issue with the fact that pro's only use pro gear simply because it won't break down is not the case. The pro recording industry is built on what everybody else is using anyway. Some gear sounds similar, some of it is way off. Some consumer gear is decent, some pro gear is way overpriced. Some consumer gear (Bose) is way overpriced, some pro gear is not that much better than simi-pro gear. This isn't a discussion about gear, it's about format. 
   
  I would just say that both scientifically and sonically, there is more information in 24-bit/96k. Whether that makes a difference to you depends on many factors. I guess I still in the does matter camp, but maybe not as much as I was. It certainly wasn't as "wow" as I thought it was going to be.


----------



## stv014

Quote: 





rtaylor76 said:


> Ok ok. Fair. I need some more samples, but I cannot say that I was always over 50%. I will say that with a better headphone amp my accuracy improved dramatically to 60-70%.


 
   
  Do you mean you can successfully tell 16-bit apart from 24-bit in a blind test in 60-70% of the attempts, but "not always" 60-70% ? That is not enough to prove you were not just guessing unless the number of trials was large.


----------



## rtaylor76

There's no pleasing you folks.


----------



## xnor

I am, since you seem to be on the right track.


----------



## rtaylor76

Can any of YOU hear differences?
   
  Instead of attacking me, tell me what you hear.


----------



## skamp

rtaylor76 said:


> There's no pleasing you folks.




See this. The p value you want to get is < 0.05. Example: a positive ABX result would be 5 good guesses out of 5, or 9 out of 10, or 12 out of 16. It's not about pleasing anyone, it's just science.



rtaylor76 said:


> Can any of YOU hear differences?
> 
> Instead of attacking me, tell me what you hear.




We _don't_ hear any difference, but we're not supposed to either.


----------



## rtaylor76

Well, if you don't hear any differences, then be happy and move on. That's fine. 
   
  I'll do more tests with source material I am more familiar with, and a system that does not up-sample and skew the output.


----------



## xnor

Quote: 





rtaylor76 said:


> Can any of YOU hear differences?
> 
> Instead of attacking me, tell me what you hear.


 
  I don't think anyone's attacking you.
   
  Except for the higher noise floor if you turn up the volume to unreasonable levels you're not supposed to hear a difference. On properly dithered files that is.


----------



## bigshot

rtaylor76 said:


> There's no pleasing you folks.




We react much better when people are honest with us about what they hear and don't insist on bending the facts to fit the argument.


----------



## ertai

I've been following this thread because I wanted to learn a thing or two about audio, which I did but somehow I think I got lost in it. I must say that I also feel very intimidated by all the technical stuff here, the reason I keep quiet because I can't keep up with the discussion. Someone could just throw in a technical jargon or question and I'm flat dead.
   
  However as an 'outsider' I did start to observe and notice a pattern here and just want to share my 2 cents.
   
  No matter the technical facts, won't we all say that at the end of the day *we are the final filter* whether we perceptually do or do not 'hear' the difference?
   
  It's like the recent shootings. One event, one fact that happened, but there will always be many different and perhaps endless perception about it.
   
  In the same way, there's probably a thousand scientific facts that may PROVE that there's actually no possible audible difference in a gear, but if someone says they still hear the difference and like it, who are we to tell them that they shouldn't or don't?
   
  It's not only the equipment we hear, since our perception and placebo is always part of the equation. And I believe a smart and wise person would always acknowledge and accept it's presence because its impossible to be free from it.
   
  If swallowing a piece of chalk actually helped you quit smoking, then its great. If a person have benefited from Hypnosis and said he had increased his happiness, why do we want to debate with that guy that hypnosis is not really the thing that help him but actually himself? It's pointless. The fact that he did, and that it worked for him, its great! It's all part of the equation, and at the end, its the perceived result that matters.
   
  If it works for you, if you can HEAR the difference, then you do. Period. But if you don't, then you don't. Or if we decide to change our minds later, then sure, its up to us. What's the point of debating against someone's perception?
   
  In a bigger picture, something that works or doesn't for you may not necessarily have to apply to others. And in most cases they don't. It's like religion. It's pointless to debate who's God makes a real difference or whether they don't. The fact that they all do at some way or another, spiritually, intrinsically, personally or whatever. Because *we are the final filter*, its so personal, there's no point debating it imo
   
  Ok. I'm done. Now you can shoot and kill me now


----------



## emmodad

Quote: 





bigshot said:


> [ who provides the fertile seed for this post, thanks! ]
> 
> 
> We react much better when people are honest with us... and don't insist on bending the facts to fit the argument.


 
   
   
  [ cue wide-eyed innocent face ]
   
   
  oh my goodness.
   
   
  this couldn't possibly be interpreted to mean that there might be fact bending, misrepresentation and selective omission "to fit the argument" in the audio hobbyist and audiophile world?
   
  heaven forbid such nefarious tactics in places ie perhaps
   
  - the a/m xiph manifesto / positioning statement for a certain lossy coding implementation, timed and published by a principal of that technology in the face of pending commercial irrelevance;
   
  - the OP's ... ahem,  interesting...  presentation of digital audio and signal processing basics in this thread;
   
  - some of Those Audio Specialty magazines and a few "let's leverage the trend and most people's lack of the difficult technical knowledge" computer-audiophile-type websites with "well-known" and "highly-experienced" (and yes - very prolific in their posting and public self-promotion....errr, presence) -  writers and "founders" (holders of any actual technology background education and knowledge:  few; acquirers of clicks, pageviews, advertising revenue and high-end audio gear:  many) who coordinate with a small cabal of similar-intentioned (and often technically suspect) "specialty" equipment manufacturers.
   
   
  [ cue image of Opus in mode of holding his chest for pain of disappointment rending his ever-pure heart ]
   
   
  and as a public service reminder to all about the technical misinformation which started this entire thread:  if you'd like some reasonably-digestible and technically-correct resources about underlying theory wrt bit depth, resolution, dynamic range, sampling theory and digital signal processing in general, wander on over to check out the links in some posts earlier in this thread @ 463; 562 (incl links to references via thread "Bit depth and sampling frequency explained"; 651
   
   
   
  now let's all go lie down in a field of Dandelions...


----------



## skamp

ertai said:


> If it works for you, if you can HEAR the difference, then you do. Period.




Except you don't, really. You're imagining it, period. Yes, placebo does make a difference, but unless it's a prescription from a physician, I'd rather people be informed about it.



ertai said:


> What's the point of debating against someone's perception?




False claims should be challenged, otherwise there's no point in anything, and there wouldn't be any actual progress. Also, there's money involved. Neil Young and Co. want you to buy your music all over again for all the wrong reasons.


----------



## bigshot

Quote: 





ertai said:


> No matter the technical facts, won't we all say that at the end of the day *we are the final filter* whether we perceptually do or do not 'hear' the difference?
> 
> In the same way, there's probably a thousand scientific facts that may PROVE that there's actually no possible audible difference in a gear, but if someone says they still hear the difference and like it, who are we to tell them that they shouldn't or don't?


 
   
  We can design equipment that can perceive frequencies, light spectrums and vibrations FAR beyond what humans can perceive. Human perception is great, but there isn't much of anything there that we can't detect, measure and precisely quantify using machines. There's no reason to sweat stuff that measures on a machine, but we can't perceive, but there's no reason to think that there is anything our five senses can detect that we can't measure.


----------



## gnarlsagan

bigshot said:


> We can design equipment that can perceive frequencies, light spectrums and vibrations FAR beyond what humans can perceive. Human perception is great, but there isn't much of anything there that we can't detect, measure and precisely quantify using machines. There's no reason to sweat stuff that measures on a machine, but we can't perceive, but there's no reason to think that there is anything our five senses can detect that we can't measure.




Kiteki made a point in another thread that we have no way to measure a driver's tone. For example a way to measure differentiation between a titanium driver and a carbon nanotube driver. They sound different in some way that isn't measurable. 

It was also said that there is no way to measure the difference between a flute and a trumpet in way that would tell us which was which. They have different tone in a way that isn't measurable. 

I'm not agreeing with this at all, but I'd like to know what more knowledgeable folks than me have to say about it.


----------



## xnor

Quote: 





gnarlsagan said:


> It was also said that there is no way to measure the difference between a flute and a trumpet in way that would tell us which was which. They have different tone in a way that isn't measurable.


 
  That's like saying a person that has never heard of either instrument cannot tell which is which just by listening to the sound.
   
  Play some notes with each instrument, record that, look at the envelope and frequency spectrum for starters. That will show some clear differences.


----------



## AstralStorm

Quote: 





bigshot said:


> That's probably head mis-alignment.


 

 Nope, perfectly aligned. The stereo separation is one of the weakest point of the format.


----------



## AstralStorm

Quote: 





gnarlsagan said:


> Kiteki made a point in another thread that we have no way to measure a driver's tone. For example a way to measure differentiation between a titanium driver and a carbon nanotube driver. They sound different in some way that isn't measurable.
> It was also said that there is no way to measure the difference between a flute and a trumpet in way that would tell us which was which. They have different tone in a way that isn't measurable.
> I'm not agreeing with this at all, but I'd like to know what more knowledgeable folks than me have to say about it.


 

 Wrong, they sound different in a perfectly measurable way: different harmonic contents, specifically D2/D3/D4/D5 ratio.
  Same with flute vs trumpet. Flute has nearly no higher and odd order harmonics, while trumpet has lots.
  Also the flute has more noise, specifically pink-like noise and sharper attack than that attainable using the trumpet without overblow.


----------



## Speedskater

I definitely agree with AstralStorm!  But the difference between the two drivers will be from lots of mechanical and construction factors not just the titanium or carbon nanotube material.


----------



## AnAnalogSpirit

.


----------



## Avi

Wow, it has taken me a while, but I have finally read all 70 pages of this thread. My (non-professional) takeaways:
   

 The only difference between 16 and 24 bit is the dynamic range (distance between loudest and quietest sample). Almost all of this falls below the threshold of audibility, and even within the range of hearing, the difference in gradation given by the extra bits in 24 vs 16 is so small as to be undetectable.
 Sampling at a given rate should, theoretically speaking, allow for the perfect reconstitution of a signal with a frequency of half that rate.
 There may be some evidence that despite the above, some technical benefit is gained for playback with sampling rates greater than the 44.1/48 that should encapsulate the entirety of human auditory response.
 There is a law of diminishing returns with sampling, beyond which any possible benefits are outweighed by the potential introduction of errors and computational requirements.
 There may be a correlation between sampling/bitrate and perceived quality, but this is almost certainly due other factors, such as quality of original recording equipment, mixing, and mastering, and not the actual sampling rates and bit depths.
   
  In a nutshell:

 *Well recorded, mixed, and mastered material* (at >= 24/96) *if properly downmixed and dithered to 16/44.1 will be functionally indistinguishable from the original recording.*
 *Music should, as often as possible, be played back at the sampling rate at which it is recorded to minimize resampling errors*
 *There may be reason to believe that commercial music offered at 24/96 may offer a better experience due to the extra care that likely went into its creation, but such music, properly downsampled, should be indistinguishable from the original.*
 *Anything greater than 24/96 is overkill on the level of swatting a mosquito with a BLU-109*.
   
  On a personal note, I have seen people refer to potential benefits of higher sampling rates capturing sonic characteristics such as "airyness" or "space". I am not sure what causes the brain to perceive these characteristics, but, bottom line, if all we can pick up is compression and rarefaction of a medium in a set band of oscillations, and the digital signal can be reconstituted so that when it is applied to a membrane it causes the exact same compression and rarefaction as the original, shouldn't that encapsulate ALL the information that was encoded?
   
  <null edit>


----------



## preproman

Sub.


----------



## Chromako

I love your summary, AVI. 
	

	
	
		
		

		
			




   
  Just my contribution: I've dabbled in upmixing my music myself, and I often like it. You might laugh at me for saying this, but there is a subtle difference, in that it _seems_ like I hear more detail. Determined with ABX testing. 
   
  Let me explain:
   
  In my photography analogies, I prefer to process my photos with Photoshop's sharpening filters. Much of it helps with interpolating, and therefore, reducing effective lens fuzziness. Also, higher contrast/more edgy edges, makes details _easier_ to notice. Is the latter preserving the original picture's information? Nope. We are altering it. But it makes it easier to notice. Good for perception. Sometimes to help our imprecise senses, the original data needs to be "helped."
   
   
  Now my audio theory. I tend to prefer the "linear phase Brickwall" interpolation more. From what I can understand, it will sometimes make the waveforms more pointy (more triangular vs curvy-sine-wave-ish). While more pointy waves (ever listen to a pure triangular tone vs a sine wave?) would give a slightly harsher (to be evil sounding) or crisper (to use a nicer word) sound, giving the illusion of more detail, or perhaps emphasising details. Either or both could be true.
   
  By the way, I absolutely hated the tube amps I've tried, even quite expensive ones. Now using a tube amp, which will have slower slew rates/ reaction times (and therefore will round off the tips of triangular and squared waves) are liked by some. I'd like to propose that using files higher than 44.1khz and using a tube amp would be pointless, as their slew rates are so slow that it wouldn't make a difference. However, higher resolution files can't hurt, and I always say that hard disk space is cheap, so just go with the most information possible. Peace of mind, and it might come in handy some day. You never know. 
   
   
   
  Now increasing bit depth. 24 bit is much better in mastering and recording, as you have more leeway and flexibility when editing files. In digital photography, using 12-bit colour has saved pictures, as I can, to use a very simple example, clip off the brightest 4 bits in an underexposed file (those brightest bits were wasted in this case) and still have 8-bits of data left. Sometimes you have to do it, and you should be prepared for when you have to rescue mistakes. 
   
  For playback, humans usually can only hear 60db of dynamic range (I'd rather round to 80db for playback to be safe). 16-bit has over 95db, so it's *good enough* for our poor little ears. My cat can probably sense a larger range, but we aren't talking about cats here. 
   
  Sorry for tl;dr.


----------



## stv014

Quote: 





> Originally Posted by *Chromako* /img/forum/go_quote.gif
> 
> Now my audio theory. I tend to prefer the "linear phase Brickwall" interpolation more. From what I can understand, it will sometimes make the waveforms more pointy (more triangular vs curvy-sine-wave-ish).


 
   
  A linear phase brick wall filter (one that has constant non-zero gain up to half the sample rate, then zero above, and the same phase delay at all frequencies) is actually the mathematically "correct" way of reconstructing the continuous time signal, and turns a digital waveform of 0,1,0,-1,0,1,0,-1,... into a perfect Fs/4 sine wave. In practice, it does not have an infinitely steep roll-off, because that would require using an infinitely long impulse response, and a less steep roll-off also reduces ringing.


----------



## jtinto

I've read this entire thread too 
	

	
	
		
		

		
		
	


	



   
  I tend to let my ears and my speakers and headphones guide me
  Right now, I fall firmly into the 24bit supporters camp
  Granted, most of my digital library is 16bit ripped from CDs, but my 24bit HD downloads sound superior
  That could be due to the masters used to produce those versions, but ... I'm not so sure
  FWIW my computer system is pretty revealing: Mac Mini > Stelllo U3 > W4S DAC-2
   
  P.S. the whole 44/48 vs 88/96 KHz is still a wide open discussion for me


----------



## phidauex

Well, having skimmed much of this thread, I'm not going to comment on people's perceptions...
   
  However - for anyone who actually wants to learn about signal processing techniques, there is a great opportunity coming up soon - *starting in February is this great free course on Digital Signal Processing: https://www.coursera.org/course/dsp*
   
  Coursera is a free online program that partners with universities to offer some of their courses in a free online format, where you learn alongside the "real" students, meaning very high quality courses. You don't get college credit for it, but do get a certificate of completion. The class is 8 weeks, and includes electronic textbooks.
   
  The professors teaching it are well known and respected in the field, and the course is intended to cover signal processing theory (rather than specific implementation) which makes it very applicable to those of us who are interested in the concepts as they apply to DSP chip, CPUs, etc. Highly recommended for anyone who thinks they know a lot about this stuff, or is aware that they don't. 
   
  -Sam


----------



## jtinto

Thanks for the heads-up phidauex


----------



## applebook

No need to read entire thread after the first few pages.
   
*Dan Lavry*: http://www.monoandstereo.com/2008/06/interview-with-dan-lavry-of-lavry.html
  "Regarding bits: The ear can not hear more then about 126dB of dynamic range under extreme conditions. At around 6dB per bit, that amounts to *21 bits*, which is what my AD122 MKIII provides (unweighted).

 Regarding sample rate: The ear can not hear over 25-30KHz, *therefore 60-70KHz would be ideal*. Unfortunately there is no 65KHz standard, but 88.2KHz or even 96KHz is not too far from the optimal rate."
   
*Bob Katz*: (at whose feet the OP allegedly worships): http://www.tnt-audio.com/intervis/digidoe.html
  "My conclusions are that the *wordlength increase is the most dramatic improvement*, with the *sample rate increase being a secondary factor*.
 I like the results, but after a number of careful tests, I feel that *44.1 kHz/24 bit can be considerably improved, never as good as 96/24*, but a lot better than people think. I am displeased with the losses I have heard with the current generation of sample rate/wordlength converters, when my 96/24 recordings were reduced to 44.1K. These converters are constantly improving, and you have not yet heard these recordings at their best in the CD format."  

 The Katz interview is fairly old now, and even though A/D converters have improved dramatically, Katz still prefers 24-bit overall because of its superior resolution. On October, 18th 2012: http://www.digido.com/
  In regards to 24-bit, "The *improvements with higher sample rates go down exponentially with each higher rate*. The most *significant improvement comes from moving to 48 kHz from 44.1 kHz*. 88.2 and 96 k are not "twice the improvement" of 48 k, but *they are a meaningful improvement*."
   
  Anonymous internet poster < Bob Katz, Dan Lavry


----------



## skamp

applebook said:


> Anonymous internet poster < Bob Katz, Dan Lavry




Christopher Montgomery (Monty, creator of Ogg Vorbis) > Bob Katz, Dan Lavry


----------



## AstralStorm

I wonder how Lavry came to the conclusion that human ear can actually hear anything at 25 kHz.
  Scientific consensus is that the extreme is 21 kHz. (actually something right beyond 20 kHz)
  This means a nearly perfect reconstruction filter is possible from 44100 Hz, which has a bandwidth of 22 kHz. There's even 4% rolloff to work with...
  Similarly, Katz was talking about the reconstruction filters of the time, usually too cheap (analog?) and too rolled off.
  Still, why is he holding onto his opinion now that the filters are much more refined in the better hardware?
   
  Or are they advocating the analog filters, which are minimum phase, but have phase shift at the highest frequencies?
  These become viable only when 48kHz is used, otherwise there is major phase shift in audible spectrum.
   
  I'd like to see a controlled experiment.


----------



## chewy4

Not only does it not make sense that he said humans can hear up to 25-30KHz, but how on earth does that equate to an up to 70KHz sample rate?


----------



## Gorillaz

the funny thing is that people who claim they hear a difference between 24/96 or 16/44 or between Flac, wav or mp3 will probably fail a blind test., I've downloaded HD tracks  of some mp3 song that I have to check if I could hear a difference, and there was no difference even the piano instructor at my college couldn't detect any difference and he knows one or two things about music, and I think if you are paying a lot of money for HD tracks you should hear the diference right off the bat, and this goes to the people who thinks that cables and burn-in make any difference too.


----------



## rtaylor76

Quote: 





astralstorm said:


> I wonder how Lavry came to the conclusion that human ear can actually hear anything at 25 kHz.


 
   
  My thought, and what I have heard from some others, is that has more to do with the filtering involved rather than how high the sampling. Higher sampling rates get the advantage of being able to use less of sloped filter adding less phasing and combing artifacts at lower frequencies. 
   
  The same idea can be said of better A/D converters. Better ones have better, more accurate filtering with fewer effects. As well as better beter input sections and power supplies and overall components. Just saying that filtering is a big part of a good A/D filter vs great ones.
   
  Isn't it also legend of Rupert Neve making mic preamp's sounding better by removing the 20k filter cap. Not because it was restricting anything, but because of the artifacts it was creating at lower frequencies and overall quality of sound.
   
  I am not trying to say this explains everything, but something I have not seen discussed in this thread. I know I can't hear anything past 15k. Some people like to talk about psychoacoustics and perception, like "your eardrum hears it, but the rest of your ear can't." Or, "eventhough you can't hear above 15k, you CAN perceive it." I don't subscribe to this, but scientifically there is more evidence in filtering artifacts. This might explain at least some of it.


----------



## stv014

Quote: 





rtaylor76 said:


> My thought, and what I have heard from some others, is that has more to do with the filtering involved rather than how high the sampling. Higher sampling rates get the advantage of being able to use less of sloped filter adding less phasing and combing artifacts at lower frequencies.


 
   
  Can you hear the difference ? Phasing or combing artifacts ? Try it with the foobar2000 ABX comparator:
   
c.flac
d.flac
   
  One file is the 96/24 format original, and the other has been converted to 44.1/16, and then back to 96/24. The upsampling used a linear phase FIR filter that approximates what you would find in a decent real DAC.


----------



## Kees

Quote: 





stv014 said:


> Can you hear the difference ? Phasing or combing artifacts ? Try it with the foobar2000 ABX comparator:
> 
> c.flac
> d.flac
> ...


 
  I don't know what is what, but the d.flac file sounds much better to my ears. More space and more detail. The c.flac file sounds comparatively compressed.


----------



## stv014

Quote: 





kees said:


> I don't know what is what, but the d.flac file sounds much better to my ears. More space and more detail.


 
   
  It is the one degraded to CD quality.


----------



## Avi

Son of Xnor?


----------



## bigshot

Test after test and they're still surprised and dismayed by the results when they finally get around to doing the test themselves.


----------



## Avi

I still think there is some truth to the correlation of sound to bit/sampling based on the mastering and production (not the ears) but that is just my extreme layman's opinion.
   
  I'd like it if Xnor or someone would do that test at least quarterly.
   
  Heck, then it can be written up in some journal


----------



## Kees

Do other people here hear a difference? 
  If so, how did it get there?


----------



## stv014

Of course, those who claim to hear a difference hopefully did use an ABX comparator, and can post the (not fake) logs ?


----------



## GrindingThud

I tried really really hard for a long time, with and without headphones......I can not discern even a placebic difference.


----------



## k4klcc

thanks for sharing...


----------



## applebook

Quote: 





skamp said:


> Christopher Montgomery (Monty, creator of Ogg Vorbis) > Bob Katz, Dan Lavry


 
  LOL. 
   
  So some software programmer is more credible than Bob Katz (whom Monty regards as THE authority in digital mastering) when it comes to digital audio? OK, sure. 
   
   
  [size=x-small]Lavry chose 75KHz as the sampling frequency because it's slightly more than double the 35KHz that he thinks some people can hear. The Nyquist theorem calls for sampling rates at twice the audible frequency range. I would certainly trust Lavry over a software programmer who has very paranoid worldviews in general (see his website). [/size]
   
  [size=x-small]It's interesting that Monty lumps 192KHz into all encompassing category for anything about 44.1Hz, even taking Lavry out of context in attempt to back up his proposition that higher frequencies are inherently worse. Many 24-bit supporters opt for 96KHz, not 192. [/size]
   
  [size=x-small]Anyway, I do empathize with a lot of people in regards to the 192KHz and 24-bit DAC trend. Since the popularity of the Benchmark DAC-1 - which I owned for years with lots of components, including Qualia, K1000, OMEGA II, $5000 speakers, etc. - every manufacturer has gone with 24/192 upsampling, which in many cases, make 16-bit recordings sound worse, not the same but worse. There are exceptions, but generally I prefer NOS. My favorite DAC is the CIAudio VDA-2 because it doesn't upsample 16-bit signals and simply sounds more natural, without artificial enhancement. 16-bit can be pretty good after all.  [/size]


----------



## applebook

Quote: 





stv014 said:


> It is the one degraded to CD quality.


 
  He shouldn't be able to hear ANY difference though, right?


----------



## applebook

Quote: 





astralstorm said:


> I wonder how Lavry came to the conclusion that human ear can actually hear anything at 25 kHz.
> Scientific consensus is that the extreme is 21 kHz. (actually something right beyond 20 kHz)


 
   
  I'd like to know as well because most people cannot hear clearly above 18KHz. 
   
  However, the human ear can definitely distinguish dynamic range above the 16-bit threshold. The sonic effects are small, but remember that to true audiophiles, even a 5% difference is enormous. People like Monty probably consider 5% to be nothing --thus their dogmatic conclusions.


----------



## chewy4

Quote: 





applebook said:


> He shouldn't be able to hear ANY difference though, right?


 
   
  If it's a sighted test and he believed an audible difference exists, then yeah he should be able to. Expectation bias is a real thing, proven time and time again.
   
  Quote: 





applebook said:


> LOL.
> 
> So some software programmer is more credible than Bob Katz (whom Monty regards as THE authority in digital mastering) when it comes to digital audio? OK, sure.
> 
> ...


 
  People can hear up to 35KHz now???
   
  I'm sorry, but you have to be insane to believe that.


----------



## xnor

Quote: 





applebook said:


> [size=x-small]Anyway, I do empathize with a lot of people in regards to the 192KHz and 24-bit DAC trend. Since the popularity of the Benchmark DAC-1 - which I owned for years with lots of components, including Qualia, K1000, OMEGA II, $5000 speakers, etc. - every manufacturer has gone with 24/192 upsampling, which in many cases, make 16-bit recordings sound worse, not the same but worse. There are exceptions, but generally I prefer NOS. My favorite DAC is the CIAudio VDA-2 because it doesn't upsample 16-bit signals and simply sounds more natural, without artificial enhancement. 16-bit can be pretty good after all.  [/size]


 
  There are couple of things I want to ask, but let's start simple. Who says that DAC doesn't upsample and what do you mean by upsampling exactly? Why do you think a lossless conversion like from 16 to 24 bits is an artificial enhancement? What makes you think that a signal left-shifted by 8 bits sound less natural?


----------



## spaark

Quote: 





applebook said:


> However, the human ear can definitely distinguish dynamic range above the 16-bit threshold. The sonic effects are small, but remember that to true audiophiles, even a 5% difference is enormous. People like Monty probably consider 5% to be nothing --thus their dogmatic conclusions.


 
  If you turn the volume way up, then yes. I don't think anybody listens at such levels to be able to hear the noise floor of 16-bit audio. Edit: Sorry, I didn't realize you were talking about dynamic range. My reply to that is still the same, though. You'd have to be listening pretty loud to notice, and of course the audio would have to have a DR greater than 96 dB.


----------



## sonitus mirus

With modern dithering techniques, you would need to crank up the volume to that of a jet engine in all but the quietest of environments.  Besides, what music has a dynamic range approaching 96dB?  I know my pathetic AC/DC CDs are probably pushing 10dB max range.


----------



## skamp

applebook said:


> So some software programmer is more credible than Bob Katz (whom Monty regards as THE authority in digital mastering) when it comes to digital audio? OK, sure.




Hell yes. He knows a hell of a lot more about digital audio than any recording or mastering "engineer".



applebook said:


> However, the human ear can definitely distinguish dynamic range above the 16-bit threshold. The sonic effects are small, but remember that to true audiophiles, even a 5% difference is enormous. People like Monty probably consider 5% to be nothing --thus their dogmatic conclusions.




I don't know what a percentage would mean in this context. Actually, the difference between 16 bit and 24 bit at audible volumes down to -54 dBFS is less than 0.1 dB; less than half a decibel at -66 dBFS; 1 dB at -73 dBFS and 2.5 dB at -80 dBFS, as I demonstrated in a previous post in this thread. The actual differences are so ridiculously small or so ridiculously quiet, good luck trying to ABX that.


----------



## bigshot

Was Bob Katz the engineer that worked with Sheffield Lab back in the direct to disk vinyl days? Sheffield Lab did great sounding records, but they put out a lot of self serving anti digital propaganda about the sampling rate being too low compared to vinyl. They ended up eating those words and releasing a line of CDs.


----------



## wuwhere

Quote: 





bigshot said:


> Was Bob Katz the engineer that worked with Sheffield Lab back in the direct to disk vinyl days? Sheffield Lab did great sounding records, but they put out a lot of self serving anti digital propaganda about the sampling rate being too low compared to vinyl. They ended up eating those words and releasing a line of CDs.


 
   
  It was Doug Sax.


----------



## bigshot

Katz-Sax. My brain is turning to jello!
   
  There were a couple of Katz having Sax outside my bedroom window last night!


----------



## spagetka

Hi, I'd love to try. Something dynamic, musical, jazz/classical....

 Patricia Barber - Companion (2000)  - #1The Beat Goes On xxx ~~~
 Patricia Barber - Modern Cool (1998) - #4 Constantinople ~~~
 Hugh Masekela - Hope (2004) - #Stimela 12
 The Dave Brubeck Quartet - Time Out (1959/r2013) - #3 Take five *** ~~~
 Anne-Sophie Mutter - Carmen-Fantasie (1993) - #6 Sarasate: Carmen Fantasy, Op.25: 1. Moderato
 Norah Jones - Come Away with Me (2012) *** ~~~
 David Chesky - Area 31 (2005) ~~~
 Johnny Cash - American IV: the Man Comes Around (2002) ~~~
 Jimmy Cobb Quartet - Jazz in the Key of Blue (2009) *** ~~~
 Collegium musicum - Collegium Musicum - Speak, Memory (2010) ~~~
 Al Di Meola, John Mclaughlin, Paco De Lucia - Friday Night in San Francisco (1981) *** ~~~
   
  *** I have different versions
  ~~~ - preferable
   
  Thanks, M.
   
  Quote: 





stv014 said:


> Can you hear the difference ? Phasing or combing artifacts ? Try it with the foobar2000 ABX comparator:
> 
> c.flac
> d.flac
> ...


----------



## stv014

Well, just upload a 30 second sample of whatever you think would be best for the test, and I will create the processed files.


----------



## spagetka

Hi, limit is 5MB which is 14s of 96/24 file
   
     
  Thanks, have a good day.
   
   
  Quote: 





stv014 said:


> Well, just upload a 30 second sample of whatever you think would be best for the test, and I will create the processed files.


----------



## stv014

Here are a few filtered files you can try to ABX against each other and/or the original sample. They were created by downsampling the source to 44.1 kHz (using a long linear phase FIR filter with very fast roll-off), and then upsampling back to 96 kHz using various lowpass filters. Of course, these are only a few random examples of the infinite possible filter responses; perhaps I should also have included a filter with a very slow roll-off, and some loopback recordings from real DACs. Also, the sample is actually not ideal for this purpose, because it does not contain much high frequency information.
   
filters.zip
  
  Frequency response (left) and group delay (right, 1 dB = 1 ms):
   
    
   
*A_lp* (red): no imaging, short impulse response, linear phase
*A_mp* (green): no imaging, short impulse response, minimum phase
*B_lp* (not shown): long impulse response (very fast roll-off), linear phase
*B_mp* (cyan): long impulse response (very fast roll-off), minimum phase
*C_lp* (yellow): no roll-off below 22 kHz, short impulse response, linear phase
*C_mp* (blue): no roll-off below 22 kHz, short impulse response, minimum phase


----------



## jaddie

Quote: 





bigshot said:


> Was Bob Katz the engineer that worked with Sheffield Lab back in the direct to disk vinyl days? Sheffield Lab did great sounding records, but they put out a lot of self serving anti digital propaganda about the sampling rate being too low compared to vinyl. They ended up eating those words and releasing a line of CDs.


 
  I still use tracks from the Lincoln Mayorga recordings for evaluating systems.  Admittedly, they've been digitized...sorry Sheffield/Doug.  I had the chance to use their plating facilities for a couple of projects, pretty good work.


----------



## spagetka

[size=10pt]Thank you for the files. I agree that 14s sample file is not so good for expetiments. However, I will try to test it the week from 3rd March to 9th March.[/size]

 [size=10pt]AFAIK FIR filter causes the oscillation. The thing is how many oscillations are "visible". For me FIR filter is something which can mask some things (and sometimes I like it ) but in the end I always got back to pure NOS DAC. But it is nice to have this possibility.[/size]

 [size=10pt]I have played with up/down sampling in JRiver (DSP & output format) and the sound was slightly changed.[/size]

 [size=10pt]In the past I have tested the files 16bit vs 24bit via Foobar ABX [/size]-> *post #688*
  Br,M.
   
  Quote:


stv014 said:


> Here are a few filtered files you can try to ABX against each other and/or the original sample. They were created by downsampling the source to 44.1 kHz (using a long linear phase FIR filter with very fast roll-off), and then upsampling back to 96 kHz using various lowpass filters. Of course, these are only a few random examples of the infinite possible filter responses; perhaps I should also have included a filter with a very slow roll-off, and some loopback recordings from real DACs. Also, the sample is actually not ideal for this purpose, because it does not contain much high frequency information.
> 
> filters.zip
> 
> ...


----------



## stv014

Quote: 





spagetka said:


> [size=10pt]Thank you for the files. I agree that 14s sample file is not so good for expetiments.[/size]


 
   
  It is not because of the length that I think the samples are not ideal (if the sample is chosen right, even a few seconds can be enough to reveal a particular artifact), but rather because they are lacking the frequencies (> 10 kHz) where the filters would have the most chance to actually make a difference.
   
  Quote: 





spagetka said:


> [size=10pt]AFAIK FIR filter causes the oscillation. The thing is how many oscillations are "visible". For me FIR filter is something which can mask some things (and sometimes I like it ) but in the end I always got back to pure NOS DAC. But it is nice to have this possibility.[/size]


 
   
  Well, the "B" filters with the very fast roll-off have rather long ringing, they use a 1 second impulse response. Can you hear it ? However, a NOS DAC will not eliminate ringing that is caused by the A/D conversion, or, in this particular case, the downsampling. For that, the best choice is in fact the "A" filter that has (almost) no imaging above 22.05 kHz, and the roll-off is as slow as possible while still keeping the attenuation at 20 kHz within 0.1 dB. The downsampler already used a 1 second long linear phase FIR filter that pre- and post-rings at 22.05 kHz. That can only be "dampened" to a shorter length if the upsampling filter has a very high attenuation already at 22.05 kHz, and its impulse response is short enough (shorter IR = slower roll-off).


----------



## spagetka

I have to say, it is pretty hard
   
  test done (nb dell e6420 + AKG K701) - no external dac or amp

   
 Quote: 





stv014 said:


> It is not because of the length that I think the samples are not ideal (if the sample is chosen right, even a few seconds can be enough to reveal a particular artifact), but rather because they are lacking the frequencies (> 10 kHz) where the filters would have the most chance to actually make a difference.
> 
> 
> Well, the "B" filters with the very fast roll-off have rather long ringing, they use a 1 second impulse response. Can you hear it ? However, a NOS DAC will not eliminate ringing that is caused by the A/D conversion, or, in this particular case, the downsampling. For that, the best choice is in fact the "A" filter that has (almost) no imaging above 22.05 kHz, and the roll-off is as slow as possible while still keeping the attenuation at 20 kHz within 0.1 dB. The downsampler already used a 1 second long linear phase FIR filter that pre- and post-rings at 22.05 kHz. That can only be "dampened" to a shorter length if the upsampling filter has a very high attenuation already at 22.05 kHz, and its impulse response is short enough (shorter IR = slower roll-off).


----------



## spagetka

forgot to attach report file...


----------



## scuttle

Quote: 





mark_h said:


> Soooo, if, as others have mentioned in different threads, they cannot hear a difference between 320 mp3 and .wav and now in this thread people are saying there is no discernible difference between 16 or 24 bit then it stands to reason there is no difference in sound quality between 320mp3 and 24bit studio quality?


 
   
  I think the "soooo" is supposed to indicate that you think you have a clever point; this was a mistake....
   
  Studio files are inputs for mixing and EQing, which is why 24 bits are useful - not because as you seem to think that more bits equal more quality (they don't) but because they given more volume range, which makes it easier to mix sources with different volumes together.


----------



## scuttle

Quote: 





sonitus mirus said:


> With modern dithering techniques, you would need to crank up the volume to that of a jet engine in all but the quietest of environments.  Besides, what music has a dynamic range approaching 96dB?  I know my pathetic AC/DC CDs are probably pushing 10dB max range.


 
   
  Out of any reasonably well-known band, the Pixies are probably the dynamic range champions with a 14db range on boutique pressings: http://www.dr.loudness-war.info/details.php?id=23683


----------



## spagetka




----------



## JoelMC

Scuttle pretty well nailed it. 24bit is used in recording to give you as much headroom as possible. With 16bit you really have to record right in the sweet spot to get maximum resolution, but with 24bit you can be a little more conservative with levels without having to worry about clipping. This is especially important with dynamic sources, like a soprano for instance, who will light up the clip light before you knew what happened. Audio processing within a DAW is also done at or above 24bit as well. 24bit is a tool to give audio engineers more room to work.


----------



## scuttle

Quote: 





joelmc said:


> Scuttle pretty well nailed it. 24bit is used in recording to give you as much headroom as possible. With 16bit you really have to record right in the sweet spot to get maximum resolution, but with 24bit you can be a little more conservative with levels without having to worry about clipping. This is especially important with dynamic sources, like a soprano for instance, who will light up the clip light before you knew what happened. Audio processing within a DAW is also done at or above 24bit as well. 24bit is a tool to give audio engineers more room to work.


 
   
  It's really simple: the extra cost of recoding in 24 bits is peanuts is compared to the alternatives of more careful miking and more repeat performances. 24 bit recordings mean that you spend less technician time miking, and need fewer repeat performances, saving on musician costs and studio time.


----------



## jaddie

scuttle said:


> It's really simple: the extra cost of recoding in 24 bits is peanuts is compared to the alternatives of more careful miking and more repeat performances. 24 bit recordings mean that you spend less technician time miking, and need fewer repeat performances, saving on musician costs and studio time.




 No bit depth compensates for proper micing. 24 bits gets you fudge room if you get levels wrong, but that's it.

Since it hasn't been mentioned here lately, there are no affordable 24bit ADCs that offer true 24bit performance. The noise floor usually stands at a solid 18-20 bits.


----------



## scuttle

jaddie said:


> No bit depth compensates for proper micing. 24 bits gets you fudge room if you get levels wrong, but that's it.


 
   
  Yes, that was my point. If only because what the rest of proper miking is beyond getting the levels right I have no idea. Probably a good idea not to have the cables immersed in water - or is that the wrong way around and the cables should be immersed? Whatever.


----------



## stv014

Quote: 





spagetka said:


> I have to say, it is pretty hard
> 
> test done (nb dell e6420 + AKG K701) - no external dac or amp


 
   
  Have you also tried comparing the resampled files to the original one ?


----------



## jaddie

scuttle said:


> Yes, that was my point. If only because what the rest of proper miking is beyond getting the levels right I have no idea. Probably a good idea not to have the cables immersed in water - or is that the wrong way around and the cables should be immersed? Whatever.




Immerse the artist and the mic. Oh, wait, only if you're John Lennon. Level adjustment is the final step and the result of proper micing, which is different for each instrument, perspective and music style. RE-20 3" from a kick drum is good, so is a spaced pair of KM-183's flown for choral ambience.


----------



## spagetka

Quote: 





stv014 said:


> Have you also tried comparing the resampled files to the original one ?


 
  No, because I preferred totally random tests and based on logic you have described I thought that all files are more or less equivalent... But of course I can do that next week.


----------



## stv014

Quote: 





spagetka said:


> No, because I preferred totally random tests and based on logic you have described I thought that all files are more or less equivalent...


 
   
  Well, the scores you have posted so far do not really prove otherwise, since the probability of guessing needs to be under 5% for a positive result, and you got 5.5% and 17.2%. However, your total score of 19/28 (all three tests combined) does give a 4.4% result, although I do not know if you had any other unsuccessful attempts, and only posted the best ones, or these 28 trials were really all ?


----------



## Speedskater

Let's not co-mingle what equipment is appropriate for recording and what is audible in playback.
   
  Decades ago, I did some recording using the Soundstream.  We could have used some more bits, yes we had some overs!  But the session cost $175 per minute so we couldn't do a lot of level adjusting.


----------



## spagetka

Look, it is only about the concentration. Of course I had unsuccessful attempts before because I am not the robot and which is more important I try to find the weak point in random part of the test track and of course it takes a while to have ears adapted.
   
  I do not want to prove anything. I just know that there is a difference.
   
  8/10 is definitely not guessing and even 7/10 quite enough in ABX test.
   
  Could you do me a favour? Please let me know (PM) if you have time - I have a small part of the track I would like to know what is there. Thank you for your time.
   
  M.
   
  Quote: 





stv014 said:


> Well, the scores you have posted so far do not really prove otherwise, since the probability of guessing needs to be under 5% for a positive result, and you got 5.5% and 17.2%. However, your total score of 19/28 (all three tests combined) does give a 4.4% result, although I do not know if you had any other unsuccessful attempts, and only posted the best ones, or these 28 trials were really all ?


----------



## stv014

Quote: 





spagetka said:


> Look, it is only about the concentration. Of course I had unsuccessful attempts before because I am not the robot and which is more important I try to find the weak point in random part of the test track and of course it takes a while to have ears adapted.


 
   
  That is fine, but cherry picked results are not suitable for calculating an overall score (like combining the two "good" tests into 15/20).
   
  Quote: 





> Originally Posted by *spagetka* /img/forum/go_quote.gif
> 
> 8/10 is definitely not guessing and even 7/10 quite enough in ABX test.


 
   
  7/10 is surely not enough, and 8/10 is still above the p-value of 0.05 which is the "standard" threshold for a statistically significant result. Given that for the "7/10" test you also had an aborted 4/8 score, I do not think that one is good enough. 8/10 is marginal, and needs more testing for a more definite result either way.
   
  Did you find some artifact in any of the files that you specifically listen for when performing the test ?


----------



## sandslash

Whereas 16-bit is enough for the playback format, I argue that if one uses digital volume control it is still better to use a 24-bit DAC. Every time you halve the digital volume control, you truncate one bit of dynamic range. Depending on the peak listening volume of yours, if you have the downstream amplification at high gain and digital volume control really low before it, the quantization error can become an audible problem at some point if you use a 16-bit DAC.


----------



## xnor

Quantization distortion is eliminated using dither. But I agree, high gain after lots of digital attenuation will cause the noise floor to become audible.


----------



## Speedskater

In another thread, in another forum, on another day they discussed volume controls.  It seems that many components volume controls aren't what they seem. Some are true digital, some are digital controlled analog, a few are analog and some are combinations.
  The problem gets deeper when one control is used to adjust both stage gain and volume.


----------



## spagetka

I am sorry but I do not understand what is going on here. Maybe I am still missing something here.
   
  I chose the track and you filters and prepared the files. As you said before, filters should not make the difference because sample does not contain much information above 10kHz.
   
  First, try to find 2 people with the score 7 and more out of 10 and then we can talk about (non) significant result. Or even try to pass the test with random selects and hope that you will be lucky.
  Second, of course I found artifacts (do not know if it is the right word but for me it is "something" different) otherwise I think that I cannot pass the ABX test.
  Third, I cannot identify which track is currently playing, I can say that there is a difference between them. 
  Fourth, and for me the most important, we are discussing the difference between the files based on listening not trial/error method.
   
  Sorry for my bad English, have a good day.
   
  Quote: 





stv014 said:


> That is fine, but cherry picked results are not suitable for calculating an overall score (like combining the two "good" tests into 15/20).
> 
> 
> 7/10 is surely not enough, and 8/10 is still above the p-value of 0.05 which is the "standard" threshold for a statistically significant result. Given that for the "7/10" test you also had an aborted 4/8 score, I do not think that one is good enough. 8/10 is marginal, and needs more testing for a more definite result either way.
> ...


----------



## chewy4

Quote: 





spagetka said:


> Or even try to pass the test with random selects and hope that you will be lucky.


 
  Ok, I just did.
   
   
_foo_abx 1.3.4 report_
_foobar2000 v1.1.14a_
_2013/03/10 13:29:48_
   
_File A: C:\Users\Computer\Music\[demonoid[Mount Kimbie - Crooks & Lovers (2010) [320]\01 Tunnelvision.mp3_
_File B: C:\Users\Computer\Music\[demonoid[Mount Kimbie - Crooks & Lovers (2010) [320]\02 Would Know.mp3_
   
_13:29:48 : Test started._
_13:29:49 : 01/01  50.0%_
_13:29:50 : 02/02  25.0%_
_13:29:52 : 03/03  12.5%_
_13:29:53 : 03/04  31.3%_
_13:29:55 : 04/05  18.8%_
_13:29:56 : 05/06  10.9%_
_13:29:58 : 06/07  6.3%_
_13:29:59 : 07/08  3.5%_
_13:30:00 : 08/09  2.0%_
_13:30:02 : 08/10  5.5%_
_13:30:05 : Test finished._
   
_ ---------- _
_Total: 8/10 (5.5%)_

   
  I didn't listen to these files. I just selected the top option every time. Got 7 out of 10 on my third attempt. 8/10 on my fifth or sixth.
   
  If you have to cherry pick your results, it generally means you're not doing so well. You need to display a level of consistency to show that you are hearing a difference.


----------



## stv014

Indeed, those not familiar with ABX tests (or statistics) often think that a simple "majority" result is good enough, but with a small sample size that is too easy to achieve by random guessing.


----------



## mikeaj

Yeah, there's some serious data snooping if you're just picking out favorable results.
   
  Getting to 5% on the ABX test means that somebody picking by complete chance (flip a coin, or whatever) would achieve the same score as you just did or better given that many trials, 5% of the time.  It's not so strong a statement.  And if you have to pick out of a subset of results, that's much weaker statistically.


----------



## xnor

Quote: 





spagetka said:


> Third, I cannot identify which track is currently playing, I can say that there is a difference between them.


 
  This doesn't make sense. Either you can identify X as either A or B because you can hear a difference between A and B or you cannot identify X because you cannot hear a difference between A and B.


----------



## scuttle

Quote: 





spagetka said:


> Look, it is only about the concentration. Of course I had unsuccessful attempts before because I am not the robot and which is more important I try to find the weak point in random part of the test track and of course it takes a while to have ears adapted.


 
   
  If you keep on guessing at random until you get a set of answers right by sheer chance, and count only that last set, then it is impossible to fail! What you did was meaningless. If you wanted practice at training you ear before taking the test then you should have used a different track and then ***decided in advance that the first attempt made with the new track would be the one that counts.***


----------



## spagetka

)))))))))))))))


----------



## leogodoy

Keep ABXing like that and you may be THE ONE who will "prove" paying large sums of money for headphone or USB cables do make a difference.


----------



## dynamics

Are there any cons to just leaving the DAC at 24bit 192hz?  Or would 16bit 96hz be ok?  I really can't hear the difference by the way.  My hearing is good too.  But I just couldn't distinguish one from the other.


----------



## jcx

24 96 is often considered the sweet spot - modern flagship DAC do deliver effective 18-20 bits of S/N performance which can be used by modern sw digital volume or other audio processing - but dithered 16 bit will usually be OK with even half right system gain structure
   
  192k for some DACs shows worse S/N and distortion than at 96k - there will always be some speed penalty in digital noise, power requirements that can impact analog output


----------



## harmonix

Quote: 





xnor said:


> Quantization distortion is eliminated using dither. But I agree, high gain after lots of digital attenuation will cause the noise floor to become audible.


 
   
  An ignorant question - dither/white noise is added which is supposedly uncorrelated with the signal.
  Well we all know these cannot be truly random sequences. So how random is this white noise and how do we really know it's uncorrelated? 
   
  Sorry if this has been asked more than once.


----------



## stv014

Quote: 





harmonix said:


> An ignorant question - dither/white noise is added which is supposedly uncorrelated with the signal.
> Well we all know these cannot be truly random sequences. So how random is this white noise and how do we really know it's uncorrelated?


 
   
  It can easily be made random enough that it is not practically different from "ideal" white noise for the purpose of dithering, unless the input signal already contains the same pseudo-random sequence for some reason. That has virtually zero chance of happening accidentally, so it is normally only an issue if the signal was already dithered once with the same noise, or possibly in the case of noise shaping, where the dither is in a feedback loop. For simple dithering in software, the pseudo-random generator can be initialized from the current system time to avoid the problem of dithering more than once with exactly the same noise.
   
  As an example, in the audio processing utilities linked in my signature, I use this simple algorithm to generate noise:

```
x[n] = (x[n - 1] * 742938285) % 2147483647
```
  where x[n] is the current state of the generator (an integer in the range 1 to 2147483646), and % is the modulo (remainder of division) operator. The output of this passes basic tests of randomness like the DIEHARD battery of tests, and is plenty good enough for generating white noise and dithering in particular (but not for more demanding scientific or cryptographic applications). The sequence does loop after 2147483646 samples, but that is not a problem for audio (it is more than 1.5 hours of stereo noise at 192000 Hz sample rate). For the purpose of dithering, the distribution of the noise is made triangular by subtracting the previous sample, which also shifts some of the noise energy into the higher frequency range and slightly reduces the weighted level of the noise at sample rates above 32000 Hz. By default, the initial 'x' in the PRNG is set from the current system time, so the output file is always slightly different. Using some optimization tricks to avoid the expensive % operator, the dithering costs about 12.5 CPU cycles per mono sample of output.


----------



## jaddie

Quote: 





stv014 said:


> It can easily be made random enough that it is not practically different from "ideal" white noise for the purpose of dithering, unless the input signal already contains the same pseudo-random sequence for some reason. That has virtually zero chance of happening accidentally, so it is normally only an issue if the signal was already dithered once with the same noise, or possibly in the case of noise shaping, where the dither is in a feedback loop. For simple dithering in software, the pseudo-random generator can be initialized from the current system time to avoid the problem of dithering more than once with exactly the same noise.


 
  All it would take would be two sequences of dissimilar length, and of sufficient length that any non-random component would be sub-audible. Noise shaping then takes the spectral distribution in a direction that takes the dither signal more out of the audible range.  
   
  This has been well-studied, worth a quick Google search.  Since the concept of dithering in digital audio is more than 30 years old, it might be a valid assumption that any issues dithering would cause would be pretty well resolved by now. 
   
  By the way, basic dither was first accomplished with fully random analog means. Even the noise of an analog mic preamp would do it.


----------



## stv014

I might not have been clear enough, but I did not mean there are actual problems in well implemented dithering or noise shaping, rather that "non-randomness" should only be an issue in some contrived theoretical cases.


----------



## jaddie

Quote: 





stv014 said:


> I might not have been clear enough, but I did not mean there are actual problems in well implemented dithering or noise shaping, rather that "non-randomness" should only be an issue in some contrived theoretical cases.


 
  Gotcha.


----------



## harmonix

Quote: 





stv014 said:


> I might not have been clear enough, but I did not mean there are actual problems in well implemented dithering or noise shaping, rather that "non-randomness" should only be an issue in some contrived theoretical cases.


 
   
  Quote: 





jaddie said:


> Gotcha.


 
   
  Quote: 





stv014 said:


> It can easily be made random enough that it is not practically different from "ideal" white noise for the purpose of dithering, unless the input signal already contains the same pseudo-random sequence for some reason. That has virtually zero chance of happening accidentally, so it is normally only an issue if the signal was already dithered once with the same noise, or possibly in the case of noise shaping, where the dither is in a feedback loop. For simple dithering in software, the pseudo-random generator can be initialized from the current system time to avoid the problem of dithering more than once with exactly the same noise.
> 
> As an example, in the audio processing utilities linked in my signature, I use this simple algorithm to generate noise:
> 
> ...


 
   
  I see. These class of rescursive PRNG's using mod functions don't really cut it for numerical/statistical generators. So the question is are they okay for implementation here? I would guess also the dither algorithm needs to be uncorrelated or "orthogonal" to you signal or it's going to be an issue. Interesting question is also if your initial seed is derived from the system clock then can that be the case. Also for low level signals wouldn't this get "absorbed" into the noise/dither floor created?


----------



## jaddie

Quote: 





harmonix said:


> I see. These class of rescursive PRNG's using mod functions don't really cut it for numerical/statistical generators. So the question is are they okay for implementation here? I would guess also the dither algorithm needs to be uncorrelated or "orthogonal" to you signal or it's going to be an issue. Interesting question is also if your initial seed is derived from the system clock then can that be the case. Also for low level signals wouldn't this get "absorbed" into the noise/dither floor created?


 
  Why not research dither a bit on your own?  It's not like it's a new concept or anything.  Great minds have spend man-years thinking about it.  Pretty sure all the issues have been thought of, and addressed, and papers written, patents filed...


----------



## stv014

Well, the dither noise basically needs to:
  - have the correct probability distribution (easy to achieve even with simple generators)
  - sound like noise to humans (again not difficult, unless the sequence is very short, or there are obvious spectral peaks)
  - be uncorrelated to the input signal (it is very unlikely for recorded music to have content that is "accidentally" similar to the output of a particular PRNG with a specific seed value, and even have the right amplitude etc. to cancel out the dither)


----------



## mikeaj

I think this is one of those situations where even if your model or function produces the "wrong" distribution, it's really not a big deal.  There are a wide range of distributions which would be acceptable.
   
  It's pretty hard to imagine any recorded music being correlated in any significant way with that.


----------



## xnor

http://en.wikipedia.org/wiki/Dither#Different_types
   
  TPDF is ime the most commonly used and easy to implement.


----------



## timbgray

Having some trouble moving from digital photography, where I am totally comfortable with the concepts of dynamic range, bit depth and resolution...  and maybe the terms have a different meaning in audio than digital photography, but to some extent digital should be digital....
   
  Dynamic range is what it is based on the sensor and has nothing to do with bit depth  (dynamic range = the difference in stops between the darkest and lightest source where the sensor can detect a difference)
   
  The bit depth is the precision with which strength of a given "piece of light" can be measured - the light is what it is and bit depth simply is a measure of precision.  In images this is relevant particularly in the editing process where changes to an 8 bit image eg: JPG = roughly analagous to MP3 (actually 3 x 8 bit = 24 one for the red, blue and green channels) where "quantize errors" are more significant than 12 14 or 16 bit images eg: TIFF or RAW - roughly analagous to FLAC etc. 
   
  The resolution is the density in a given area of the photosites of the sensor and would seem to correspond to the samples per second in audio.  the more photosites (pixels) the higher the resolution. 
   
  So the way I see this in audio is take a given sound pressure say 100db - the bit depth would determine the difference between 100.0000000000 (lower bid depth)  and 100.000000000012345 (higher bit depth)  - whether that is an audible difference is probably still open for discussion, but I don't see that bit depth is relevant to dynamic range, it certainly isn't in digital photography.


----------



## xnor

Quote: 





timbgray said:


> Having some trouble moving from digital photography, where I am totally comfortable with the concepts of dynamic range, bit depth and resolution...  and maybe the terms have a different meaning in audio than digital photography, but to some extent digital should be digital....
> 
> Dynamic range is what it is based on the sensor and has nothing to do with bit depth  (dynamic range = the difference in stops between the darkest and lightest source where the sensor can detect a difference)
> 
> ...


 
   
  Some A/D or D/A converters output/accept 24-bit data but don't even reach 16-bit performance, but that's now what we're talking about here.
   
  In digital audio theory *the dynamic range is limited by quantization error*. This is the ideal case (not limited by converter noise).


----------



## jaddie

Quote: 





timbgray said:


> Having some trouble moving from digital photography, where I am totally comfortable with the concepts of dynamic range, bit depth and resolution...  and maybe the terms have a different meaning in audio than digital photography, but to some extent digital should be digital....
> 
> Dynamic range is what it is based on the sensor and has nothing to do with bit depth  (dynamic range = the difference in stops between the darkest and lightest source where the sensor can detect a difference)


 
   Partially true, in that the sensor is the limiting factor, but so is bit depth. The confusing occurs in the scaling a camera does between sensor output and the digital conversion.  So an 8 stop sensor can still be scaled so it is digitized to 12 bits per channel, even though the actual sensor dynamic range is much less than what 12 bits per channel is capable of.  In photography we are also concerned with who big the steps are in the gray scale.  This is one way digital audio a digital imaging differ.
  Quote:


timbgray said:


> The bit depth is the precision with which strength of a given "piece of light" can be measured - the light is what it is and bit depth simply is a measure of precision.  In images this is relevant particularly in the editing process where changes to an 8 bit image eg: JPG = roughly analagous to MP3 (actually 3 x 8 bit = 24 one for the red, blue and green channels) where "quantize errors" are more significant than 12 14 or 16 bit images eg: TIFF or RAW - roughly analagous to FLAC etc.


 
   The precision of measurement idea is right, but the analogies are a bit off.  JPG images are reduced in size by eliminating duplicated pixels during jpg encoding, then predicting them and reinserting them on display.  It's done by considering groups of pixels and the degree to which they differ, keeping the most different ones and dumping the similar ones.  The degree to which that is done is chosen by the jpg quality setting, which is pretty high in cameras, variable in image processing software.  mp3 (technically MPEG-2, Layer 3) processing is a bit different in that it uses the concept of masking to determine what's needed and what's not.  Masking is where a dominant loud frequency makes another close by, but lower level frequency inaudible.  While that's sort of similar to jpg image processing, audio is changing over time, so the data that can be eliminated because it's not audible changes in definition on a continual basis.  Also, when you compare jpg or mp3 compression, the discussion of bit depth is technically a separate issue.  You're right there being larger approximations for lower bit depth, but that's only a related issue to the actual data reduction methods.  TIFF and RAW are "uncompressed", as is FLAC and AIFF, WAV and ALC, but a TIFF image can also have meta data, and a RAW image has tags that are required for proper rendition, and are camera specific as to dynamic range, gamma, color etc.  None of that happens in any of the audio formats.  Part of what goes into a RAW file is determined by the scaling and calibration of the sensor.  In audio, there isn't any of that going on.
  Quote:


timbgray said:


> The resolution is the density in a given area of the photosites of the sensor and would seem to correspond to the samples per second in audio.  the more photosites (pixels) the higher the resolution.
> 
> So the way I see this in audio is take a given sound pressure say 100db - the bit depth would determine the difference between 100.0000000000 (lower bid depth)  and 100.000000000012345 (higher bit depth)  - whether that is an audible difference is probably still open for discussion, but I don't see that bit depth is relevant to dynamic range, it certainly isn't in digital photography.


 
  The resolution analogy is good as far as pixel count vs sample rate.  However, bit depth is always related to dynamic range in both photography and audio.  The fewer the bits the less range between the maximum signal level or light level and the minimum (and noise) level.  In audio, quantization is linear, meaning there's no scaling pre conversion.  So there is a fixed relationship between bit depth and available dynamic range, which is roughly 6dB per bit, not counting noise shaping and dither.  16bit audio is basically capable of 96dB between maximum and noise.  The same is true in photography, except there is scaling dictated by the sensor.  So your blackest black and whitest white of the sensor lands somewhere between the minimum and maximum of the digital word and bit depth, even if the actual sensor output is non-linear.  The key to decoding the scaling data is provided in meta tags, an is important for RAW decoding.  Every get the RAW profile wrong in Photoshop? Probably not, because that's mostly been fixed now, but early on, you could sometimes mis-decode a raw image, the results were interesting, but not useful. But it's that correction that's got you confused. The other issue is color profiles in display and output devices.  So you sensor may capture 10 stops, but your display can't display that, and you certainly can't print that, so what profiles do is again apply a correction to let your (hopefully calibrated) screen "fake" a 10 stop image.  We don't do that in digital audio either. 
   
  Trying to say it simply, bit depth always relates to DR.  In audio, the steps have a fixed size, in imaging, the size of the step is scaled to the sensor/scanner, and then again to the display or output device, such that the sensor's minimum black is still within the bit depth, and the sensor's maximum white is also below maximum defined by bit depth.


----------



## jaddie

Just to add to xnor's post, he's right about quantizing error limiting 24 bit audio. There are only a tiny handful of 24 bit A/D converters that realize full 24 bit performance, and they are expensive even in the pro market. * Here's the one I'm most familiar with, 140 bit DR for real. * Mostly we're at 18-20 bits of real quantization, with a whole lot of noise/dither/q-noise taking up the bottom few bits.  There's no point to 24 bit playback for dynamic range, 16 is more than we can typically use, but there is a point to having 24 bit or more to work with in processing/dsp.  
   
  In imaging, we are mostly display limited.  For 12 bit quantization, we have about at 4000:1 contrast ratio.  Displays that go farther fake it with local dimming.  But in projection, we get a real 2000: 1 contrast ratio in real rooms and theaters because of stray light.  For example, if you projector can theoretically do a 100,000:1 contrast ratio, a guy in the room with a white shirt on will reflect enough light back to the screen to kick that to around 2000:1.  A candle at 10' from the screen is ends up worse than that.  LCD screens in lit rooms have a similar issue.


----------



## jcx

I don't consider these ranged input stage ADC as "true 24 bit" - where superposition linearity, INL, DNL, S/N are limited by 24 bit lsb size - at all signal levels, all of the time
   
  another problem sorting audio ADC claims is noise weighting functions - most audio ADC/DAC marketing bullet point numbers a A weighted - again a fail by the flat, full bandwidth S/N spec expected in instrumentation ADC specs
   
  spurious free dynamic range can also be a useful spec - complex mixed signal systems often have odd, non harmonic spurious frequency lines in their output at very low levels
   
  I think it is currently safe to say there are no Audio ADC meeting the most stringent interpretation of "24 bit" resolution, linearity and unweighted noise floor all at the same time


----------



## jaddie

Quote: 





jcx said:


> I don't consider these ranged input stage ADC as "true 24 bit" - where superposition linearity, INL, DNL, S/N are limited by 24 bit lsb size - at all signal levels, all of the time
> 
> another problem sorting audio ADC claims is noise weighting functions - most audio ADC/DAC marketing bullet point numbers a A weighted - again a fail by the flat, full bandwidth S/N spec expected in instrumentation ADC specs
> 
> ...


 
  Agreed, ranged ADC isn't quite the same as true 24 bit, but it's as close as we can come today, and definitely better than garden-variety 24bit ADCs that only do a real 18.  Audio ADCs aren't instrumentation ADCs, never meant to be.  Weighting in noise specs is probably valid, but I would agree that the specific A curve is not.


----------



## obuckley

I came across this thread by chance and I have read the first page with interest. I notice the thread is 75 pages long now. Unfortunately I don't have a couple of weeks spare to read it all, so apologies if my point has already been made/disputed/disproved.
  Digital audio reproduces sound along two axes. Frequency and Amplitude. The shape of the waveform is a function of the frequencies conventionally along the horizontal (x) axis and the amplitude along the vertical (y) axis.
  One tends to imagine that if you sample a smooth analog curve every so often and draw a bar chart of the results, you get a jagged edge in place of the smooth curve. The more frequently you sample, the smoother and less jagged the digital representation and as the sampling frequency approaches infinity you arrive at a perfectly smooth curve. That is what differential calculus is all about. In theory you do not need to do this. It is not easily understood and is in any case counter-intuitive, that by sampling at a higher frequency than is used in CD, you do not get a closer approximation to the shape of the original analog waveform, (closer to a smooth curve than a bar chart), but Nyquist has proved this and I don't have the maths to argue. According to his theorem, 44.1kHz is enough of a sampling frequency to reproduce perfectly a waveform of up to 20kHz content. You obviously need a greater sampling frequency to reproduce accurately waveforms of higher frequency than human hearing is capable of, but we are here considering human audio.
  However bit depth is a different matter. In practice, the 144 dB that 24-bit allows, does not translate (and is not intended to translate) into nearly 200dB of sound, destroying the ear-drums. You can always turn the volume down, after all.
  A single musical digit somewhere along the x-axis is a number from -32,767 to +32,767. If you try to code a number higher than these into a recording, it gets clipped off. The equipment cannot understand what 32,769 is intended to mean and you usually end up with some very odd and unpleasant artefacts. There is a useful function here. If you take a snatch of music (say a sine wave) ranging from e.g. -12,000, through zero to +12,000; when this has been through a DAC and amplified out into speakers, this will play at a certain sound pressure level. If you change nothing else, but double the digits arithmetically to range from -24,000 to +24,000 - you double the volume. This makes trimming digital music to increase or reduce the volume very easy.
  What this is also saying is that even in CD-quality 16bit sound, you can record 65,000 or so different levels of volume of whatever instrument you are recording. Whether the human ear can distinguish between a sine wave ranging from -12,000 through zero to +12,000 and one doing the same but at 12,001, I do not know. Electronic keyboards used to have something like 127 different volume settings according to how hard you hit the key. That was acknowledged to be inferior to the analog results of striking a piano key, but 65,000?
  So if you get in amongst the digits and start adjusting them, with 24bit sound, you do not multiply the numbers up from 32,768 to some astronomical number - still none of your equipment would understand what any number above 32,767 meant. In practice, you get the option to adjust each digit not a whole digit at a time, but to a decimal place. So you can vary the volume not just from 12,000 to 12,001, but from 12,000.6 to 12,000.7 etc. In this way, you get an increase in dynamic range by adding decimal point precision to your amplitudes. Sounds that were previously recorded at the same amplitude (let's say Volume) which were rounded in CD-quality 16bit sound to the nearest whole digit, may now be represented by different, more precise numbers when recorded in High-Res.
  Whether many human ears can actually detect a difference is a relevant question, but at least (unlike Nyquist and sampling frequency) nobody has yet come up with a mathematical proof that it makes no difference.


----------



## skamp

obuckley said:


> nobody has yet come up with a mathematical proof that it makes no difference.




Does this suit you?

http://www.head-fi.org/t/415361/24bit-vs-16bit-the-myth-exploded/930#post_8946131


----------



## xnor

If you turn the volume down you drown the low-level details in your rooms noise floor, which is exactly why high bit depth doesn't really matter for playback.
   
  Also, the 65536 discrete values are not different volume levels as you mention. Samples values fall somewhere between discrete values most of the time. That just results in quantization error, which is being randomized using dither.
   
  Playing a 16-bit audio file through a 24-bit DAC _does _multiply each sample. -32768 (16 bit) = -8388608 (24 bit) = -1.0 (normalized float) = 0 dBFS.


----------



## obuckley

The post that Skamp refers me to is not a proof. It is a guess. We may consider that it is a reasonable guess, but that's all it is.
   
  I do not disagree with Xnor either about a noise floor affecting one's ability to perceive signal. Noise is unhelpful. However, it is a particularly human gift to be able to screen out parts of the noise spectrum (even when large parts of that would normally be regarded as signal) and focus the ears on e.g. what the 3rd violin to the left in the second row of an orchestra is playing and conclude that he/she is exceptionally gifted/total rubbish. I am not sure that is easily quantifiable.
   
  The discrete values from -32k to +32k are discrete volume levels. If you want to reduce the volume a bit e.g. to remove segments of a recording which have gone into clipping by exceeding 32k, you take the numbers down for a couple of microseconds in the waveform, or you take the whole waveform down a notch, or a larger part of it. One way or another, high numbers, high volume - one to one correlation.
   
  I don't know at a bit level what over-sampling DACs do - I don't care much - what I do know is that if you take a digital recording in 24 bit resolution and look at the numbers, they still range from -32k to +32k but there is a decimal point in there for you to fool with which is not available in 16 bit. I know this, because I do it.
   
  I even kind of get the impression from the little I read of audio product reviews that non-oversampling now tends to get a better press than oversampling. You have already irrevocably lost any extra precision. Oversampling always struck me as unproductive - you can no more conjure up a change of waveform by multiplying the numbers than you can retrieve what is lost in a 128kbps MP3 by re-recording it as a 320kbps. If you start with a recording where that precision is there from the start, it stays there until you eliminate it e.g. by compressing 24bit data down to 16bit. Garbage in, garbage out. 
   
  Please don't think I am claiming to be able to tell the difference. I can barely hear anything above 12,000 Hz these days, let alone 25,000 and it took me quite a while, including a trip to the doctors' to get my ears syringed before I finally realised that my teenaged kids had blown the tweeters on my stereo . To my partial credit I did notice something was wrong. 
  I can't help feeling that CD specs were put together 35 years ago when digital technology was at an early stage. If we can't do better than that with the technology that's available to us now, I'd be a bit surprised - and it is not difficult to do. Most studio masters these days are done in digital form at 24/196 and it's pretty easy to copy digits.


----------



## xnor

Quote: 





> Originally Posted by *obuckley* /img/forum/go_quote.gif
> 
> I do not disagree with Xnor either about a noise floor affecting one's ability to perceive signal. Noise is unhelpful. However, it is a particularly human gift to be able to screen out parts of the noise spectrum (even when large parts of that would normally be regarded as signal) and focus the ears on e.g. what the 3rd violin to the left in the second row of an orchestra is playing and conclude that he/she is exceptionally gifted/total rubbish. I am not sure that is easily quantifiable.


 
  Assume you boost an extremely low-level sound in a dithered 16 bit file so that you can hear the noise floor clearly. (This noise floor usually is recorded noise but let's assume the recording is perfect and the noise floor is just dither.)
  Even if that sound is below the noise floor, you can still hear it. Have you ever had such a freaking extreme clean recording where you had to turn up the volume so much that you could hear the dither noise clearly in order to hear low-level details?

 While I welcome recordings with great dynamic range, it does get annoying at some point. Like in movies where you have to turn up the volume to understand what the characters are saying in dialogs but that volume would blow your ears during action scenes. Also, if we take a look at (dynamically uncompressed) concert hall recordings, we rarely see >70 dB dynamic range due to the noise floor. Sure, maybe there are some details in the noise floor like someone scratching his/her nose, so what?
   
  Quote: 





> The discrete values from -32k to +32k are discrete volume levels. If you want to reduce the volume a bit e.g. to remove segments of a recording which have gone into clipping by exceeding 32k, you take the numbers down for a couple of microseconds in the waveform, or you take the whole waveform down a notch, or a larger part of it. One way or another, high numbers, high volume - one to one correlation.


 
  No, they are discrete values for individual samples. But sounds often consist of several hundreds or thousands of samples, not single samples. So the different volumes of a sound are not fixed to those discrete levels.
  Let's take 8 bit quantization to make things simpler:
  127 = -0.07 dBFS
  126 = -0.14 dBFS
   
  Generate a sine wave at -0.10 dBFS and quantize it to 8 bits using simple triangular dither. Take a look at the spectral analysis and compare it to that of the initial signal. They match, despite the fact that -0.10 dBFS = 126.5 (8 bit) and an 8 bit sample can only have the value 126 or 127, nothing in between (also see below).
   
  Quote: 





> I don't know at a bit level what over-sampling DACs do - I don't care much - what I do know is that if you take a digital recording in 24 bit resolution and look at the numbers, they still range from -32k to +32k but there is a decimal point in there for you to fool with which is not available in 16 bit. I know this, because I do it.


 
  I'm not speaking about the implementation of DACs either. The range of discrete values is a function of bit depth:
  24 bit: -2^23 to 2^23-1 or -8388608 to 8388607
  16 bit: -2^15 to 2^15-1 or -32768 to 32767
  .. and so on ..
   
  Integers do not have a decimal or fractional component. Integers consist of natural numbers (0, 1, 2, 3 ...).
   
  If you "blow up" a 16 bit file to a 24 bit one you will have unused values. For example 32767 -> 8388352 and 32766 -> 8388096 which, surprise, is a difference of 256 (= 2^8).
   
  Quote: 





> I even kind of get the impression from the little I read of audio product reviews that non-oversampling now tends to get a better press than oversampling. You have already irrevocably lost any extra precision. Oversampling always struck me as unproductive - you can no more conjure up a change of waveform by multiplying the numbers than you can retrieve what is lost in a 128kbps MP3 by re-recording it as a 320kbps. If you start with a recording where that precision is there from the start, it stays there until you eliminate it e.g. by compressing 24bit data down to 16bit. Garbage in, garbage out.


 
  It seems you're confusing several different things here. Oversampling, as the name suggest, deals with the sample rate not bit depth. Lossy codecs throw away information based on psychoacoustic models.
  Please be more specific about each of these terms if you want a more detailed response.
   
  Quote: 





> Please don't think I am claiming to be able to tell the difference. I can barely hear anything above 12,000 Hz these days, let alone 25,000 and it took me quite a while, including a trip to the doctors' to get my ears syringed before I finally realised that my teenaged kids had blown the tweeters on my stereo . To my partial credit I did notice something was wrong.
> I can't help feeling that CD specs were put together 35 years ago when digital technology was at an early stage. If we can't do better than that with the technology that's available to us now, I'd be a bit surprised - and it is not difficult to do. Most studio masters these days are done in digital form at 24/196 and it's pretty easy to copy digits.


 
  You have to distinguish all the recording and processing from the final playback/reproduction. Let's assume there is recording both available in 44.1/16 and 44.1/24 but you cannot hear a difference between the two. You wouldn't buy the usually more expensive 44.1/24 file right?
  But those people who earn money by selling you such formats also try to sell you that there are huge differences..


----------



## bigshot

The maximum comfortable dynamic range for music is around 40 dB. Beyond that, you have to listen so loud to hear the quiet parts that the loud parts are uncomfortable.


----------



## obuckley

I know exactly what the last two posters mean about e.g. turning up the volume to hear dialog in movies and then having to turn it down again when the ordnance goes off. Surely that is not a function of excess dynamic range in the recording, it's a function of bad mixing. Given how loud the ordnance is going to be, a good sound engineer should presumably set the amplitude for the spoken parts at a level that does not require volume knob adjustment. They often don't, but that is what is wrong.
   
  I am simplifying slightly when I talk about the amplitude of a single sample. You probably cannot hear a single sample in isolation, although in my experience of e.g. cleaning up clicks and pops on digitised vinyl, you often find that it is a single sample which takes the click into clipping and by attenuating that amplitude (reducing the value from above +/- 32k to below that level), you solve the problem. You may solve it better by adjusting a few adjacent samples as well.
   
  As regards the 8bit sine wave, are you saying Xnor, that the waveform matches exactly because all of the 126.5 values have been randomly allocated to 126 or 127 by the dithering process? Because if so, I don't think I would consider that to be an exact match.
   
  I suppose the decimal points that I was referring to are an implementation in particular software. I understand that a 24bit binary number will be an integer. If you look at a 24bit digital waveform in Adobe Audition or the earlier Syntrillium equivalents, the samples all still range from - to + 32k, but they are all given decimal point precision, which you can adjust minutely as you wish. It had not occurred to me that the actual samples have a higher range adjusted through arithmetic, but I suppose that must be the case.
   
  I do seem to have mixed sampling with bit-depth in what I wrote concerning precision, but I think the point is still that you do not achieve much by over-sampling during playback where you are just sampling the same unchangeable digit on your CD more than once. Isn't that a consequence of Nyquist's theorem? What I am saying in all of these cases (bit depth, sampling frequency, psycho-acoustic legerdemain) is that you cannot create useful extra information from a source that does not contain it, so it is easy enough to translate a 16/44.1 waveform into 24/196, but you have not added anything useful either as regards the bit depth or the sampling frequency, the old slow samples bear exactly the same relationship to each other. Just as it is easy to re-code a 128kbps MP3 file to 320kbps, but that is not the same as taking the original analog or digital source material and coding it to MP3 at 320kbps. Precision, if you apply that term in this context, has already been irrevocably lost, hence the adjective "lossy". 
   
  So all of this extra precision is useful to the recording engineers and for post-production, but once they have settled on the sound they want, they may as well pare it all down to 16/44.1 and throw away any extra precision because nobody can tell any difference. And a format invented before the existence of the personal computer happened to get the specification so adequate in all respects that no improvement is even theoretically possible nearly 40 years later. Impressive.
   
  And the sound engineers responsible for voicing your CD base the final mix on a pair of "average" loudspeakers. So if you happen to own that model of speaker, you hear it as they intended, whereas if you spend $300,000 on your speakers, you get a massive over-emphasis in the bass which most average speakers cannot cope with and who knows what differences in the rest of the spectrum. Maybe Van Gogh had the right idea...


----------



## xnor

On the volume of dialogs: a normal conversation is the range of 40 - 60 dB SPL, a rifle being fired can hit 170 dB SPL, a stun grenade even 10 dB more which is comparable to small explosions I guess. I don't think any sound system can realistically reproduce such events, which is why the dynamic range is decreased a lot (maybe so that peaks only reach about 100 dB SPL) by the engineer.
  Sure you can call it bad mixing, but the issue is dynamic range. You can "fix" such movies by using a dynamic range compressor.
   
  Yes, the level of the 8 bit sine wave in the spectrum analyzer matches that of the source file. If it didn't, you'd see the sine wave at -0.07 or -0.14 dBFS.
   
  My Adobe Audition has a window called amplitude statistics and shows min/max sample values. These values are not limited to +/-32k at all.
   
  Oversampling in the DAC offers huge benefits over old non-oversampling DACs. But you're right, you cannot add information by converting that was not there in the first place.


----------



## jaddie

Quote: 





obuckley said:


> So all of this extra precision is useful to the recording engineers and for post-production, but once they have settled on the sound they want, they may as well pare it all down to 16/44.1 and throw away any extra precision because nobody can tell any difference. And a format invented before the existence of the personal computer happened to get the specification so adequate in all respects that no improvement is even theoretically possible nearly 40 years later. Impressive.


 
  Pretty much true, though most engineers now agree that 44.1 is a little low.  That frequency was chosen because early digital recording systems used video recorders to record the data in a standard NTSC video field, and 44.1 works out as the right sampling frequency to evenly fit bytes on scan lines in video fields of monochrome video decks. Consumer versions of that system sampled at 44.056 because they were stuck with the scan rates of NTSC color.  Early non-video systems sampled at 50KHz, which would have been more desirable then, and now.  60KHz might be all that we aver need.
  Quote: 





obuckley said:


> And the sound engineers responsible for voicing your CD base the final mix on a pair of "average" loudspeakers. So if you happen to own that model of speaker, you hear it as they intended, whereas if you spend $300,000 on your speakers, you get a massive over-emphasis in the bass which most average speakers cannot cope with and who knows what differences in the rest of the spectrum. Maybe Van Gogh had the right idea...


 
  Monitor speakers and control rooms are generally average with respect to each other, but are considerably above average with respect to home systems. Dubbing stages for film work are highly standardized, and also exceed home systems.  A 300K home system will not by definition result in extremely high bass response.  I've calibrated a $350k audio system, it's response wasn't different from a more modest system, and in fact, there were many aspects that were much worse.  Bass response is mostly a room size and shape, and acoustic treatment issue.
   
  Van Gogh had a lot of right ideas, but pinna removal probably wasn't one of them.


----------



## stv014

Quote: 





> Originally Posted by *obuckley* /img/forum/go_quote.gif
> 
> And a format invented before the existence of the personal computer happened to get the specification so adequate in all respects that no improvement is even theoretically possible nearly 40 years later. Impressive.


 
   
  Well, human hearing has not improved over the last 40 years, the limits are still the same.


----------



## tintin220

Quote: 





stv014 said:


> Well, human hearing has not improved over the last 40 years, the limits are still the same.


 
   
  This. It's been said before, but I'll say it again. Really, it has nothing to do with whether there is a measurable, calculable difference. It's just whether that difference is audible, and given the extreme limitations of human biology, it really is not. When we get our bionic ear implants, then we'll reconsider. But as it is, our ears are not evolving any better in this short amount of time, if anything they're worse on average since there's so much more constant, loud noise sources in the environment.


----------



## xnor

Theoretically on a computer you can create audio signals sampled at a couple of MHz and with a bit depth of 64 bits or higher. Purely theoretical you can increase the sampling rate and bit depth to infinity.
   
  But when we have certain requirements like storing frequencies up to 20 kHz (based on human hearing limit) we don't need crazy high sampling rates. Also, all the analog stuff has its limits as well. Many mics for example roll-off a bit above 20 kHz.


----------



## ToddTheMetalGod

I plan on purchasing a SimAudio Moon 100D DAC ( http://www.simaudio.com/moon100D.htm ) for a setup that I'm building soon. It has a BurrBrown PCM1793 DAC, which has an 8x oversampling digital filter. In this particular case, would using 24/96 recordings make a difference (perhaps avoiding the oversampling and allowing better accuracy)? I'm sorry if this isn't an intelligent question, I don't know very much about sound science.


----------



## stv014

Quote: 





toddthemetalgod said:


> In this particular case, would using 24/96 recordings make a difference (perhaps avoiding the oversampling and allowing better accuracy)?


 
   
  96 kHz input would be oversampled, too. The only difference is that the filter cuts off at 48 kHz instead of 22.05 kHz. If you cannot hear frequencies higher than 20-21 kHz (like the majority of people), chances are that it would sound the same. Note, however, that the 24/96 recordings could be mastered better (for reasons related to marketing, rather than technical advantages of the format), and in that case they will sound noticeably better; however, if you downsample them to 44.1 kHz with a good converter, they would sound the same, and better than the badly produced 44.1/16 version.


----------



## sonitus mirus

Quote: 





stv014 said:


> 96 kHz input would be oversampled, too. The only difference is that the filter cuts off at 48 kHz instead of 22.05 kHz. If you cannot hear frequencies higher than 20-21 kHz (like the majority of people), chances are that it would sound the same. Note, however, that the 24/96 recordings could be mastered better (for reasons related to marketing, rather than technical advantages of the format), and in that case they will sound noticeably better; however, if you downsample them to 44.1 kHz with a good converter, they would sound the same, and better than the badly produced 44.1/16 version.


 
   
  I did a bunch of ABX testing with 24/96 flac files converted to 16/44.1 lame mp3, and the files sounded identical to me.  I actually had to ask for assistance to make sure was I doing everything correctly and was provided with step-by-step instructions to make the conversion correctly. (might have even been you)
   
  I was amazed at my own results. Although I have a bit of tinnitus and can't hear anything over ~12-13KHz at normal volume levels after drinking too much caffeine or not getting plenty of rest.


----------



## jaddie

Some of the claims of the proponents of high rate files suggest that the improvements happen at least in part on the recording side.  That would mean that some high rate files we can download would not contain that supposed benefit because they may or may not have originated that way.  Up-sampled files wouldn't count, and things that started with old analog recordings wouldn't either. That takes the valid high rate files down to a much smaller number, and frankly, what you can get from on-line sellers usually comes with a disclaimer that they had nothing to do with how the files were created, and don't have any information about them.  
   
  Then, at least one high-rate proponent (Bill Schnee, Bravura Records) claims he had to have a "special" A/D converter built before he could hear the difference (his ProTools gear simply wasn't good enough), and even then he claims you need 192KHz before the benefits are readily apparent. If you reduce his claims to specifics, it's his "special" converter at 192/24, his "special" console, his "special" mix to 2 tracks, and then it's all wonderful, even if down sampled, though if that's done at least some of the "wonder" goes away.  Of course, he doesn't actually sell anything, at least yet.  When asked what, specifically, causes the improvement, he doesn't claim that it is not just the additional bandwidth, though, it's something else.  But unfortunately, he places the improvement into the "mysteriously unmeasurable" zone.  
   
  That would imply that garden-variety 192/24 ADCs mask the benefits of that bit rate.  So files we can buy online won't contain the full high-rate glory.  And you pretty much need...well...his entire studio and him to get those kind of dramatic results.  That means there's no way for the rest of us to hear this stuff outside of the Christmas music downloads on his site.  He didn't call DACs into question, and I'm sure I have no idea why not, some would. 
   
  I'm not saying I agree with any of this, but I find it interesting that someone of that stature would focus the issue that tightly and try to build a record company on that basis.


----------



## xnor

I guess you listened to that htg podcast. I tried too but closed the window after he said that his special stuff is also full of tubes.. oh yeah and that his studio is the best he's ever seen (or something like that).


----------



## ToddTheMetalGod

Thanks a lot for the help 
	

	
	
		
		

		
			





.


----------



## krtzer

Wow, I did not realize this thread was still going. I totally want to be a part of this. Has anyone brought up ripple distortion on the digital reconstruction of the signal? I couldn't read through all the comments but in what I saw, it didn't seem like there was anything mentioned.


----------



## chewy4

Quote: 





krtzer said:


> Wow, I did not realize this thread was still going. I totally want to be a part of this. Has anyone brought up ripple distortion on the digital reconstruction of the signal? I couldn't read through all the comments but in what I saw, it didn't seem like there was anything mentioned.


 
  Ripple distortion?


----------



## bluepumpkin

Perhaps he is talking about the Gibbs effect (http://en.wikipedia.org/wiki/Gibbs_effect).


----------



## stv014

Quote: 





krtzer said:


> Wow, I did not realize this thread was still going. I totally want to be a part of this. Has anyone brought up ripple distortion on the digital reconstruction of the signal? I couldn't read through all the comments but in what I saw, it didn't seem like there was anything mentioned.


 
   
  It was discussed already, just not called "ripple distortion". Basically, with any reasonable implementation of filtering at 44.1 kHz, the effect is too short and is at too high frequency to be audible. Also, human hearing does not really care about what waveforms "look" like, since it senses bands of frequencies, rather than a simple time domain signal like in a WAV file. There is an example of a real reconstruction filter in a DAC here, which shows that the filtering only significantly affects frequencies above 20 kHz, and that the pre- and post-ringing have a duration of only about +/- 1 ms. Additionally, there were some ABX tests on the forum that could be used to compare (among others) different reconstruction filters, but the positive results are lacking so far.


----------



## xnor

Or perhaps he's talking about ripple in the frequency domain:
   

   
  In which case there's no need to worry either, because the ripple is usually too low to make an audible difference.


----------



## krtzer

Quote: 





xnor said:


> Or perhaps he's talking about ripple in the frequency domain:
> 
> 
> 
> In which case there's no need to worry either, because the ripple is usually too low to make an audible difference.


 
  xnor, yes that was exactly what I was talking about. The ripple in the passband. I think that is one of te main arguments for 96khz recordings; to try to reduce the amount distortion. It's been a couple of years since my DSP class so I'm trying to refresh myself.


----------



## stv014

Frequency response ripple in the passband is only an issue in outdated or very low end DAC chips (such as older onboard codecs, some portable players, etc.). It is easy to reduce to well below the threshold of audibility with a reasonable FIR filter design; very low ripple and very high stopband rejection can be achieved by using the right window function, at the expense of requiring a somewhat longer impulse response. The DAC example linked above does not have any significant ripple either.


----------



## weirdo12

Quote: 





jaddie said:


> Then, at least one high-rate proponent (Bill Schnee, Bravura Records) claims he had to have a "special" A/D converter built before he could hear the difference (his ProTools gear simply wasn't good enough), and even then he claims you need 192KHz before the benefits are readily apparent. If you reduce his claims to specifics, it's his "special" converter at 192/24, his "special" console, his "special" mix to 2 tracks, and then it's all wonderful, even if down sampled, though if that's done at least some of the "wonder" goes away.  Of course, he doesn't actually sell anything, at least yet.  When asked what, specifically, causes the improvement, he doesn't claim that it is not just the additional bandwidth, though, it's something else.  But unfortunately, he places the improvement into the "mysteriously unmeasurable" zone.
> 
> That would imply that garden-variety 192/24 ADCs mask the benefits of that bit rate.  So files we can buy online won't contain the full high-rate glory.  And you pretty much need...well...his entire studio and him to get those kind of dramatic results.  That means there's no way for the rest of us to hear this stuff outside of the Christmas music downloads on his site.  He didn't call DACs into question, and I'm sure I have no idea why not, some would.
> 
> I'm not saying I agree with any of this, but I find it interesting that someone of that stature would focus the issue that tightly and try to build a record company on that basis.


 
   
Barry Diament has made similar observations about recording at 24/192 and his Soundkeeper Recordings is built on delivering his hi-res recordings. He has full songs in different resolutions available for download and comparison:
   
http://www.soundkeeperrecordings.com/format.htm


----------



## julian67

weirdo12 said:


> ....He has full songs in different resolutions available for download and comparison...




At least one of the 16/44 wavs has a bad header or corrupted wav container so won't play in some players. I downloaded the Maria samples from Americas. I remember the same thing a few months ago with the sample. I can't remember if I tried the other offerings. Anyway if your player chokes on the 16/44 wav you can just "convert" it losslessly to a new wav file which will have identical pcm audio but will actually play in any player/device/app that supports wav. ffmpeg will do it with "ffmpeg -i input.wav -c:a copy output.wav". You could also use foobar2k's convert utility.


----------



## Loz2103

Not sure if this has been discussed yet or not. Apologies if it has. So, I listened to my first hi-res 24bit/88.2kHz recording the other day. At the time to be honest it sounded pretty normal to me. Then I realised that I was listening to it on my secondary system, which the DAC only has 3 sample rates, 32 kHz, 44 kHz and 48 kHz. I first thought, oh that must be why I can't hear any noticeable difference in quality. What I really should have been wondering was, why is this working on this DAC? Yes the sample rate was a multiple of 2 of 44.1 kHz and indeed that is what the DAC seemed to lock on to (a red LED told me so). But is this it? Did the DAC just average the two bits instead or one? Any explanation on why this worked would be really appreciated. I can see how it might be possible, but assumed the DAC would be more fussy than that.


----------



## xnor

My guess is you're using Windows with DirectSound. If the sample rates don't match the sound engine will resample it accordingly.


----------



## Loz2103

Sorry, I should have explained my setup. I'm streaming the music over ethernet from my music server to a squeezebox, then from the squeezebox to the DAC. I'm pretty sure that the Squeezebox server is not resampling on the fly, but I suppose it could be.


----------



## peterBj

I just posted this reply in the topic 24/96 files and I thought it might be of interest here as well.
   


> I think you quote me a bit out of context.
> I am talking about the recording process,
> i.e.multi track audio being recorded and mixed.
> That is were the benefit of i.e.24/96 resolution is to be gained.
> ...


----------



## xnor

Sure, 24 bits during multitrack recording is obviously advantageous but recording at higher sample rates does not necessarily mean better sound.
   
  Quote: 





> When recording in 24/96 we are at mix down using less equalization and the reverb tails just sound so much better,than at 16/44 or 24/44.


 
  So you apply less effects to 24/96 files and say it sounds better than heavier processed files that happen to be sampled at 44.1 kHz? Isn't that dishonest to the customers?
  Also, check if your nonlinear plugins/devices are broken (for example if they don't oversample).
   
  That reminds me of the files that Linn offered for format "comparison" quite some time ago. The 24/96 files had a _clearly visibly different waveform_ (my guess was less compression, EQ ...) than the 16/44.1 file. The different processing is what made the 24/96 file sound better, not the delivery format, but of course the 24/96 files are more expensive. Is constraining oneself to do less processing so hard that it justifies a higher price?


----------



## peterBj

no what I mean is we dont need to equalize as much when we are mixing the multitrack down to stereo.
  And we only sell our own recordings.No remasters.We just sell the Wav file as a one to one copy of the multitrack mixdown.
So I dont know what you mean,dishonest?
   
take a look at our site


----------



## xnor

I didn't know you just offer 24/96 files. What I was going at is companies offering 24/96 and 16/44.1 files but processing the 16/44.1 more (more EQ, compression etc.) so they really do sound worse and then advertise that the more expensive and higher bitrate files sound better due to the format and suppress the fact that they processed them differently.
   
  It's like offering two "identical" meals, one made from organic foods (more expensive) and the other one made from conventional foods (cheaper) but to make the one from conventional food taste worse the chef secretly throws in a "bit" of extra spices.
  The analogy is not great because organic foods are _actually_ quite a bit more expensive. Recording at 24/96 just requires more disk space which is really cheap these days.
   
  I still don't understand why you'd need to use less EQ with 96 kHz.


----------



## peterBj

> I still don't understand why you'd need to use less EQ with 96 kHz.


 
   
  hi Xnor
   I don't now why that is is either.It is just a thing we noticed.Maybe some ''technocrat'' here can explain it.
   


>


----------



## Jwooten

Just a comment about being able to actually hear the difference in 16 bit vs 24 bit:
  The main thing I noticed when, years ago, moving from cassette to CD is how clear the audio is on CD versus cassette. This is at least partly due to the dynamic range (noise floor) of CD vs Cassette. CD noise floor in many times better than cassette giving the audio a more up front, seemingly bottomless, clarity. It seems to me that similarly the same would be true for 16 bit vs 24 bit. Moving away from the noise floor gives a perceived increase of audio quality without actually changing anything else. Lower volume passages are much higher above the noise floor on the 24 bit recordings than on 16 bit recordings. I know that in a lot of music such as rock and other intense styles, the audible dynamic range envelope is much less, and in a lot of cases, it's hard to distinguish the music from the noise anyway; but, in certain cases, for instance when recording a soft acoustic guitar passage, a whispering vocal part, or other types of delicate passages, the extra dynamic range can be recognized as the "in your face" sound quality without actually being loud. The further away the material is from the noise floor, the clearer the audio sounds. Again, this isn't noticeable on louder, compressed, recordings, but when recording subtle passages, the extra dynamic range of 24 bit over 16 bit recordings is much appreciated and, whether real or imagined, welcomed.


----------



## xnor

With noise shaping the perceived noise floor of a 16 bit track can reach about -120 dB. The recording needs to have an _annoyingly _huge dynamic range in order for the noise floor to become a problem.
   
  If you think you have recordings where 16 bits are audibly "unclearer" to the 24 bit version please post something like a 30 sec snippet of the 24 bit track.


----------



## xnor

Quote: 





peterbj said:


> I don't now why that is is either.It is just a thing we noticed.Maybe some ''technocrat'' here can explain it.


 
   
  Once you're done mastering the 96 kHz file all you need to do is resample it to 44.1 kHz and convert the bit depth to 16 bit, ideally with shaped dither. Should you have no quality resampler try SoX.
   
  No further processing is necessary.


----------



## bigshot

In order to hear the difference in noise floor between a CD and high bitrate, you would need to turn the volume up so loud, you would incur hearing damage. It's identical at normal non deafening volume levels.


----------



## audiosampling

*16bit vs 8bit, *the myth exploded (again) !
   
http://www.audiocheck.net/blindtests_16vs8bit.php


----------



## GSARider

Thanks for the link - very interesting.


----------



## audiosampling

Quote: 





xnor said:


> With noise shaping the perceived noise floor of a 16 bit track can reach about -120 dB. The recording needs to have an _annoyingly _huge dynamic range in order for the noise floor to become a problem.
> 
> If you think you have recordings where 16 bits are audibly "unclearer" to the 24 bit version please post something like a 30 sec snippet of the 24 bit track.


 
   
  You are totally right. And for the purpose of building a convincing demo I used 8-bit files, but the principles are exactly the same for 16-bit files (only 48 dB quieter). Have a listen here, the audio files are online :
   
http://www.audiocheck.net/audiotests_dithering.php
   
  A 8-bit file is able to capture intelligible speech as low as quiet as -66 dbFS (the dynamic of a 8-bit file is limited to 48 dB in theory). This would translate into -112 dBFS if a 16-bit file was used (-66-48).


----------



## MrTechAgent

Interesting thanks


----------



## obuckley

[size=medium]It has been a while, sorry. The world and this thread have moved on.[/size]
  [size=medium]However, I do have some issues with a couple of replies to my earlier posts:[/size]
  [size=medium]Mastering speakers: We have all (or we should all have, as there are plenty out there) seen adverts for speakers claiming that these are THE speakers used by recording studios in mastering sound for the final cut. Examples include the Yamaha NS4 and NS10, also the Genelec 1031, even Auratones. A bit further up the scale, names like Lipinsky, Focal, Dynaudio. The only mastering studio I know quite well (no names no packdrill) used some rather tacky looking Genelecs coupled with a tiny (5" cube) Carver sub to check their final mixes. I suspect (not least from the price label) that the Lipinskys are quite good, but many of the others are not. Sorry, but they just aren’t. They are selected more than partly because they are popular budget speakers typical of the market that the mix will end up either pleasing or not.[/size]
  [size=medium]And FWIW I was not talking about a $300k set-up, I was talking about $300k speakers (e.g. Wilson Alexandria specials – not that I have a pair). [/size]
  [size=medium]Bass emphasis: So, allow that a pair of such final review speakers does not produce much below 80Hz, quite a lot less at 60Hz and not much at all below 40Hz. For ease of illustration, say flat down to 80Hz, then 80-60 -3dB, 60-40 a further -3dB, 40-20 a further -12dB. Our engineer masters his final mix based on what sounds good from his NS-10s. You then play the resulting CD on your Wilson Alexandrias (allow, flat down to 20Hz). What happens to the bass? Well, of course room acoustics play a part, but other things equal, you will be getting 3dB more than planned from 80-60 Hz, 6dB more from 60-40Hz and a whole dimension you could not tell even existed from 40Hz on down. [/size]
  [size=medium]Even if you have equipment that can be “voiced” to produce a certain frequency response in your particular room, what is the objective? Of course, to minimise peaks and troughs caused by resonances and reinforcements/cancellations in your special environment, but what then? Are you aiming for a flat response 20 – 20k? Because that surely was not what our mixing engineer was hearing coming out of his NS-10s.[/size]


----------



## xnor

Well, there's a difference between studio and *reference *monitors. You use something like the NS-10 to check how well the mix translates to consumer stuff with little bass. But yeah, I've seen people (mostly in home recording) using something like NS-10's only. Then of course you can get into trouble with low frequencies, but respectable engineers know that.
  The master studios I know use something like Lipinski L707's and L150's in surround setups calibrated to be "flat" down to close to 20 Hz, but also hi-fi speakers and grado headphones to check how the mix translates.
   
  But this is kinda off-topic. If you want to continue the discussion in this direction maybe open a new thread? I'm sure some other guys would be interested as well, but here it's just getting buried.


----------



## bigshot

The mixing stages I have worked with all had lead engineers whose responsibility it was to design, maintain and calibrate the room. There were usually consumer speakers in the room to check the mix on home equipment at the end, but the main work was done on the calibrated monitors, which were sometimes built into the cowl overhead. Maybe you didn't see the main monitors because they were built ins.
   
  The point to calibrating the frequency response to flat is it will sound the same if you have to switch to another mixing stage halfway through your mix. No need to start all over again. Every studio will provide the same sound.
   
  Lousy studios don't pay for a lead engineer and don't calibrate their monitors. I only worked in a studio like that once and it was a catastrophe.


----------



## jincuteguy

The reality is, use 24bit whenver u can, therefore 24bit is better. No need to argue here.


----------



## xnor

Quote: 





jincuteguy said:


> The reality is, use 24bit whenver u can, therefore 24bit is better. No need to argue here.


 
  No, 24 bits are utter crap compared to 32 bits. 
	

	
	
		
		

		
			




   
  [  ] You understood what this thread is about.
  [X] You did not.


----------



## ab initio

Quote: 





xnor said:


> No, 24 bits are utter crap compared to 32 bits.
> 
> 
> 
> ...


 
   
  But but but.... the 24/96 is really necessary to open up the sound stage, whereas the 16/44 really sounds veiled. Perhaps you can't hear the difference because your interconnects have too much resistance and capacitance, which are filtering out the details above 18kHz that are necessary to gain the full benefit of hiRez files?
   
  :trollface:


----------



## xnor

Hmm, I was told before that I should shorten my 20 meter interconnects. Maybe there is something to it after all.


----------



## jaddie

I took my 20m interconnects and kept knotting them up until they were short enough, but now they sound...uh...knotty.  I know I was only getting like 23/95 out of them.  
   
  So I un-knotted them and took a big cutter and chopped off the extra wire.  But now they don't work, because I cut off one of the connectors,  so I'm pretty sure I'm down to 0/0.  
   
  But boy!  Talk about deep black!  You guys gotta hear this!


----------



## silverharbinger

Quote: 





jaddie said:


> I took my 20m interconnects and kept knotting them up until they were short enough, but now they sound...uh...knotty.  I know I was only getting like 23/95 out of them.
> 
> So I un-knotted them and took a big cutter and chopped off the extra wire.  But now they don't work, because I cut off one of the connectors,  so I'm pretty sure I'm down to 0/0.
> 
> But boy!  Talk about deep black!  You guys gotta hear this!


 

 I tried this "ultimate black" mod on my interconnects. I still picked up very a slight hiss that I couldn't detect on the 128/8000 version. This might have been due to the tubes I was using though. They pick up even the slightest variance in musicality. I'll try again after 400 hours of burn-in and see of this has improved any.


----------



## jaddie

Quote: 





silverharbinger said:


> I tried this "ultimate black" mod on my interconnects. I still picked up very a slight hiss that I couldn't detect on the 128/8000 version. This might have been due to the tubes I was using though. They pick up even the slightest variance in musicality. I'll try again after 400 hours of burn-in and see of this has improved any.


 
  Ah, well you have one more mod to try then.  It's well known that over time tubes accumulate more hiss.  It's expensive, but if you can find a #70 diamond tipped drill bit and a Mototool, you can drill a tiny hole in the base of the tube and let the excess hiss out.  Just do it over something that you don't mind damaging though.  Free flowing hiss is quite caustic.


----------



## silverharbinger

Quote: 





jaddie said:


> Ah, well you have one more mod to try then.  It's well known that over time tubes accumulate more hiss.  It's expensive, but if you can find a #70 diamond tipped drill bit and a Mototool, you can drill a tiny hole in the base of the tube and let the excess hiss out.  Just do it over something that you don't mind damaging though.  Free flowing hiss is quite caustic.


 
   
  All I have is a #71. 
	

	
	
		
		

		
		
	


	




 Fortunately, fleece audio is selling one for a princely sum, which is cheap at twice the price. I've already donated a kidney and an eye I rarely used, so when the bit arrives I'll try that. Thanks for your help!


----------



## jaddie

OMG, I forgot to say...you need the left-handed bit!  Hope you ordered that one, if not, cancel it right away and get the left-hand bit.  It's like night and day.  And, I fear, a bit more costly due to the rarity of left-hand polarized diamonds.  Most diamonds come from the southern hemisphere, and you need northern hemisphere diamonds for the correct polarization.  Resist the urge to get the lab-polarized southern hemisphere diamonds, or even the man-made ones.  Just not the same.  Your #71 might have worked if you didn't mind doing a smidge of careful reaming...sorry to tell you so late. 
   
  One more thing, most of us that do this keep the work surface in a helium atmosphere because it's inert, and there is a slight danger of explosion if there's enough latent hiss buid-up.  But if you're a "live dangerously" guy, just go for it.  Spontaneous hiss explosions are very rare.  Well, they hardly ever happen. Well, at least none so far this week.  Well, ok, today.  At least the really big ones.  You know, the crater where your city block used to be level.  Well, anyway, they are usually much smaller.  When they happen.  I mean if.  Oh, just get the helium, ok?  I don't want to read about you on the news.


----------



## silverharbinger

Quote: 





jaddie said:


> OMG, I forgot to say...you need the left-handed bit!  Hope you ordered that one, if not, cancel it right away and get the left-hand bit.  It's like night and day.  And, I fear, a bit more costly due to the rarity of left-hand polarized diamonds.  Most diamonds come from the southern hemisphere, and you need northern hemisphere diamonds for the correct polarization.  Resist the urge to get the lab-polarized southern hemisphere diamonds, or even the man-made ones.  Just not the same.  Your #71 might have worked if you didn't mind doing a smidge of careful reaming...sorry to tell you so late.
> 
> One more thing, most of us that do this keep the work surface in a helium atmosphere because it's inert, and there is a slight danger of explosion if there's enough latent hiss buid-up.  But if you're a "live dangerously" guy, just go for it.  Spontaneous hiss explosions are very rare.  Well, they hardly ever happen. Well, at least none so far this week.  Well, ok, today.  At least the really big ones.  You know, the crater where your city block used to be level.  Well, anyway, they are usually much smaller.  When they happen.  I mean if.  Oh, just get the helium, ok?  I don't want to read about you on the news.


 
   
  Will do! You sir are a life-saver!!
   
  I failed to mention that I "donated" my hands in order to purchase the tubes for my rig, but I happen to have the helium for the job and I have strong teeth with adequate bite pressure. I've read that only freshly harvested helium from a class B type 2 star (either core or surface helium per Jason @ fleece) is appropriate for this type of hissterectomy procedure. I will have to make due with a leftover tank of hybrid class B type 4 He that I occasionally use in my casual listening environment. Also, it tends to be a tad more neutral from my experience.
   
  I know I'm fooling myself not to go into this without the right gasses, but after I dropped my life savings to have the samo-flange coupler rebalanced on my mototool xlr+ (rev. 6) I just have to take my chances somewhere.


----------



## jaddie

I believe there would be some significant scientific value in documenting this procedure.  If you would, perhaps, jaw a camera into position before attempting the hissterectomy (wow, autocorrect had a field day with that one!), and perhaps make some aural notes, at least if you haven't harvested your own larynx to fund the project, I feel that the eventual trickle-down to the unwashed masses would have world-changing impact.


----------



## AudioBro

what is the best place to buy high quality audio racks?
   
  is itunes pretty good?


----------



## silverharbinger

Quote: 





audiobro said:


> what is the best place to buy high quality audio racks?
> 
> is itunes pretty good?


 
   
  Personally I get varying results from itunes. Sometimes I have to check other places to find a version that sounds a little better on my system. You can almost never go wrong with searching for other versions to import if it's a song you really like and you want to go to the effort.
   
   
  Quote: 





jaddie said:


> I believe there would be some significant scientific value in documenting this procedure.  If you would, perhaps, jaw a camera into position before attempting the hissterectomy (wow, autocorrect had a field day with that one!), and perhaps make some aural notes, at least if you haven't harvested your own larynx to fund the project, I feel that the eventual trickle-down to the unwashed masses would have world-changing impact.


 

 I totally agree, and time shouldn't be wasted when there is so much enrichment to provide, so on to the results:
   
   

   
   
  Turned out a smidge brighter than I thought initially, but *PERFECT* blacks!
   
   
  Just as an aside to all this, I really enjoy all the premium cables and the like. They're often nice to look at if nothing else, and that's actually useful in this hobby. If you can't make fun of yourself, you probably need to start.


----------



## jaddie

There's apparently a band wagon to be jumped on...(a tiny bit off topic, but what's so unusual about that in this thread?)
   
  Another DSD download producer:
http://www.twice.com/articletype/news/acoustic-sounds-launches-major-label-dsd-music-downloads/108153
   
  A good part of their offerings come from analog masters. 
   
  And so the obvious question would be, "Where does I gets me a DSD-native DAC and software"?  
   
  Oh, you had to ask:
   
https://docs.google.com/spreadsheet/ccc?key=0AgVhKcl_3lHfdFVyenBBNjNpQ2lieG81WGpqQTNfVUE#gid=0
   
  Looks like if you wants to hear your raw, native, un-rendered, unadulterated DSD, and who wouldn't, you pretty much gotta pay up.  
   
  Hope it's worth it, 'cuz we can do 24/96 and 24/192 without much trouble and cost, and the audible improvements seem to be under a little dispute.
	

	
	
		
		

		
		
	


	




  So if we have trouble paying $500 for a good 24/192 DAC, show of hands, who whats to pay $3000 for a DSD- native DAC?  C'mon, there must be somebody...ok, there's one. 
   
  So, if I get this right, we have analog masters with built in THD, IMD, guidance errors (recorded in), flutter (recorded in), scrape flutter (recorded in), noise, modulation noise, all that good stuff, and we're capturing it in all its glory with DSD, selling it at $25/pop, and selling $3000 DACs to reproduce all of that.  By the way, tape guidance errors result in interchannel phase misalignment that is the equivalent of someone moving your speakers toward and away from you, each in opposite directions, about at least half inch at a time, perhaps more.  So much for "time alignment" and "phase alignment".  That, as well as flutter, is correctable in the digital domain, so long as you stay in PCM.  Sorry DSD, you're stuck with that faithful reproduction of the master.  
   
  Yeah, I'm cranky this morning.


----------



## Baxide

I put async USB and DSD in the same league. Someone invented the illness, told you the symptoms to expect and then sold you the medication. In the old days such people were called quacks.


----------



## Digitalchkn

Personally, I am waiting for someone to invent a variable step ADC/DAC so that we can have PCM in floating point. Then I would bet good money if anyone can distinguish differences between bit depths


----------



## jcx

SACD does have a major advantage - for Sony  - multiple layers of copy protection


----------



## bigshot

And a way to sell you Dark Side of the Moon for the fifth time.


----------



## Thalik

Just read the whole thread.. very educational! Not only 24/16 bit but a lot of other interesting topics.
  Thanks a lot Gregorio, it was great post, made me know a bit more about digital audio and saved some money for the family budget  Also reminded university years and Kotelnikov theorem. Kotelnikov? Yes, that was actually the guy who first formulated and proven discussed here theorem in 1932. Shannon did it in 1949.. at least this is what Russian science and Russian wiki say 
   
  Talking about audiophile music. Why labels would not release 2 versions of music  - one for general market and one for audiophiles? Surely they would be able to charge premium for audiophile version and surely it would make them more money as also some of general public would get curious / aspirational and pay more just to own prestigious audiophile version or maybe even both.
  Or is the audiophile market considered too small and making another master is just too much hassle for the Labels?


----------



## silverharbinger

Quote: 





thalik said:


> Talking about audiophile music. Why labels would not release 2 versions of music  - one for general market and one for audiophiles? Surely they would be able to charge premium for audiophile version and surely it would make them more money as also some of general public would get curious / aspirational and pay more just to own prestigious audiophile version or maybe even both.
> Or is the audiophile market considered too small and making another master is just too much hassle for the Labels?


 
   
  Some of them do.
   
http://www.waste.uk.com/Store/waste-radiohead-did-35-10198-tkol+rmx1+wavmp3+digital.html
   
  I can post up many other examples of this, but I also notice that the major labels aren't that interested. It probably would lead to more issues than it's worth to offer up formats that aren't natively compatible in iphones and other devices. Let's face it, audiophiles are probably mostly nerds, me included. We know what to do with these files, but we're the minority.


----------



## stv014

A better mastered version would not necessarily need a different format. Although it is obviously more marketable in 96/24, regardless of whether the format upgrade itself is actually useful.


----------



## silverharbinger

Quote: 





stv014 said:


> A better mastered version would not necessarily need a different format. Although it is obviously more marketable in 96/24, regardless of whether the format upgrade itself is actually useful.


 
  Maybe it is or it isn't, but the perception is that mp3s are lossy, and wav/flac files are lossless. That ties into the placebo effect and consumer confidence. If you're going to market to audiophiles just sell the version they're looking for. We will convert it if we want to.


----------



## Thalik

I found this article on other forums here but havent seen it discussed in this relevant thread.
http://people.xiph.org/~xiphmont/demo/neil-young.html
   
  Interesting read, also says that "high definition" format might in fact damage sound of certain systems by introducing distortions in audible range..


----------



## UltMusicSnob

Quote: 





thalik said:


> I found this article on other forums here but havent seen it discussed in this relevant thread.
> http://people.xiph.org/~xiphmont/demo/neil-young.html
> 
> Interesting read, also says that "high definition" format might in fact damage sound of certain systems by introducing distortions in audible range..


 

 I have to notice these portions:
   
  "_Double-blind_ listening tests are the gold standard; in these tests neither the test administrator nor the testee have any knowledge of the test contents or ongoing results. Computer-run ABX tests are the most famous example, and there are freely available tools for performing ABX tests on your own computer[19]. ABX is considered a minimum bar for a listening test to be meaningful; reputable audio forums such as Hydrogen Audio often do not even allow discussion of listening results unless they meet this minimum objectivity requirement [20].
    

  I personally don't do any quality comparison tests during development, no matter how casual, without an ABX tool. Science is science, no slacking."
   
  "First, confirmation bias does not replace all correct results with incorrect results. It skews the results in some uncontrolled direction by an unknown amount. How can you tell right or wrong _for sure_ if the test is rigged by your own subconscious? Let's say you expected to hear a large difference but were shocked to hear a small difference. What if there was actually no difference at all? Or, maybe there _was_ a difference and, being aware of a potential bias, your well meaning skepticism overcompensated? Or maybe you were completely right? Objective testing, such as ABX, eliminates all this uncertainty.
  Second, "So you think you're not biased? Great! Prove it!" "
   
  And once you prove it? Then what?


----------



## xnor

Quote: 





ultmusicsnob said:


> And once you prove it? Then what?


 
  You post your bias-free sighted test results and also your double blind test results (which you did after the sighted one). Then others will try to reproduce your results.
   
  If everything works out then we know ... for that one test.


----------



## UltMusicSnob

Quote: 





xnor said:


> You post your bias-free sighted test results and also your double blind test results (which you did after the sighted one). Then others will try to reproduce your results.
> 
> If everything works out then we know ... for that one test.


 

 Let me just catch up on the terminology: I'm familiar with "single-blind" and "double-blind" (I regularly conduct reviews of others' research under double-blind conditions), but haven't come across "sighted test" before, and it doesn't seem to appear in the article. I infer it means "neither single nor double blind", but that's just my guess by implication.
   
  This forum http://hddaudio.net/viewtopic.php?id=3341 purports to provide a definition. However, under the definition of "sighted" there, the terms "bias-free" and "sighted test" would have to be considered mutually exclusive--which I took to be one point among many of the article at http://people.xiph.org/~xiphmont/demo/neil-young.html. Additions/corrections welcome.
   
  I was also thinking about this quote:
     "Second, "So you think you're not biased? Great! Prove it!" The value of an objective test lies not only in its ability to inform one's own
      understanding, but also to convince others. Claims require proof. Extraordinary claims require extraordinary proof."
   
  I'm not sure where "extraordinary proof" applies here; I would not consider double-blind--as for example an ABX of the type described in the article--an extraordinary proof, but rather more like "minimum acceptable standard when experimental conditions allow double-blind testing" (as they do in the cause of audio formats testing). The quote in this form originated with a debunker of paranormal phenomena (Marcello Truzzi), which is sort of outside the normal bounds of audio testing.
   
  Anyway, this just goes to the level of commitment (high) the article's author shows toward the results of double-blind testing (ABX) for audio comparisons.


----------



## xnor

Yeah they are pretty much mutually exclusive - so the IF in my second sentence is a big one. I wouldn't expect anyone to get similar scores in both tests, except if he rigged the results/test setup. 
	

	
	
		
		

		
		
	


	



   
  Yeah, the extraordinary comes from the paranormal, but I think what he's talking about are claims like blatantly audible differences when all that's changed is somewhere above the human hearing range, or day/night differences between expensive cables etc.
  Such claims are clearly a step above claims like: "I can ABX a +x dB boost with an EQ at 500 Hz" or being able to distinguish mp3 from lossless etc.
   
  Yeah, (double) blind tests should be the standard. As J Gordon Holt said: "high-end audio lost its credibility during the 1980s, when it flatly refused to submit to the kind of basic honesty controls (double-blind testing, for example) that had legitimized every other serious scientific endeavor since Pascal."


----------



## hogger129

I have a 24/176.4 copy of Let It Bleed, as well as the same album on a regular CD.  Listening to them on my BD player (24/176.4 are WAV files), the only difference I can detect is that the 24-bit version runs a little more smoothly.  It doesn't sound as "digital."  Probably due to the higher sample rate.  But it isn't a night and day difference.  I don't think it's _that _much better sounding than a regular CD.
   
  I know some people prefer their 24-bit stuff to a CD, but for me personally it isn't worth the extra cost or the extra space it consumes.


----------



## UltMusicSnob

There may be a very fundamental conceptual problem to address, which I'd say is the notion of transparency.
    The original source is the original live acoustic signal (and even that disappears when we go to electronically generated program material which never passed through the air to begin with).  After that, everything "distorts" it:

 the air in between (atmosphere attenuates)
 the mic capsule, mic circuitry, mic settings (multi-pattern, pad, hi-pass filter to take out cable/stand bumps, etc)
 the pre-amp, the mixing board (Neve now famous for their great "sound"), sucessive sub-busses and main bus, sends, mastering eng. and equip.
 distribution formats analog and digital--all with their own advantages and disadvantages
 all the playback stages discussed widely on this forum
   
  In other words, there is ultimately no authoritative form of the signal. We don't even hear the same way at the every end of the playback change--everyone's room is different, everyone's physiological hearing equipment is different, everyone's music-decoding brain is different.
   
  The closest we get to a reference standard form of the signal is what the mastering engineer delivers, and even then: how many of those who will purchase the product will have the same room/playback chain as the mastering engineer?--I'm guessing zero. The M.E. delivers what he hopes is a robust form of the signal, providing good music under many varying conditions (and thus containing his own judgments about tradeoffs and compromises).
   
  Yeah, the signal is distorted, no matter what form it's in. The **only** way to get the real thing is to show up and hear Jon Vickers do it live. No comparison, then. (I was fortunate to hear him from 6 feet away, onstage, as a member of the opera chorus). Authority stops there. After that, it's just **which** form of 'distortion' one prefers (AND, can prove that one hears in double-blind, i.e. ABX).


----------



## UltMusicSnob

Quote: 





hogger129 said:


> I have a 24/176.4 copy of Let It Bleed, as well as the same album on a regular CD.  Listening to them on my BD player (24/176.4 are WAV files), the only difference I can detect is that the 24-bit version runs a little more smoothly.  It doesn't sound as "digital."  Probably due to the higher sample rate.  But it isn't a night and day difference.  I don't think it's _that _much better sounding than a regular CD.
> 
> I know some people prefer their 24-bit stuff to a CD, but for me personally it isn't worth the extra cost or the extra space it consumes.


 

 "Let It Bleed" was released 1969, so it was mastered analog, and since then sampled/converted in many different forms. Rather than purchase a "high definition" copy, I think you should be able to get close to the same result upsampling it yourself for free.
   
  There **might** be some advantage to hearing something originally recorded at 24/176.4, but I'm thinking it would be a very small difference indeed in anything except a live binaural recording of classical music in a near-acoustically-perfect venue.


----------



## hogger129

Quote: 





ultmusicsnob said:


> "Let It Bleed" was released 1969, so it was mastered analog, and since then sampled/converted in many different forms. Rather than purchase a "high definition" copy, I think you should be able to get close to the same result upsampling it yourself for free.
> 
> There **might** be some advantage to hearing something originally recorded at 24/176.4, but I'm thinking it would be a very small difference indeed in anything except a live binaural recording of classical music in a near-acoustically-perfect venue.


 
   
   
  From what I understand, the copy I have which is from HDTracks is the same master as the 2002 SACD.  If I upsampled a CD, wouldn't I just be basically "padding" 16/44 and not really getting any better sound quality out of it?


----------



## UltMusicSnob

Quote: 





hogger129 said:


> From what I understand, the copy I have which is from HDTracks is the same master as the 2002 SACD.  If I upsampled a CD, wouldn't I just be basically "padding" 16/44 and not really getting any better sound quality out of it?


 

 What was the master used for the SACD, the original analog mixdown? Or was it re-mastered? In the latter case, if that's the only way you can get it, it might be worth having, for huge fans of the Rolling Stones. My comment about doing it yourself would only apply if the Redbook CD I could get in the store now would have the same master as the SACD you're talking about.
    I have upsampled some of my 16/44's, and it sounds different and better to me (VERY subtly), which I attribute to the fact that my DAC necessarily plays back the different sample rates...well, **differently**. Filters and all that. I also find the high resolution rates smoother, but that's a very general impression.
    If you don't want to use the Hard Drive space, then for those purposes the discussion is already over (not worth it)--I don't mind using the space, myself.
    One thing you could do is get hold of someone's post-2002 "Let It Bleed" CD, upsample it yourself with software (I would never buy an upsampling hardware device just to upsample), and then do an ABX test with your purchased high-res version, see if it makes any difference or not.


----------



## stv014

Quote: 





> Originally Posted by *UltMusicSnob* /img/forum/go_quote.gif
> 
> I have upsampled some of my 16/44's, and it sounds different and better to me (VERY subtly), which I attribute to the fact that my DAC necessarily plays back the different sample rates...well, **differently**. Filters and all that. I also find the high resolution rates smoother, but that's a very general impression.
> If you don't want to use the Hard Drive space, then for those purposes the discussion is already over (not worth it)--I don't mind using the space, myself.


 
   
  Putting discussion regarding the usefulness of upsampling aside, it is not necessary to use any additional hard disk space, since the upsampling can also be performed in real time (for example, using the SoX resampler plugin in foobar2000, which - with the right settings - can produce the same results as the iZotope converter you used).


----------



## bigshot

The Stones have been remastered a bunch of different times. Even on LP, the singles were mastered differently than the same song on the LP. The SACD even has a different master on the redbook layer as the SACD layer. You can't compare formats using the Stones as an example, because it's never the same.


----------



## Digitalchkn

There is one thing that doesn't make much sense to me and that's the loose use of the term "remastered". As you probably know mastering is production step after the sources are mixed. In a sense, a master is already a second generation copy of the original recording. I am guessing the exception to that would be done in 50's and 60's.  In either case, if you take the master and copy it over to another format (be it in 16/44.1, 24/96, DSD, whatever) and create another master, in principle the best you can get is what is on the original master. At that point one could tinker with the sound to make improvements, but it is still a copy of the original (at which point I may start to cry foul because I want the best original recording possible).  If the recorded sources are available, then you can truly create a new master but at that point there is inevitable remixing because there is no way to recreate precisely the same mix as was done before.  My point is, the term "remaster" seems to be thrown around too loosely by marketing - a carry over from early days of CDs I guess (when simply using the word "digital" was all the rage). Is it just me who's unclear on this? Shouldn't we as a consumer how what is being done?


----------



## bigshot

There was a link earlier in the thread that outlines what mastering is.


----------



## Digitalchkn

Quote: 





bigshot said:


> There was a link earlier in the thread that outlines what mastering is.





>


 
  I must have missed that one. So is there an standard definition then?


----------



## bigshot

I think this was the link... http://en.wikipedia.org/wiki/Audio_mastering


----------



## Digitalchkn

Quote: 





bigshot said:


> I think this was the link... http://en.wikipedia.org/wiki/Audio_mastering


 
  Thanks, but the point is trying to make is that the sources used to master varies. Sometimes only an original master is available. Sometime source multitrack is available. Some mastering engineers like to rerun the dub at different rates on best equipment for each rate, others simply downconvert in software. Many labels just don't specify how this is done, I think.


----------



## bigshot

It's been my experience back when I was still working with analogue material recorded on 24 track, the 24 track mix would be bounced down to two tracks of a 4 track ADAT digital master. That became the master element and protection copies would be made. I would guess that if the 24 track mix can't be gone back into or is missing, they would use the first generation analogue bouncedown for that... a two track tape sub master, then to ADAT. Today, there isn't much of a problem with generation loss, a digital submaster is going to be a bit for bit copy of the master. Not like in the old days where submasters were sometimes several generations from the source.


----------



## Digitalchkn

bigshot said:


> It's been my experience back when I was still working with analogue material recorded on 24 track, the 24 track mix would be bounced down to two tracks of a 4 track ADAT digital master. That became the master element and protection copies would be made. I would guess that if the 24 track mix can't be gone back into or is missing, they would use the first generation analogue bouncedown for that... a two track tape sub master, then to ADAT. Today, there isn't much of a problem with generation loss, a digital submaster is going to be a bit for bit copy of the master. Not like in the old days where submasters were sometimes several generations from the source.
> 
> 
> I am assuming we are talking about analog sources here. Generally, why would you remaster a digital master?
> ...


----------



## bigshot

Did you take a look at the wikipedia article? There's a whole raft of processing that they can do, from compression to noise reduction to sweetening with reverbs or EQ.


----------



## UltMusicSnob

digitalchkn said:


>


 
 A remastered version could easily be better, worse, or as-good-as-but-in-a-different-way, as the earlier master. Mastering is not just a technical exercise; it involves aesthetic choices, trade-offs and compromises, and a multitude of solutions to anticipated uses of the material.


----------



## xnor

Sadly, remastering today often just means a v-shaped EQ curve and compression, compression, compression.


----------



## UltMusicSnob

Mastering for delivery on vinyl meant taking into account all sorts of physical conditions of the record material that just don't apply any more. The spacing of the grooves is not constant, for example--this is why you can see the bands of material when you hold up a record  and let light bounce off its surface--some are dense, the quiet passages; some are further apart, the louder passages. A treatment called the RIAA equalization curve was used (in part) to limit the required size of grooves, allowing more playing time per side on LP's.
  
 A CD does not care how loud the material is. It holds a Redbook CD's worth of data, period. CD's also have some track flexibility here, but the main point is that the amplitude of the signal does not affect playing time on a CD.
  
 If I were a copyright holder/label owner requesting a remaster of a classic rock album, the first thing I'd do is ask the mastering engineer to go back and examine all the LP-ready compromises that went into the first master. Probably I'd want it mixed down again, not just remastered, so that I could fully 'un-compromise' the result. Maybe even bring back the band, if they're around, show them the capabilities that couldn't get onto the original record because of compromises made for LP vinyl distribution.


----------



## bigshot

They usually go back to the original master, not the LP era submaster. No compromises in the original session tapes.


----------



## Digitalchkn

bigshot said:


> They usually go back to the original master, not the LP era submaster. No compromises in the original session tapes.


 
 Do they use compression on LP submasters? I imagine they have to since the dynamic range of vinyl has got to be fairly limited and they the dynamics are very smooth and soft. Why can't this soft compression be applied to digital masters? Why the drive to sound until you are practically hearing square waves.


----------



## jaddie

ultmusicsnob said:


> Mastering for delivery on vinyl meant taking into account all sorts of physical conditions of the record material that just don't apply any more. The spacing of the grooves is not constant, for example--this is why you can see the bands of material when you hold up a record  and let light bounce off its surface--some are dense, the quiet passages; some are further apart, the louder passages. A treatment called the RIAA equalization curve was used (in part) to limit the required size of grooves, allowing more playing time per side on LP's.


 
  What's being referred to here is called "pitch control", and was automated by using a preview head exactly one disc revolution in advance of the actual cutting signal.  The system monitored level and adjusted the lathe pitch to allow for higher levels of low frequency modulation.  Pitch control did directly affect how the disc was mastered audibly, however, it just prevented groves from colliding with each other.  The RIAA curve, on the other hand, needs to be accurate both in recording and with the exact inverse applied during play.  Getting those two curves to be exactly reciprocal through the entire system including the cartage is often assumed, but usually isn't right.
 Quote:


ultmusicsnob said:


> A CD does not care how loud the material is. It holds a Redbook CD's worth of data, period. CD's also have some track flexibility here, but the main point is that the amplitude of the signal does not affect playing time on a CD.


 
 What's important to note here is what about the LP's capabilities would affect how it was audibly mastered.  The LP has a maximum record level that varies with frequency...a lot, and also varies with pan position.  One of the reasons kick and bass are usually mixed center is that a center signal results in lateral groove modulation, which has a higher maximum than a L or right dominant signal, which would result in a vertical groove modulation component.  The vertical limit is cutting a groove so shallow it won't hold a stylus, or cutting it so deep the cutter stylus slams into the aluminum substrate.  High frequency limits are there too caused by the maximum cutter velocity.  Move the cutter stylus to fast and the back facet digs into the groove wall just cut be the front facet. The HF limit is level and frequency dependent, i.e., the maximum level that can be cut is lower at higher frequencies.  This caused many mastering engineers to employ a high frequency limiter, which definitely affects the sound, or simply reduce high frequencies with an equalizer.  The myth conflict here is that analog vinyl has such a wide bandwidth it goes into the ultrasonic range.  Simply not true. 
  
 But note that these limitations of the LP medium are all related to high levels.  If you don't master for maximum loudness, these factors are less important, and it is in fact possible to master without anything other than proper RIAA eq and pitch control, without regard for the mix.  The LP is essentially a flat medium below its maximum limits.  There have been numerous direct-to-disc recordings made directly to the lathe without any of the typical mastering processing other than some basic mix decisions and manual pitch control.  They sound gorgeous. 
  
 The CD is a flat medium that will record and reproduce the same maximum level regardless of frequency or pan position (the exception being those using pre-emphasis).


----------



## gjfs

gregorio said:


> So, can you actually hear any benefits of the larger (48dB) dynamic range offered by 24bit? Unfortunately, no you can't. The entire dynamic range of some types of music is sometimes less than 12dB. The recordings with the largest dynamic range tend to be symphony orchestra recordings but even these virtually never have a dynamic range greater than about 60dB. All of these are well inside the 96dB range of the humble CD.


 
  
 Most of your post is right, but this bit is wrong.  You're confusing musical dynamic range with signal processing dynamic range.  No one listens to music with a white noise floor of -12 dBFS, that would sound awful.  By this definition of dynamic range, "music" consisting of a single sine wave tone would have 0 dB dynamic range, right, because the level never changes?  It could therefore be reconstructed by any converter, even 1 bit, right?  No, obviously not. The recording still has a large dynamic range, due to the noise floor at _other frequencies_ than the sine tone. A sine wave with lots of hiss in the background vs a sine wave with little hiss in the background.
  
 But the overall conclusion still applies: recordings made with real analog equipment can't have a dynamic range above 120 dB, and therefore noise-shaped 16-bit is sufficient to reproduce them.
   
 Quote:


gregorio said:


> I know that some people are going to say this is all rubbish, and that “I can easily hear the difference between a 16bit commercial recording and a 24bit Hi-Rez version”. Unfortunately, you can't, it's not that you don't have the equipment or the ears, it is not humanly possible in theory or in practice under any conditions!!


 
  
 Often they hear a difference because the recordings were mastered differently.    They're not actually comparing the same recording.
  


gregorio said:


> 2 = The concept of the perfect measurement or of recreating a waveform perfectly may seem like marketing hype. However, in this case it is not. It is in fact the fundamental tenet of the Nyquist-Shannon Sampling Theorem on which the very existence and invention of digital audio is based. From WIKI: “In essence the theorem shows that an analog signal that has been sampled can be *perfectly* reconstructed from the samples”. I know there will be some who will disagree with this idea, unfortunately, disagreement is NOT an option. This theorem hasn't been invented to explain how digital audio works, it's the other way around. Digital Audio was invented from the theorem, if you don't believe the theorem then you can't believe in digital audio either!!


 
  
 Largely true, but don't get carried away.  Nyquist-Shannon is mathematical theory, based on impossible devices like ideal brickwall lowpass filters (real-life filters cause aliasing, so the reconstruction can never be perfect), and this discussion of quantization actually isn't part of the theory.  It deals with discrete-time analog, not digital.  Non-linear quantization error, etc is not included in Nyquist-Shannon.


----------



## marone

Not much point in arguing with an objectivist whose belief system is that all that is measurable can be heard and all that you hear can be measured with current technology. Same types made the same arguments 10, 20, 50 years ago.

Heck, they said it about 16 bit in 1978 and how I cannot hear the difference between the $20 gear on my desk with different chipsets and how I cannot hear DSD as better than whatever.

I see no point in engaging people like gregorio. He has his bias, which he puts forth as not being biased.

On to better topics:
Vinyl rips converted to FLAC definitely sound better, more natural mid-range, than the ripped CD of same. I actually prefer ripped vinyl over the CD. Only versions that sound as good are very high bit master to FLAC or DSD not PCM for CD but the higher desk bitrate mixdown before any other consolidation is made.

_Then_ you get to a point where the vinyl is bettered.


----------



## gjfs

marone said:


> Heck, they said it about 16 bit in 1978 and how I cannot hear the difference between the $20 gear on my desk with different chipsets and how I cannot hear DSD as better than whatever.


 
  
 Every difference you hear is either imagined or measurable.  The way to distinguish between imagined and measurable differences is by doing ABX testing.  There is nothing else.


----------



## stv014

Have you ever tried to convert a "high resolution" file (like a vinyl rip to 96/24 FLAC) to 44.1/16 format, then convert the result back to the original format (96/24 or whatever else), and compare it to the original with an ABX comparator ?
  
 Your vinyl rips will of course sound obviously different than the CD version which is from a different master (if it is newer, it is very likely more compressed/clipped).


----------



## xnor

marone said:


> Not much point in arguing with an objectivist whose belief system is that all that is measurable can be heard and all that you hear can be measured with current technology. Same types made the same arguments 10, 20, 50 years ago.
> 
> Heck, they said it about 16 bit in 1978 and how I cannot hear the difference between the $20 gear on my desk with different chipsets and how I cannot hear DSD as better than whatever.
> I see no point in engaging people like gregorio. He has his bias, which he puts forth as not being biased.


 
  
 Not much point in arguing with somebody who doesn't bring forward factual arguments but discards facts because he doesn't like something about the person making them.
  
 Such ad hominem are boring and just show that *you have no real arguments*. Let me guess: you invested a considerable sum in high-res stuff so what you always believed to be true _has _to be true, right?
  
  


> Vinyl rips converted to FLAC definitely sound better, more natural mid-range, than the ripped CD of same. I actually prefer ripped vinyl over the CD. Only versions that sound as good are very high bit master to FLAC or DSD not PCM for CD but the higher desk bitrate mixdown before any other consolidation is made.
> 
> _Then_ you get to a point where the vinyl is bettered.


 
 What a complete non sequitur. If such logical fallacies are what you base your opinion on then better not spread it.
  
  
 (I'm sorry if this sounds harsh, but I got inspired by your post.)


----------



## UltMusicSnob

Do people have far better vinyl than I've gotten in my purchases? I *still* play all my vinyl records, and I enjoy them, but I would never say they were better than digital, just on the surface noise and pops and clicks alone. The best pressings I own are by Deutsche Gramophon, and they're excellent. For years I religiously cleaned vinyl surfaces before playing. With the best that I could do, with the equipment I could afford (mid-fi Technics direct-drive and Audio-Technica cartridge with an *expensive* hyperelliptical stylus, I still got nowhere near the noise floor of digital.
   Perhaps there's some level of pressing quality and playback quality that makes a vinyl rip worth it, but I never once heard that level in years of meticulous LP purchasing and playing.


----------



## nick_charles

marone said:


> Not much point in arguing with an objectivist whose belief system is that all that is measurable can be heard





> *That is not the objectivist position which would posit that you can measure stuff that cannot be heard*





> and all that you hear can be measured with current technology. Same types made the same arguments 10, 20, 50 years ago.





> *[size=x-small]Our knowledge of psychophysics is incrementally improving but some objective thresholds of hearing are pretty well [/size]defined[size=x-small] such as the Fletcher-Munson curves and the thresholds for detecting particular types of artifact/distortion - if you choose not to believe in them that is your choice they are entirely disinterested[/size]*





> Heck, they said it about 16 bit in 1978 and how I cannot hear the difference between the $20 gear on my desk with different chipsets and how I cannot hear DSD as better than whatever.





> *Can you point to peer-reviewed papers that substantively contradict that opinion, I can point to papers from bodies like JVC back in the late 70s that were able to define the requirements for audio transparency. Regarding DSD vs PCM you need to read Blech and Yang who empirically tested subjects ability to tell them apart, using trained expert listeners including Tonnmeiser students they found that over 97% of subjects could not tell them apart*





> I see no point in engaging people like gregorio. He has his bias, which he puts forth as not being biased.





> *Everyone has bias including you  and me, however if you state facts that are facts and not just opinions then these are facts even if you are particularly attached to them, opinions without empirical backing are just opinions *


----------



## nick_charles

ultmusicsnob said:


> Do people have far better vinyl than I've gotten in my purchases? I *still* play all my vinyl records, and I enjoy them, but I would never say they were better than digital, just on the surface noise and pops and clicks alone. The best pressings I own are by Deutsche Gramophon, and they're excellent. For years I religiously cleaned vinyl surfaces before playing. With the best that I could do, with the equipment I could afford (mid-fi Technics direct-drive and Audio-Technica cartridge with an *expensive* hyperelliptical stylus, I still got nowhere near the noise floor of digital.
> Perhaps there's some level of pressing quality and playback quality that makes a vinyl rip worth it, but I never once heard that level in years of meticulous LP purchasing and playing.


 
  
  
 Vinyl is just inherently noisy, it was especially annoying to me back in the 80s trying to listen to quiet passages in classical music, all the imperfections were easily apparent, with headphones it was positively painful. But some recordings are not out on CD and once the music gets to a certain level the noise is drowned out. I still remember my first encounter with CD in Charing Cross Records in 1984. I was in the demo studio and heard the opening bars of Mahler's 1st (Solti/CSO) played back on a Marantz CD63 and there was no noise whatsoever - it was a revelation.


----------



## bigshot

ultmusicsnob said:


> Do people have far better vinyl than I've gotten in my purchases? I *still* play all my vinyl records, and I enjoy them, but I would never say they were better than digital, just on the surface noise and pops and clicks alone.




Clean copies of records pressed before the oil crisis can sound very good. In the mid 70s, they started recycling vinyl and quality took a nose dive.


----------



## ferday

analog has limits which cannot be overcome
  
 digital has limits which can theoretically be overcome
  
 preference for either...is just that.  throw audibility of these limits into account, and that's where the blurred lines are...


----------



## bigshot

Thesholds of audibility aren't that fuzzy. Establshed thresholds can get you in the ballpark, and a simple DBT can nail it down nicely for your own particular ears.


----------



## ferday

bigshot said:


> Thesholds of audibility aren't that fuzzy. Establshed thresholds can get you in the ballpark, and a simple DBT can nail it down nicely for your own particular ears.


 
  
 agreed
  
 but there aren't any (valid) arguments from the subjective camp regarding measurements...only about the validity of the ABX/DBT.  it was my political answer to call those blurred lines LOL


----------



## UltMusicSnob

nick_charles said:


> Vinyl is just inherently noisy, it was especially annoying to me back in the 80s trying to listen to quiet passages in classical music, all the imperfections were easily apparent, with headphones it was positively painful. But some recordings are not out on CD and once the music gets to a certain level the noise is drowned out. I still remember my first encounter with CD in Charing Cross Records in 1984. I was in the demo studio and heard the opening bars of Mahler's 1st (Solti/CSO) played back on a Marantz CD63 and there was no noise whatsoever - it was a revelation.


 
 Exactly. Instead of that needle-on-vinyl sound before the music starts, with my first exposure to CD there was **nothing**---then music. As you say, a revelation. I was actually spinning disks on public radio in Kentucky as they made their transition, both vinyl and our new CD's as they came in. I made it a major point during the fund drive periods, to build up our collection of music with this pristine (by comparison) format--no pops and clicks, it was a big deal.


----------



## Don Hills

marone said:


> Not much point in arguing with an objectivist whose belief system is that all that is measurable can be heard and all that you hear can be measured with current technology. Same types made the same arguments 10, 20, 50 years ago.


 
  
 A difference may be measurable, but it may not be audible. However, the corollary is not true. If you can hear a real difference, it can be measured. Current technology is not required, it's been possible to do so for decades.


----------



## proton007

Measurable =!= Audible. Dogs can hear frequencies humans can't.
  
 Audible == Measurable ??  Not always. What you 'hear' is your brain activity. You may see and hear things that don't exist.


----------



## bigshot

How do you apply those ideas to getting better sound out of your music, Proton?


----------



## UltMusicSnob

proton007 said:


> Measurable =!= Audible. Dogs can hear frequencies humans can't.
> 
> Audible == Measurable ??  Not always. What you 'hear' is your brain activity. You may see and hear things that don't exist.


 
 For different but related example, in a large orchestral texture, the melody may be played at unison by strings, flutes, oboes, clarinets. The conductor halts the rehearsal and says "2nd clarinet, you're not blending in at the 3rd bar, please match the tone of the 1st chair there."
  
 The engineers cannot parse the 2nd clarinet's sound out of the mixdown signal (obviously they could if every part was separately miked). The conductor's brain can take the binaural signal arriving at his/her ears and distinguish every characteristic of every part. The engineer may be able to run stats and see a slightly different spectrum, different peaks, maybe a vanishingly small RMS amplitude difference--but they can neither measure nor point to the difference that the conductor hears, the one s/he cares about.
  
 This happens at virtually every orchestra rehearsal (with a decent conductor) everywhere. It's not quite the same as "measuring"; probably this should be called "decoding the signal". Your brain can separate the components of the sound in ways that are not available via hardware or software.


----------



## proton007

bigshot said:


> How do you apply those ideas to getting better sound out of your music, Proton?


 
  
  
 Well, I've kept my DAC + Amp as clean and neutral as I can (ODAC + O2), use some bit of calculation to match the impedance and power (as in don't get over/underpowered equipment) ,  and use the headphone for the variation in sound.
  
 I don't justify buying expensive cables and equipment for a negligible real difference, something that might not even be noticeable, and always consider the possibility that whatever difference I'm hearing might just be placebo, or might be influenced by my mood at that time.
  
 The job of the audio chain is to faithfully reproduce the 'signal' from digital to analog. An ideal chain should disappear from the overall system, seemingly linking the source and the transducer directly.


----------



## proton007

ultmusicsnob said:


> For different but related example, in a large orchestral texture, the melody may be played at unison by strings, flutes, oboes, clarinets. The conductor halts the rehearsal and says "2nd clarinet, you're not blending in at the 3rd bar, please match the tone of the 1st chair there."
> 
> The engineers cannot parse the 2nd clarinet's sound out of the mixdown signal (obviously they could if every part was separately miked). The conductor's brain can take the binaural signal arriving at his/her ears and distinguish every characteristic of every part. The engineer may be able to run stats and see a slightly different spectrum, different peaks, maybe a vanishingly small RMS amplitude difference--but they can neither measure nor point to the difference that the conductor hears, the one s/he cares about.
> 
> This happens at virtually every orchestra rehearsal (with a decent conductor) everywhere. It's not quite the same as "measuring"; probably this should be called "decoding the signal". Your brain can separate the components of the sound in ways that are not available via hardware or software.


 
  
  
 I'm pretty sure its possible to analyze all the different notes being played. Signal analysis is pretty sophisticated nowadays.
 Or by 'blending' if you mean 'musically pleasing' then its a different concept altogether.


----------



## Makiah S

strangelove424 said:


> Just when I thought I found the sane and rational Head-Fi sub-forum.




lol agreed. I also support the claim that 24bit is n improvenent over 16. But sound wise they offer the same sonic quality

nice view as well proton. Ive always thought of custom cables as a.visual bonus. Monprice rca sounds just as nice as Monster what evers


----------



## marone

don hills said:


> If you can hear a real difference, it can be measured. Current technology is not required, it's been possible to do so for decades.




So not true. The human senses pick up things that cannot be yet measured.

Explain the intuition of being watched, stared at or hunted?

Impossible by science - there is no way you could tell someone is looking at you from behind, or that an animal is tracking you.



ultmusicsnob said:


> nick_charles said:
> 
> 
> > Vinyl is just inherently noisy, it was especially annoying to me back in the 80s trying to listen to quiet passages in classical music, all the imperfections were easily apparent, with headphones it was positively painful.
> ...




Both of you lack the ability to hear past the clicks and surface noise, to the more life-like mid-range and body of the instruments in the sound-field. I can ignore those artifacts and hear the more realistic and engaging audio from a FLAC rip of a vinyl source, or an AM radio. FLAC rips of vinyl sound better than the CD of same except in rare cases of a DSD or super FLAC direct master from the non-PCM master.

You cannot. That's preference, not better sound.

When you traded the LP for CD because there was 'nothing but music', you lost tonal accuracy, mid-range realism, clear highs, and many other aspects that are superior to vinyl and are - just now, 35 years later - being matched and bettered by the DSD format.

To you the artifacts of the needle, wow, flutter, rumble, inner groove compression, RIAA curve, and all the rest were more important than the fact that the mid-range was killed by 16-bit and a digital sheen was overlaid upon the music and the highs became grating.

It's similar to listening to state of the art SS recordings from the very early 1960's and comparing them to valve recordings 5 years prior. The SS has less distortion and a quieter background, all measurable of course, but the SS lacks the echo, reverb, decay and lifelike mid-range that the valve recordings have and had - all not measurable but audible.

This is very apparent on Streisand very early recordings where the SS dead air is apparent. Then compare to anything recorded in the late 1950's. Compare the SS recordings of the late 60's with the Living Presence or Mercury releases of 10 years prior.

There is little point in this discussion. You have science as your belief system, and I use my ears.

Neither of us will ever convince the other.


----------



## stv014

> Originally Posted by *UltMusicSnob* /img/forum/go_quote.gif
> 
> The engineers cannot parse the 2nd clarinet's sound out of the mixdown signal (obviously they could if every part was separately miked). The conductor's brain can take the binaural signal arriving at his/her ears and distinguish every characteristic of every part. The engineer may be able to run stats and see a slightly different spectrum, different peaks, maybe a vanishingly small RMS amplitude difference--but they can neither measure nor point to the difference that the conductor hears, the one s/he cares about.


 
  
 Analyzing the music itself, and simply making sure it is reproduced accurately are two separate issues. The latter just needs accurate hardware, which is already available and has been for some time, while the former is more of a software (even artificial intelligence if you want to make it really advanced) issue; after all, humans can still separate the parts of music, recognize voices, etc. with 22 kHz 8-bit format digital signals, so it is not a question of "resolution".


----------



## stv014

marone said:


> There is little point in this discussion. You have science as your belief system, and I use my ears.


 
  
 So, why not try using your ears (and only those, rather than your imagination) in some tests that have been suggested before ?


----------



## xnor

marone said:


> There is little point in this discussion. You have science as your belief system, and I use my ears.


 
 Yeah, we're interested in reality and, at least I am, interested in minimizing faith. If you're interested in the imagined and maximizing ignorance then go ahead, no one is stopping you. But think about where you're posting.
  


> Neither of us will ever convince the other.


 
 That may be very well true for you. It's called closed-mindedness.
  
 By now it's pretty much clear you're just spouting things you believe to be true (some of which are blatantly wrong) and stir things up. We don't need that.
  
 If you have a proper argument as to why 24 bits are better during playback please let us know. We're open-minded and very interested in sound arguments and evidence, ready to change our minds.


----------



## jaddie

marone said:


> So not true. The human senses pick up things that cannot be yet measured.
> 
> Explain the intuition of being watched, stared at or hunted?
> 
> Impossible by science - there is no way you could tell someone is looking at you from behind, or that an animal is tracking you.


 
  What's described here are "feelings", not sensory input.  You're right, so far we can't measure feelings.  We can, however, measure the stimulus that becomes sensory input.
  
 Quote:


marone said:


> When you traded the LP for CD because there was 'nothing but music', you lost tonal accuracy, mid-range realism, clear highs, and many other aspects that are superior to vinyl and are - just now, 35 years later - being matched and bettered by the DSD format.
> 
> To you the artifacts of the needle, wow, flutter, rumble, inner groove compression, RIAA curve, and all the rest were more important than the fact that the mid-range was killed by 16-bit and a digital sheen was overlaid upon the music and the highs became grating.


 
 Please allow me to speak as someone who has had an opportunity most listeners will never have, that of comparing the direct un-recorded output of an analog mixing console to the exact signal passed through A/D and D/A.  We had the return of an early Sony digital recording system sent back to the console monitor selector switch, which permitted a direct A/B comparison to un-ditigized "live" vs digitally recorded and reproduced.  In sighted A/B tests, we would often think we could pick the digital version out, but in unsighted A/B tests, our best and youngest ears scored 50/50 (guessing).  In fact, the monitor selector often got left in the digital monitor position by accident, and nobody in the control room noticed until the looked at the switch.  However, if we played a vinyl record in the same room, in sync, and with accurate and verified RIAA eq, and compared it to the digital master, the difference was always clear and obvious.  On pristine vinyl with very low noise and no defects, we could hear higher distortion in the vinyl, though from a frequency response, stereo perspective, reverb, and dynamics viewpoint they sounded nearly identical.  None of that amazing vinyl sound everyone expects.  When you compare vinyl and digital, you aren't comparing the same recordings and mastering, you're comparing two entirely different signal chains.  No fair, and meaningless.
  
  


> It's similar to listening to state of the art SS recordings from the very early 1960's and comparing them to valve recordings 5 years prior. The SS has less distortion and a quieter background, all measurable of course, but the SS lacks the echo, reverb, decay and lifelike mid-range that the valve recordings have and had - all not measurable but audible.
> 
> This is very apparent on Streisand very early recordings where the SS dead air is apparent. Then compare to anything recorded in the late 1950's. Compare the SS recordings of the late 60's with the Living Presence or Mercury releases of 10 years prior.
> 
> ...


  

 Again, this is comparing apples and pork chops.  Two different recordings, 10 years apart, different studios, mics, acoustics, producers, and you conclude the older one is better because it didin't use a particular technology, which had the least impact of any of those things I just listed.
  
 After working in the industry for 40+ years (audio engineering, broadcasting, recording, music production, etc.), looking for to see if analog anything is better than digital (since it's introduction) I'm sorry, I can't find it.  I find reasons why the results are different, but they are all, and I mean ALL related to decisions humans made along the line, not the technology itself.  I've made masters both ways simultaneously from the same console output.  Digital was always better, no question, completely transparent, an identical copy of the console output.
  
 There are some really horrible analog/vinyl recordings too.  As one example, RCA "Living Presence" of Chicago Symphony/Reiner playing "Mysterious Mountain" by Alan Hovhaness.  I have the vinyl and the "remastered" cd.  Both are loaded with distortion and intermod.  The simply hit tape way to hard.  The record suffers from sub-standard vinyl, so it's full of defects (tried several copies, all are this way), and the CD has none of that.  Both have exactly the same soundstage.  The performance was spectacular, and RCA captured Reiner's string sound, but smashed the daylights out of high levels.  So much for analog.
  
 You can find good and bad examples in all types of recordings. It's people, not technology.


----------



## gjfs

marone said:


> So not true. The human senses pick up things that cannot be yet measured.


 
 That you don't know how to measure something does not make it unmeasurable.


----------



## Makiah S

jaddie said:


> Please allow me to speak as someone who has had an opportunity most listeners will never have, that of comparing the direct un-recorded output of an analog mixing console to the exact signal passed through A/D and D/A.  We had the return of an early Sony digital recording system sent back to the console monitor selector switch, which permitted a direct A/B comparison to un-ditigized "live" vs digitally recorded and reproduced.  In sighted A/B tests, we would often think we could pick the digital version out, but in unsighted A/B tests, our best and youngest ears scored 50/50 (guessing).  In fact, the monitor selector often got left in the digital monitor position by accident, and nobody in the control room noticed until the looked at the switch.  However, if we played a vinyl record in the same room, in sync, and with accurate and verified RIAA eq, and compared it to the digital master, the difference was always clear and obvious.  On pristine vinyl with very low noise and no defects, we could hear higher distortion in the vinyl, though from a frequency response, stereo perspective, reverb, and dynamics viewpoint they sounded nearly identical.  None of that amazing vinyl sound everyone expects.  When you compare vinyl and digital, you aren't comparing the same recordings and mastering, you're comparing two entirely different signal chains.  No fair, and meaningless.
> 
> 
> Again, this is comparing apples and pork chops.  Two different recordings, 10 years apart, different studios, mics, acoustics, producers, and you conclude the older one is better because it didin't use a particular technology, which had the least impact of any of those things I just listed.
> ...


 
  
 Nice read thank you for chimming in here! 
  
 Like the last part as well


----------



## Speedskater

ultmusicsnob said:


> For different but related example, in a large orchestral texture, the melody may be played at unison by strings, flutes, oboes, clarinets. The conductor halts the rehearsal and says "2nd clarinet, you're not blending in at the 3rd bar, please match the tone of the 1st chair there."
> The engineers cannot parse the 2nd clarinet's sound out of the mixdown signal (obviously they could if every part was separately miked). The conductor's brain can take the binaural signal arriving at his/her ears and distinguish every characteristic of every part. The engineer may be able to run stats and see a slightly different spectrum, different peaks, maybe a vanishingly small RMS amplitude difference--but they can neither measure nor point to the difference that the conductor hears, the one s/he cares about.
> This happens at virtually every orchestra rehearsal (with a decent conductor) everywhere. It's not quite the same as "measuring"; probably this should be called "decoding the signal". Your brain can separate the components of the sound in ways that are not available via hardware or software.


 
 If you were to place binaural microphones (sort of like headphones) on that conductor's head and make a recording.
 See:
Stereo Recording & Rendering - 101
 http://www.linkwitzlab.com/Recording/AS_creation.htm
  
 Then another equally skilled conductor could hear the same problem when listening to the playback.


----------



## Makiah S

speedskater said:


> If you were to place binaural microphones (sort of like headphones) on that conductor's head and make a recording.
> See:
> Stereo Recording & Rendering - 101
> http://www.linkwitzlab.com/Recording/AS_creation.htm
> ...


 
  
 Now didn't 24bit offer more room for mastering? Your article helped confirm this [as did Adelle] but obviously recodring and mastering have a big effect on sound [duh] and this thread has nothing to do with that, as a track mastered in 24bit then dwn sampled to 16 bit will sound much the same, but most often we don't find 24bit tracks down sampled. I know in my case my chesky Albums are dwn sampled to 24bit from what 32bit 192Sample
  
 So I suppose the only reason I own any 24bit it out of laziness, as I agree that 16 bit and 24 bit sound the same 
  
 Still honestly shouldn't we start a new thread, [Sorry I'm,  doing algebra home work and I can hear my Proffessor saying "Answer the question asked, don't just give me numbers]
  
 That being said, what is the question we are even talking about any more? [And honestly I only ask that our of curiosity ] that and


----------



## UltMusicSnob

proton007 said:


> I'm pretty sure its possible to analyze all the different notes being played. Signal analysis is pretty sophisticated nowadays.
> Or by 'blending' if you mean 'musically pleasing' then its a different concept altogether.


 
 Sometimes--also depends on noise. But in this case the notes are not different. The winds and strings are unison on the melody (very common in orchestral music). No current system can separate out the different instruments playing that one same note together--but the brain can.


----------



## UltMusicSnob

speedskater said:


> If you were to place binaural microphones (sort of like headphones) on that conductor's head and make a recording.
> See:
> Stereo Recording & Rendering - 101
> http://www.linkwitzlab.com/Recording/AS_creation.htm
> ...


 
 Yes, exactly. This goes more to "understanding" or "decoding" the signal rather than measuring it per se, in the ways recording engineers usually think about. In the example you reference above, the decoder in both cases is a human brain. Perhaps not forever, but we're still ahead of the machines in this respect!


----------



## UltMusicSnob

marone said:


> Both of you lack the ability to hear past the clicks and surface noise, to the more life-like mid-range and body of the instruments in the sound-field.





> Umm, wrong. Technically, that's a non sequitur. Don't mistake an appreciation for lack of surface noise in digital for a lack of appreciation for vinyl, or lack of ability generally.


----------



## jaddie

speedskater said:


> If you were to place binaural microphones (sort of like headphones) on that conductor's head and make a recording.
> See:
> Stereo Recording & Rendering - 101
> http://www.linkwitzlab.com/Recording/AS_creation.htm
> ...


 
  
 Sorta...
  
 The problem with binaural has always been the differences between individuals hears, heads, chests, etc.  For the individual, his own "set" is learned or burned into the brain, so it decodes life really well.  But if presented with a binaural recording made with someone (or something) else's set of hears, head, chest, etc., it may not match well.  Add the possible non-perceptually-flat response of a set of headphones, and the precision of dimensional reproduction begins to fail, often quite badly. And, in some cases, it still works well enough.  But but if you record with mics in your own ears, and play back to yourself, binaural can be astounding.


----------



## marone

Maybe you guys simply cannot hear what others can.

To assume everyone has the same ability to hear detail would be foolish, and to assume that every aspect of hearing that enters into the brain interface can be measured by something as crude as FR or spectral analysis would assume we know all there is and history has shown that to be false. Techniques have improved with time. Would you be able to hear something in the future that could be measured then, but not measured now?

I am listening right now on 668b's to an mp3 320/44.1 rip of the Act's 'Too Late at 20' and before that a rip of the CD of The Woodentops 'Giant'. The rip of the vinyl sounds as I recall the record sounding and the CD rip does not.

Yes, there is no surface noise on the Giant rip but much of the detail, warmth, upper bass, midrange and room behind the instruments is gone. This agrees with the sound of the record and subsequent CD, both of which I owned and compared.

Compare that to an amateur home PC rip on likely a very cheap turntable and needle setup of The Act, which has so much more body to the instruments with internal tonal and harmonic continuity.

In other words the vinyl rip sounds better and more lifelike even though one can hear the surface noise and other vinyl artifacts. Even though the CD was professionally done and the vinyl rip was probably on a $30 rig. Even though the cheap electronics of the vinyl rip chain are obvious, the body of the instruments and sound is simply better.

You cannot hear this - that's fine.

But don't hide behind science as your belief system to justify your opinions and bias.

I state plainly that I can hear this and no amount of science will alter this as I know what I know, see what I see, and hear what I hear. I know what I state is opinion and I stick to it. You hide behind your opinion as being 'real' and based on 'science' and being 'objective' when it isn't.

I would be much more accepting of your point of view if you merely confessed that your bias is to believe science knows it all now and that you cannot hear what cannot currently be measured, even though what can be measured has changed over time.

Would you then be able to hear it because it could be measured at some future date with better technology?


----------



## skamp

marone said:


> so much more body to the instruments with internal tonal and harmonic continuity.




What the _hell_ does that even mean?



marone said:


> I state plainly that I can hear this and no amount of science will alter this as I know what I know, see what I see, and hear what I hear.




Except that both your eyes and your ears can easily be fooled. You don't have to be a scientist to know that.

Also, I thought you said you couldn't convince anyone, and no-one could convince you?


----------



## stv014

marone said:


> I am listening right now on 668b's to an mp3 320/44.1 rip of the Act's 'Too Late at 20' and before that a rip of the CD of The Woodentops 'Giant'. The rip of the vinyl sounds as I recall the record sounding and the CD rip does not.


 
  
 The point of this thread is that for normal music listening, 16 bit PCM format is enough. Since your vinyl rip has already been in 44.1/16 format, and then even converted to MP3, I am not sure what you are trying to prove, especially comparing two different tracks.


----------



## Don Hills

marone said:


> Maybe you guys simply cannot hear what others can.


 
  
 In this case, just about anyone should be able to hear it.
  


> To assume everyone has the same ability to hear detail would be foolish, and to assume that every aspect of hearing that enters into the brain interface can be measured by something as crude as FR or spectral analysis would assume we know all there is and history has shown that to be false. Techniques have improved with time. Would you be able to hear something in the future that could be measured then, but not measured now?


 
  
 If there is an audible difference - a real difference, detectable in a blind test - the difference is measurable. It's always been been measurable, the only thing that has changed over time is the ease and precision of measurement.
  


> ...  The rip of the vinyl sounds as I recall the record sounding and the CD rip does not.


 
  
 Well, of course. I doubt there are many here who would have difficulty telling which is a rip of the LP and which is a rip of the CD. 
  
 The crucial point is that your references are the original LPs of the recordings. To you, LP sounds "better, more lifelike" than CD. That is your expressed opinion. You're entitled to it, but it is no more valid than the countering opinion that the CD sounds better because it doesn't contain the distortions and imperfections that the LP does. What you call the "detail, warmth, upper bass, midrange and room behind the instruments" of the LP is another person's "euphonious distortion".
  


> ...  You cannot hear this - that's fine.
> 
> But don't hide behind science as your belief system to justify your opinions and bias. ....


 
  
 They can indeed hear it. And unlike you, they don't have to rely on faith or dogma as a belief system. They can point to real physical differences to justify their opinions.


----------



## Makiah S

don hills said:


> The crucial point is that your references are the original LPs. To you, they sound "better, more lifelike" than the CDs. That is your expressed opinion. You're entitled to it, but it is no more valid than the countering opinion that the CD rip sounds better because it doesn't contain the distortions and imperfections that the LP rip does. What you call the "detail, warmth, upper bass, midrange and room behind the instruments" of the LP rip is another person's "euphonious distortion".
> 
> 
> They can indeed hear it. And unlike you, they don't have to rely on faith or dogma as a belief system. They can point to real physical differences to justify their opinions.


 
 Agree'd an honestly imo. Digital has always sounded better, but some genres of music do sound nice with the extra noise and color of an OLD Lp player, But what does this have to do with the Thread topic? In addition most of the 94bit Vinyl records I've gotten [and I have  GOOD source mind you] sound a little grainy and a touch noisy... the Beyers don't hide that like other cans might. So I still prefer CD to Vinyl rips


----------



## jaddie

marone said:


> I am listening right now on 668b's to an mp3 320/44.1 rip of the Act's 'Too Late at 20' and before that a rip of the CD of The Woodentops 'Giant'. _*The rip of the vinyl sounds as I recall the record sounding and the CD rip does not.*_


 
  The reference memory was of vinyl, so why wouldn't the rip sound like....well, vinyl?  I'm not going to bother with the discussion of how auditory memory works...or doesn't.
 Quote:


marone said:


> Yes, there is no surface noise on the Giant rip but much of the detail, warmth, upper bass, midrange and room behind the instruments is gone. This agrees with the sound of the record and subsequent CD, both of which I owned and compared.
> 
> Compare that to an amateur home PC rip on likely a very cheap turntable and needle setup of The Act, which has so much more body to the instruments with internal tonal and harmonic continuity.
> 
> In other words the vinyl rip sounds better and more lifelike even though one can hear the surface noise and other vinyl artifacts. Even though the CD was professionally done and the vinyl rip was probably on a $30 rig. Even though the cheap electronics of the vinyl rip chain are obvious, the body of the instruments and sound is simply better.





>


 
 So, and we've talked about this before, the total path the audio takes to get to your years for the CD vs the vinyl or vinyl rip is different.  The people who mastered the CD weren't the same people who mastered the vinyl, and would not have made the same decisions.  It may not have even been the same master tape.  If you start with the same original tape, and do nothing different in creation of the vinyl and CD master, other than what it normally takes to compensate for the losses in vinyl, you end up with identically sounding vinyl and CDs except for the additional noise and distortion of vinyl.  Identical.  Not different in any other way.  Did I say "identical"?   I've done this, twice.  When you get the entire process and all systems under control, the vinyl/digital differences vanish except for surface noise, record wear, and tracking related distortion, which on a first play of a new pressing are pretty darn small.  All comparisons of vinyl to CD have in common the complete lack of knowledge of the origin of each.  Since the creation of the two masters span time and often space, and people, you can be sure there will be differences.  
  
 Now, here's the key.  A lot of older vinyl was mastered by people who really knew there stuff.  A lot of CDs, in the past and today, are mastered by people who either don't know their stuff, or are forced into "competitive" decisions by the producer (or guy writing their check).  Yes, in many cases the vinyl sounds better, but it's not because its vinyl. But the important thing here is the vinyl rip, even 16/44.1, will sound indistinguishable from the vinyl, whereas the CD will often sound different.  It's true, it's audible, measurable, and not magic. Just be aware of the cause.


----------



## Makiah S

jaddie said:


> Now, here's the key.  A lot of older vinyl was mastered by people who really knew there stuff.  A lot of CDs, in the past and today, are mastered by people who either don't know their stuff, or are forced into "competitive" decisions by the producer (or guy writing their check).  Yes, in many cases the vinyl sounds better, but it's not because its vinyl. But the important thing here is the vinyl rip, even 16/44.1, will sound indistinguishable from the vinyl, whereas the CD will often sound different.  It's true, it's audible, measurable, and not magic. Just be aware of the cause.


 
 Good point


----------



## xnor

However good the points you make are, don't expect him to read/understand them. After all he's still writing the same nonsense he did 2 pages back with countless corrections in between...


----------



## Makiah S

xnor said:


> However good the points you make are, don't expect him to read/understand them. After all he's still writing the same nonsense he did 2 pages back with countless corrections in between...


 
  
 xnor how do we keep meeting on these... almost pointless threads... I like to learn and most of you guys are pretty knowledgeable but every thread I meet u at.. ends up a repetion of the same things over and over again... 
  
 I say we... ignore him? And... move on to more LEARNING I actually wouldn't knowing the history behind the advances in Digitial Audio... was 16bit the first commonly used format? Or did they start WAY up HIGH in like 32bit


----------



## xnor

For speech you can get away with a lot less. See http://en.wikipedia.org/wiki/G.711 for example.


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## duncan1

Mshenay--= www.research.philips.com/technologies/projects/cd/technology.html  A big debate occurred in the UK when Philips used 14 bit converters to make 16 bit cds they used --oversampling at first 4 times later much more. Many " Golden Ears" argued about the sound quality in relation to the later 16 bit-red-book recordings.  ---but thats subjectivity banned here isnt it?


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## stv014

Modern converters have only a few bits of actual hardware resolution, and use even more oversampling and noise shaping to achieve ~20 bits or even better resolution in the audio band. Of course, "golden ears" still argue in favor of older style R2R converters and no oversampling.


----------



## jaddie

mshenay said:


> move on to more LEARNING I actually wouldn't knowing the history behind the advances in Digitial Audio... was 16bit the first commonly used format? Or did they start WAY up HIGH in like 32bit


 
 Original digital audio recording experiments were 8 bits, mono, to prove the concept, then moved up from there.  The more bit depth, the more challenging from a hardware standpoint.  It was known quite early that 16 bits would be the ultimate goal.
  
 This guy was a pioneer:
 http://en.wikipedia.org/wiki/Thomas_Stockham
  
 The article doesn't mention it, but he did 8 bit digital recordings in the 1960s which were actually pretty astounding. A/D was at high bit depths and adequate sampling rates was a big challenge, as was getting that data on and off a storage medium.  Data tapes were one of the original storage mediums.  HDDs were not even around, and when they were they had very low packing density initially.  Dr. Stockham's Soundstream system was 16 bits at 50KHz.
  
 The reason CDs ended up at 44.1KHz is that the original recoding system used a slightly modified 3/4" U-Matic video recorder with audio data formatted to fit within a video frame.  44.1KHz at 16 bits in stereo was a rate that matched with the horizontal and vertical scanning frequencies of NTSC video (monochrome) at 30 frames per second, 60 fields per second and a horizontal scan frequency of 15,750Hz.  Consumer video-based digital audio recorders initially sampled at 44.056KHz so they could utilize unmodified consumer video cassette recorders operating at the NTSC color frame rate of 29.97 frames per second, and horizontal scan of 15,734.26 Hz. That data could be placed on a redbook CD with negligible upward pitch shift.  Sony made the PCM-F1 and a related family of digital audio recoding adapters for video recorders.  They were all capable of 16 bits or 14 bits at user discretion.  Other manufacturers made similar devices, but stopped with 14 bits.


----------



## Digitalchkn

ferday said:


> analog has limits which cannot be overcome
> 
> digital has limits which can theoretically be overcome


 
  
 One way to think of it is we live real-world that is analog by nature. Digital (or more accurately quantized discrete-time arithmetic) is a mathematical abstraction of real-world. For most real-world signals, digital is only an approximation. Analogue recording is just one possible means of data storage. Digital recording is another method of data storage. The question that should be asked (but is hardly asked) is when is the digital capture of a signal a good approximation of the real signal? At some point the resolution of the sampling process greatly exceeds the resolution of the useful information contained in the signal.


----------



## jaddie

digitalchkn said:


> One way to think of it is we live real-world that is analog by nature. Digital (or more accurately quantized discrete-time arithmetic) is a mathematical abstraction of real-world. For most real-world signals, digital is only an approximation.


 
 Agreed.  But analog is also an abstraction of real-world, and is also only an approximation.


digitalchkn said:


> The question that should be asked (but is hardly asked) is when is the digital capture of a signal a good approximation of the real signal? At some point the resolution of the sampling process greatly exceeds the resolution of the useful information contained in the signal.


 
 Oh, that's been asked, fundamentally, during the development of digital recording.  And, we have our answer, been using it for some time.  The test is if someone can discern a signal passed through A/D > D/A from an original (not recorded) live version.  Been done many times. My own experience was *mentioned here.* And that was in the mid 1980s.
  
 The question isn't asked much, informally, today probably because the test is hard to do, and at this stage, pretty much proven.


----------



## nick_charles

digitalchkn said:


> One way to think of it is we live real-world that is analog by nature. Digital (or more accurately quantized discrete-time arithmetic) is a mathematical abstraction of real-world. For most real-world signals, digital is only an approximation. Analogue recording is just one possible means of data storage. Digital recording is another method of data storage. The question that should be asked (but is hardly asked) is when is the digital capture of a signal a good approximation of the real signal? At some point the resolution of the sampling process greatly exceeds the resolution of the useful information contained in the signal.


 
  
 It also depends on whether you are digitizing an already captured analog signal (say tape) or taking an all digital capture. Even though the SNR of a decent digital capture exceeds the SNR of almost all analog tape it is still imperfect so still adds a very very very small amount of noise. In the context of a medium with an SNR of say 75db the noise added by a a digitization with an SNR of say 96db will be utterly insignificant.
  
 Taking a live source and a direct digital capture thereof will give you something which several members have opined as being completely indistinguishable from the source. However, that is not the same as capturing all the information. You added the proviso "useful information" which we can arbitrarily place where we like in terms of dynamic range and frequency response.


----------



## Digitalchkn

jaddie said:


> Agreed.  But analog is also an abstraction of real-world, and is also only an approximation.
> Oh, that's been asked, fundamentally, during the development of digital recording.  And, we have our answer, been using it for some time.  The test is if someone can discern a signal passed through A/D > D/A from an original (not recorded) live version.  Been done many times. My own experience was *mentioned here.* And that was in the mid 1980s.
> 
> The question isn't asked much, informally, today probably because the test is hard to do, and at this stage, pretty much proven.


 
  
 Sure. Analog recording is an approximation of the real analog signal captured on a magnetic medium such as tape, or mechanical medium such as vinyl.
 By the same token, you can argue that the signal at the far end of a cable is an approximation of the signal at the output of, say, the source.
  
 I also agree this test is hard to do because you are introducing additional signal processing elements in your A/D -> D/A chain and as such are not isolating the actual quantization processes. In your experiments you have to consider the source signal.  How noisy was the source? What is it's spectral characteristics?
  
 At some point, increased precision follows a law of diminishing returns.


----------



## Digitalchkn

nick_charles said:


> It also depends on whether you are digitizing an already captured analog signal (say tape) or taking an all digital capture. Even though the SNR of a decent digital capture exceeds the SNR of almost all analog tape it is still imperfect so still adds a very very very small amount of noise. In the context of a medium with an SNR of say 75db the noise added by a a digitization with an SNR of say 96db will be utterly insignificant.
> 
> 
> 
> ...






 


Digital capture of an analog tape is already a copy of a copy of the input signal.


 


The concept  of "useful information" is not that vague. It is well understood in the context of Shannon's theorem as no loss of information. Anything beyond that is added to the signal is considered as "noise".

In practice, noise is unavoidable, correct? There is noise (you could argue distortion falls into category) added at every processing stage, be it in analog or digital domain. The amount of noise added determines whether information is lost or added in the process of recovering the original "desired" signal.


----------



## jaddie

digitalchkn said:


> nick_charles said:
> 
> 
> > It also depends on whether you are digitizing an already captured analog signal (say tape) or taking an all digital capture. Even though the SNR of a decent digital capture exceeds the SNR of almost all analog tape it is still imperfect so still adds a very very very small amount of noise. In the context of a medium with an SNR of say 75db the noise added by a a digitization with an SNR of say 96db will be utterly insignificant.
> ...


 
 Yes, if the noise added is significantly below the noise floor of the original, it can be considered negligible. Not all processes in the digital domain add noise.  All processes in the analog domain, with the exception of complimentary noise reduction systems, add noise.


----------



## Digitalchkn

jaddie said:


> Not all processes in the digital domain add noise


 
 In practice - sure they do. Even simple operation of gain requires a requantization step back to integer arithmetic (there is an exception of literarly linearly doubling the amplitude by simple bit shifting). Needless to say more complex math operations such as filtering, rate conversions all add computational noise. It's simply that the amount of noise added is typically substantially lower than doing the same operation through physical/analog circuitry.


----------



## jaddie

digitalchkn said:


> In practice - sure they do. Even simple operation of gain requires a requantization step back to integer arithmetic (there is an exception of literarly linearly doubling the amplitude by simple bit shifting). Needless to say more complex math operations such as filtering, rate conversions all add computational noise. It's simply that the amount of noise added is typically substantially lower than doing the same operation through physical/analog circuitry.


 
  
 Sure, but "in practice", it's mostly below the noise floor of the original analog original and A/D.  We're doing that theoretical vs practical thing now.
  
 It's why a lot of original material is recorded at 24 bits, run through post at 24 bits, then down-sampled to 16 bits for release.


----------



## Digitalchkn

jaddie said:


> Sure, but "in practice", it's mostly below the noise floor of the original analog original and A/D.  We're doing that theoretical vs practical thing now.
> 
> It's why a lot of original material is recorded at 24 bits, run through post at 24 bits, then down-sampled to 16 bits for release.


 
 Or >24bits or floating point math.  We are just about always adding noise the further we processing steps we take. Just because it is below the noise floor of the source, doesn't mean it's negligible. Noise sources will add. It's just in analog domain the contribution of noise sources is typically greater.


----------



## jaddie

digitalchkn said:


> Or >24bits or floating point math.  We are just about always adding noise the further we processing steps we take. Just because it is below the noise floor of the source, doesn't mean it's negligible. Noise sources will add. It's just in analog domain the contribution of noise sources is typically greater.


 
  
 Right.  So let's add three bits of dither to 24 bit data containing a digitized analog noise floor of -90dBFS.  And what would be the result?   While we're at it, lets measure the result with that pesky A-weighting filter.
  
 And, if you don't mind being in the real world, lets look at what we really get out of a real-world 24 bit ADC, which is more like 20 bit performance at best.  To that we're adding that processing noise to a noise floor already sitting at -120dBFS which is 24dB above the theoretical already.  
  
 What would the end result be?


----------



## stv014

One dithered quantization to 24 bits adds roughly -144 dBFS A-weighted noise. That would need to be repeated more than 200 times to add up to -120 dB. Internal processing can easily have better than 24 bit PCM resolution, in fact, 64 bit floats are common in software.


----------



## jaddie

stv014 said:


> One dithered quantization to 24 bits adds roughly -144 dBFS A-weighted noise. That would need to be repeated more than 200 times to add up to -120 dB. Internal processing can easily have better than 24 bit PCM resolution, in fact, 64 bit floats are common in software.


 
  
 Uh-huh.  That's what I'm sayin'....


----------



## Digitalchkn

jaddie said:


> Uh-huh.  That's what I'm sayin'....


 
 At some point you have to make it back to integers. So you don't want to do that back and forth too many times. Moreover, it also depends on other factors, for instance the type of windowing is being used to go to freq domain calculations.
  
 Theoretical SNR should be about -144dB, but not A-weighted. I figure with A-weighted SNR should be  even greater.


----------



## stv014

> Theoretical SNR should be about -144dB, but not A-weighted. I figure with A-weighted SNR should be  even greater.


 
  
 With +/- 1 LSB triangular dither (a widely used "standard" dither that is easy to implement), the unweighted SNR at 24-bit resolution is ~141.5 dB, or exactly 23.5 bits. The A-weighted SNR depends on the sample rate, and the spectrum of the noise (white vs. shaped). It is easy to push some of the noise energy of triangular noise into the top octave if it is generated by differentiating 1 LSB uniform distribution noise, and this is even cheaper to compute than triangular white noise. Here is a table that shows the A-weighted SNR at various sample rates with both types of noise (white first), at a measurement bandwidth of 22000 Hz:
  
 44100 Hz: 143.9 dB / 145.5 dB
 48000 Hz: 144.3 dB / 146.1 dB
 96000 Hz: 147.3 dB / 150.9 dB
  
 Dithering 24-bit PCM samples might actually be overkill, and it is in fact often not used (undithered quantization produces some low level distortion, but lower noise level). But with dither, it is easier to calculate the overall noise level for multiple stages of quantization by simply summing the noise power, since the quantization noise is made uncorrelated to the input signal. That is why I assumed TPDF dither in my previous example.


----------



## marone

nick_charles said:


> > *Everyone has bias including you  and me, however if you state facts that are facts and not just opinions then these are facts even if you are particularly attached to them, opinions without empirical backing are just opinions *




What you don't understand is that 'empirical backing' is an opinion and bias, also; that an opinion with empirical backing is 'just opinion' also.


----------



## Digitalchkn

stv014 said:


> undithered quantization produces some low level distortion, but lower noise level


 
 I think dithering will also add non-linear distortion. Either way, dithering is deliberate addition of spectrally shaped noise. We are simply shifting power spectrum of noise elsewhere in frequency to "fool" our ears. We are not actually improving the situation.The further we push it away from useful signal information, the more we preserve the original signal.  There is a benefit of higher than necessary sampling rate from that perspective.
  
 As far as distortion it is not clear to me. So I am still not convinced that superflous processing iterations in computations are all that benign.  Obviously it is more forgiving than generational copies of analog signal on analog mediums, but it is not transparent. Or put it another way, it can be measured.  It's a different debate whether anyone can perceive that or not.


----------



## stv014

digitalchkn said:


> I think dithering will also add non-linear distortion.


 
  
 No, correctly implemented dither will eliminate distortion, and replace it with uncorrelated noise (a demonstration of this can be seen here).
  


> As far as distortion it is not clear to me. So I am still not convinced that superflous processing iterations in computations are all that benign.


 
  
 That depends on what kind of processing is being performed. For many simple effects that can be mathematically well defined (e.g. volume control, filters, delay), the error can be made very small. For some other, more complex ones (like pitch shifting/time stretching or noise reduction), the algorithm used does affect the sound quality, but the issue is usually not numeric precision.


----------



## bigshot

marone said:


> What you don't understand is that 'empirical backing' is an opinion and bias, also; that an opinion with empirical backing is 'just opinion' also.




Empirical evidence lowers the noise floor on opinions.


----------



## bigshot

digitalchkn said:


> Obviously it is more forgiving than generational copies of analog signal on analog mediums, but it is not transparent. It's a different debate whether anyone can perceive that or not.




Beneath the threshold of perception = transparent.


----------



## Digitalchkn

stv014 said:


> No, correctly implemented dither will eliminate distortion, and replace it with uncorrelated noise (a demonstration of this can be seen here).
> 
> 
> That depends on what kind of processing is being performed. For many simple effects that can be mathematically well defined (e.g. volume control, filters, delay), the error can be made very small. For some other, more complex ones (like pitch shifting/time stretching or noise reduction), the algorithm used does affect the sound quality, but the issue is usually not numeric precision.


 
  
 All processing affects sound quality in a way that can be measured. Generally any signal processing step (except the proverbial "ideal unity gain") impacts the signal, be it in digital form or analog form (or the psuedo in-between like high precision float).  The rest of the debate focuses around perception and human factor.


----------



## jaddie

digitalchkn said:


> All processing affects sound quality in a way that can be measured.


Time delay would be an exception. SQ is unaffected in any measurable way.


digitalchkn said:


> The rest of the debate focuses around perception and human factor.


...and until we start recording and producing recordings for some other being, the human factor is all that really matters.


----------



## stv014

digitalchkn said:


> All processing affects sound quality in a way that can be measured.


 
  
 I do not recall saying that is not the case. But much of the time the "artifacts" can be reduced to a level that is negligible for practical purposes. There are more important things to worry about than a -300 dB "noise floor" added by some simple processing performed using 64-bit floats.


----------



## stv014

jaddie said:


> Time delay would be an exception. SQ is unaffected in any measurable way.


 
  
 That is, assuming that the amount of delay is an integer number of samples, which is often an acceptable limitation for simple constant time delays. Otherwise, interpolation is needed.


----------



## Digitalchkn

stv014 said:


> That is, assuming that the amount of delay is an integer number of samples, which is often an acceptable limitation for simple constant time delays. Otherwise, interpolation is needed.


 
 It's ideally an all pass+constant group delay filtering operation. Fractional sample delays are going to deviate from that slightly.


----------



## jaddie

digitalchkn said:


> It's ideally an all pass+constant group delay filtering operation. Fractional sample delays are going to deviate from that slightly.


 
  
 Seriously over-thinking what "time delay" is.  I didn't say "group delay".  Fractional sample delays?  Really, no need.


----------



## Digitalchkn

jaddie said:


> Seriously over-thinking what "time delay" is.  I didn't say "group delay".  Fractional sample delays?  Really, no need.


 
  
 Point taken.  -300dB noise floors is also more than "good enough".   Judging by the title of this thread, so is 16 bit resolution then.


----------



## jaddie

I love a theoretical discussion as much as anyone right up until the theoretical position paints the practical position as "wrong", because that makes everything, analog and digital, "wrong" and fatally flawed.  While true in the absolute, it's far from the reality we deal with every day.  I also greatly appreciate perfection, but until we have it, I'm also more than willing to embrace excellence in what we have in our hands, appreciate and enjoy it, flaws and all.
  
 16 bit, like any tool, is appropriate for some jobs, and inappropriate for others.  The trick is knowing what tool to buy for the best result without overpaying for one that provides no additional benefit.


----------



## UltMusicSnob

jaddie said:


> I love a theoretical discussion as much as anyone right up until the theoretical position paints the practical position as "wrong", because that makes everything, analog and digital, "wrong" and fatally flawed.  While true in the absolute, it's far from the reality we deal with every day.  I also greatly appreciate perfection, but until we have it, I'm also more than willing to embrace excellence in what we have in our hands, appreciate and enjoy it, flaws and all.
> 
> 16 bit, like any tool, is appropriate for some jobs, and inappropriate for others.  The trick is knowing what tool to buy for the best result without overpaying for one that provides no additional benefit.


 
 I'm sure this has been covered before, but, for example, making ambient recordings in very quiet nature environments, I had to either use 24-bit or end up with no useful result.


----------



## bigshot

I haven't been paying attention... have we figured out how many angels can dance on the head of a pin yet?


----------



## proton007

bigshot said:


> I haven't been paying attention... have we figured out how many angels can dance on the head of a pin yet?


 
  
 Let me sum up most of what I've seen:
  
 -- Everything is opinion (sticks fingers in ears) la la la la la .....
 -- Analog good! Digital bad! Hulk Smash!
 -- I'm analog, you're analog. Everything's analog. Digital is analog, analog is digital. Why can't we live in peace?
 -- I can hear and see things no one can. But you won't understand!  
	

	
	
		
		

		
			





 
  
 -- A lot of technical information.


----------



## jaddie

proton007 said:


> Let me sum up most of what I've seen:
> 
> -- Everything is opinion (sticks fingers in ears) la la la la la .....
> -- Analog good! Digital bad! Hulk Smash!
> ...


 
  
 You're killin' me!!!


----------



## jaddie

ultmusicsnob said:


> I'm sure this has been covered before, but, for example, making ambient recordings in very quiet nature environments, I had to either use 24-bit or end up with no useful result.


 
  
 Nah, you were limited by preamp noise and mic self noise.  People been doing that for decades, even pre-digital. You needed a dead-quiet LDC, and a decent pre, and 16 bits woulda worked.


----------



## wnmnkh

After 4 years, this thread is still relevant to these days with all of these silly replies.
  
 But it seems everything got worse and worse with all DSD nonsense has been unleashed for money grab.


----------



## UltMusicSnob

jaddie said:


> Nah, you were limited by preamp noise and mic self noise.  People been doing that for decades, even pre-digital. You needed a dead-quiet LDC, and a decent pre, and 16 bits woulda worked.


 
 No, the microphone was spectacular: Neumann TLM 103 (yes, I carried a pair off into the woods). And the preamp (Aphex 107) was likewise studio quality. I *really did* need 24 bits. The occasional tractor passing on the road or jet overhead meant the true acoustic dynamic range was huge, and I wasn't going to pack in an entire compressor as well. When I opened up the early-inexperienced 16-bit file in Sound Forge, you could literally see the stair steps on the waveform (only after zooming way in--on normal default scale it looked like a flat line with the occasional blip.


----------



## marone

bigshot said:


> Empirical evidence lowers the noise floor on opinions.




That is a belief.

The sooner you understand this, the sooner you can realise your method is neither better nor worse than others.


----------



## Don Hills

Quote:


ultmusicsnob said:


> ... When I opened up the early-inexperienced 16-bit file in Sound Forge, you could literally see the stair steps on the waveform (only after zooming way in--on normal default scale it looked like a flat line with the occasional blip.


 
 What it looked like is irrelevant and misleading. What did it sound like when boosted to the intended listening levels?
  
 What were you recording?


----------



## UltMusicSnob

don hills said:


> Quote:
> What it looked like is irrelevant and misleading. What did it sound like when boosted to the intended listening levels?
> 
> What were you recording?


 
 Quantization noise, of course, lots of it. Boosting a tiny signal like that meant that the resulting amplified signal brought the stair steps with it, so to speak. The recording only used a tiny portion of the bottom end of the dynamic range available in 16 bits. Bringing that up to -18db meant that the lack of sufficient resolution on the vertical axis (differences in voltage) was made painfully apparent.
  
 These were ambient environment nature recordings, dawn and dusk, far out in the countryside (KY).
  
 As another member has suggested to me directly, a very quiet preamp with more available gain would have addressed this challenge before the A/D stage.


----------



## Digitalchkn

wnmnkh said:


> After 4 years, this thread is still relevant to these days with all of these silly replies.
> 
> But it seems everything got worse and worse with all DSD nonsense has been unleashed for money grab.


 
  
 To add fuel to the fire I will go out on the limb and state that 25bits is better than 24bits. This statement should prolong this thread for another year or so.


----------



## bigshot

marone said:


> That is a belief. The sooner you understand this, the sooner you can realise your method is neither better nor worse than others.




All opinions are not created equal. Some are supported by evidence and some are not.


----------



## stv014

digitalchkn said:


> To add fuel to the fire I will go out on the limb and state that 25bits is better than 24bits.


 
  
 Both are old now, 384/32 seems to be the new highest quality "audiophile" format.


----------



## Digitalchkn

stv014 said:


> Both are old now, 384/32 seems to be the new highest quality "audiophile" format.


 
  
 Oh man! Now I have to upgrade that brand new 25bit headamp/dac combo I just spends thousands on again.
  
 Sounds like there's a need to start a new thread.


----------



## SonicSavour

Here is an interesting and highly entertaining video on the subject. 
http://www.youtube.com/watch?v=cIQ9IXSUzuM#t=276
  
 I also recommend Ethan Winers 'The Audio Expert', all of his videos on youtube as well as his articles (for example on audio perception) to clarify common beliefs. The thing is he provides Test tracks and stuff, so you can test yourself rather than loose yourself in endless descussions.


----------



## nick_charles

sonicsavour said:


> Here is an interesting and highly entertaining video on the subject.
> http://www.youtube.com/watch?v=cIQ9IXSUzuM#t=276
> 
> I also recommend Ethan Winers 'The Audio Expert', all of his videos on youtube as well as his articles (for example on audio perception) to clarify common beliefs. The thing is he provides Test tracks and stuff, so you can test yourself rather than loose yourself in endless descussions.


 
  
 This video has been mentioned many times here, a link to it should be a sticky, it is a very useful and extremely understandable explanation.


----------



## Digitalchkn

nick_charles said:


> This video has been mentioned many times here, a link to it should be a sticky, it is a very useful and extremely understandable explanation.


 
  
 There is a minor detail in the explanation about the stairstepped digital waveform appearing as a perfect copy of the analog signal. The theory says that only a perfect reconstruction filter at the output of zero-hold process is required to produce the exact representation of the original sampled signal. Unfortunately, that filter has a impulse response of a perfect brickwall filter) and something that is not physically possible to design in real circuits.


----------



## nick_charles

digitalchkn said:


> There is a minor detail in the explanation about the stairstepped digital waveform appearing as a perfect copy of the analog signal. The theory says that only a perfect reconstruction filter at the output of zero-hold process is required to produce the exact representation of the original sampled signal. Unfortunately, that filter has a impulse response of a perfect brickwall filter) and something that is not physically possible to design in real circuits.


 
  
 Fair point, let's just call it a very good facsimile.


----------



## marone

bigshot said:


> marone said:
> 
> 
> > That is a belief. The sooner you understand this, the sooner you can realise your method is neither better nor worse than others.
> ...




My ears and what I hear are my evidence.

I can accept your belief system (Science) without insult.

Why do you empiricists lack the ability to reciprocate?


----------



## Don Hills

ultmusicsnob said:


> Quantization noise, of course, lots of it.
> 
> These were ambient environment nature recordings, dawn and dusk, far out in the countryside (KY).
> 
> As another member has suggested to me directly, a very quiet preamp with more available gain would have addressed this challenge before the A/D stage.


 
  
 What you should have heard was hiss. Lots of it. With a little ambient nature noise mixed in.
  
 The other member mentioned the most likely cause of the problem - insufficient gain. The loudest internal noise in the system should be the microphone self noise or preamp noise. If you close the microphone in a padded case to eliminate external noise, you need enough preamp gain to bring the recorded noise level up to -80 dBFS or so. 80 dB of dynamic range should be enough unless you're recording thunderstorms or space shuttle launches... (See below.)
  
 I recall reading a piece by a "nature sounds" enthusiast some time ago, where he mentioned that it's getting very hard to find locations in the US where there are long periods (more than 10 minutes or so) of freedom from man-made noise such as traffic or aircraft.
  
 I have the "Thunderstorm in the Rockies" disc by Mobile Fidelity, and it stretches the capabilities of vinyl. I also have Bob Katz's recording of a shuttle launch, in 24/96 4-channel. That's a little too dynamic for 16 bit.


----------



## UltMusicSnob

marone said:


> My ears and what I hear are my evidence.
> 
> I can accept your belief system (Science) without insult.
> 
> Why do you empiricists lack the ability to reciprocate?


 
 This is off into Philosophy of Science.
  
 There is no such thing as private evidence in science. If your interest is "my evidence"--and that's perfectly legitimate as an activity, but it has no bearing on Sound Science--then you have nothing with which to persuade others. The way to proceed persuasively here is to attempt to prove yourself wrong ("Can I actually hear the difference?") and fail to do so (SEE: Karl Popper).


----------



## bigshot

marone said:


> My ears and what I hear are my evidence. I can accept your belief system (Science) without insult. Why do you empiricists lack the ability to reciprocate?




Oh man! That is the EASIEST to answer! Because I can't hear with your ears... and I can't think with your brain. But if you want to make some points that apply to both of us, I'm happy to listen.


----------



## Makiah S

bigshot said:


> Oh man! That is the EASIEST to answer! Because I can't hear with your ears... and I can't think with your brain. But if you want to make some points that apply to both of us, I'm happy to listen.




he walked right into that one


----------



## proton007

bigshot said:


> Oh man! That is the EASIEST to answer! Because I can't hear with your ears... and I can't think with your brain. But if you want to make some points that apply to both of us, I'm happy to listen.


 
  
 BOOM !
 HEADSHOT


----------



## dizzyorange

sonicsavour said:


> Here is an interesting and highly entertaining video on the subject.
> http://www.youtube.com/watch?v=cIQ9IXSUzuM#t=276
> 
> I also recommend Ethan Winers 'The Audio Expert', all of his videos on youtube as well as his articles (for example on audio perception) to clarify common beliefs. The thing is he provides Test tracks and stuff, so you can test yourself rather than loose yourself in endless descussions.


 
  
 this is such an awesome video.


----------



## stv014

nick_charles said:


> Fair point, let's just call it a very good facsimile.


 
  
 It can be very good if the frequency is not too close to Fs/2. For example, at 44100 Hz sample rate, an impulse response length of only 2 ms is enough to attenuate the 24100 Hz image of a 20 kHz tone by more than 120 dB. Although the oversampled signal still needs to be reconstructed with an analog filter, so there will still be some images, but far outside the audio band, and at a relatively low level.


----------



## jcx

digitalchkn said:


> There is a minor detail in the explanation about the stairstepped digital waveform appearing as a perfect copy of the analog signal. The theory says that only a perfect reconstruction filter at the output of zero-hold process is required to produce the exact representation of the original sampled signal. Unfortunately, that filter has a impulse response of a perfect brick wall filter) and something that is not physically possible to design in real circuits.


 
  
 in theory the perfect reconstruction filter likes "perfect" weighted Dirac Impulse train "analog" input
  
  
 a zero order hold has in band frequency response roll off and requires EQ even with a perfect reconstruction filter
  
  
 of course zero order hold is a lot easier to produce with our current electronics - so you always see the stair step waveform in popularized explanations


----------



## Digitalchkn

jcx said:


> in theory the perfect reconstruction filter likes "perfect" weighted Dirac Impulse train "analog" input
> 
> 
> a zero order hold has in band frequency response roll off and requires EQ even with a perfect reconstruction filter
> ...


 
  
 But I guess nobody produces a traditional binary-weighted DAC anymore, so this issue is somewhat moot. Delta-sigma is the way.


----------



## jcx

even the last generation of full resolution "flagship" grade audio ladder DACs were advertised for at least 8x oversampling
  
 so upsampling with digital filtering, correction were here already
  
 but yes multi-bit delta sigma seem to have won the market
  
 but there are a few hold outs claiming delta-sigma noise shaping loop dynamics, "noise floor modulation" is evidence of some mysterious musical soul destroying "time domain error" that engineers are overlooking


----------



## Digitalchkn

jcx said:


> even the last generation of full resolution "flagship" grade audio ladder DACs were advertised for at least 8x oversampling
> 
> so upsampling with digital filtering, correction were here already
> 
> ...


 
  
 It's probably easier, hence cheaper than a precision controlled ladder DAC. Could be a simple case of economics overruling the performance benefits.
  
 Adding any kind of noise (autocorrelated to source signal or not) is by definition a "time domain error".  Modulating the noise is an intruguing notion that probably ultimately depends ultimately on the ranges of SNR achieved. If it's "lowenough" then it's "good enough"


----------



## jcx

recently discussed at diyAudio - one of my replies:
  


> I think I have all of ESS public app notes, white papers
> 
> all I can find is one graph suggesting the "bad competitor" DAC audio band noise floor rose ~ 10 dB from -117 dB to -106-7 dB as signal amplitude rose into the top -10 dB to 0 dB fs of the converter
> 
> ...


----------



## stv014

For testing such problems, difference extraction like shown here can be useful. If the residual, with a huge amount of gain, still does not have any plainly audible distortion or other artifacts, then there is most likely to be no real - practically relevant - issue.


----------



## SonicSavour

marone said:


> My ears and what I hear are my evidence.
> 
> I can accept your belief system (Science) without insult.
> 
> Why do you empiricists lack the ability to reciprocate?


 
  
 Ya right. Here is what your vision alone(!) can do to your audio signal chain.
http://www.youtube.com/watch?v=G-lN8vWm3m0
 No digital, no wires. Purely analog human hearing and perception. Highly fascinating field by the way. Where again was your evidence in hearing? It is true that you are free to believe what you want but unless you didn't reliably test yourself and hear a difference in blind-tests or abx-testing, there is no evidence and a high probability that you imagine the difference. Believe what you want personaly, however, the problem arises when people seek advice on if they should buy high-rez files or expensive gear and they are told without evidence that there is a huuuge difference. It is misleading and the basis of so many fraud in the audio world. That's why we can't reciprocate.


----------



## dizzyorange

sonicsavour said:


> Ya right. Here is what your vision alone(!) can do to your audio signal chain.
> http://www.youtube.com/watch?v=G-lN8vWm3m0
> No digital, no wires. Purely analog human hearing and perception. Highly fascinating field by the way. Where again was your evidence in hearing? It is true that you are free to believe what you want but unless you didn't reliably test yourself and hear a difference in blind-tests or abx-testing, there is no evidence and a high probability that you imagine the difference. Believe what you want personaly, however, the problem arises when people seek advice on if they should buy high-rez files or expensive gear and they are told without evidence that there is a huuuge difference. It is misleading and the basis of so many fraud in the audio world. That's why we can't reciprocate.


 
  
 That is nuts!


----------



## Makiah S

HOLD up, totally un realted but xnor is BANNED... when did that happen :O I was JUST reading his posts here. And I find a thread with him and I see that his title is banned? Is that an error on my part or is he really banned [if he is REALLY banned... I won't ask why] just curious
  
 also this my be unrealted but listened to my fully balanced DT 880 today, from dac to amp to can... 24bit still sounded just as spacious as 16


----------



## stv014

mshenay said:


> when did that happen :O


 
   
I think about a week ago, so he might be back soon.


----------



## Makiah S

stv014 said:


> I think about a week ago, so he might be back soon.


 
  
 *face palm* sheesh, well I guess we'll see


----------



## bigshot

We all take turns being thrown out into the street here in Sound Science. I'm sure Galileo and Darwin were treated like this by the rabble with their torches and pitchforks too.


----------



## Marleybob217

It's probably already been mentioned a hundred times in this thread, but for some reason 24 bit albums are mastered differently. Does anyone know why exactly?
  
 So it is quite possible to hear a difference between 16 and 24 bit, but not because of added resolution.


----------



## bigshot

They master at 24 bit so the filters they apply for sweetening can take advantage of the lower noise floor and higher sampling rate. Once it's processed, it sounds exactly the same as the 16 bit bouncedown.

High bitrates are an advantage for mixing and mastering. For normal listening, redbook is all you need.


----------



## Marleybob217

bigshot said:


> They master at 24 bit so the filters they apply for sweetening can take advantage of the lower noise floor and higher sampling rate. Once it's processed, it sounds exactly the same as the 16 bit bouncedown.
> 
> High bitrates are an advantage for mixing and mastering. For normal listening, redbook is all you need.


 
  
 So people claiming to hear a difference based on the different mastering, are BS-ing?


----------



## UltMusicSnob

marleybob217 said:


> So people claiming to hear a difference based on the different mastering, are BS-ing?


 
 The people claiming can always run a double-blind test in ABX and find out for sure.


----------



## bigshot

marleybob217 said:


> So people claiming to hear a difference based on the different mastering, are BS-ing?




No not at all. The mastering is different, so it's going to sound different.

Again, 24 bit is useful for the sorts of filtering and sweetening done as a part of the mastering process. It helps to have a wider range to draw from when you are doing noise reduction and equalization. 24 bit mastering can sound better than 16 bit mastering.

However, once you bounce the finished, mastered track from 24 bit to 16 bit to put it on a CD, there is no audible difference between the 24 bit and 16 bit bouncedown. The higher bitrate is only an advantage in MASTERING. There is no advantage to high bitrates when just listening to a track on your home stereo.

Is that clearer?


----------



## Marleybob217

bigshot said:


> No not at all. The mastering is different, so it's going to sound different.
> 
> Again, 24 bit is useful for the sorts of filtering and sweetening done as a part of the mastering process. It helps to have a wider range to draw from when you are doing noise reduction and equalization. 24 bit mastering can sound better than 16 bit mastering.
> 
> ...


 
  
 Very clear! Thanks


----------



## stv014

marleybob217 said:


> It's probably already been mentioned a hundred times in this thread, but for some reason 24 bit albums are mastered differently. Does anyone know why exactly?


 
  
 Because they target different markets. It has nothing to do with any technical limitations of the formats. The high dynamic range "audiophile" version would still sound fine converted to 44.1/16 Red Book format, but it would be then less obvious to the casual consumers why they would want to buy it (again).


----------



## stv014

marleybob217 said:


> So people claiming to hear a difference based on the different mastering, are BS-ing?


 
  
 If the mastering is different, they can of course hear a real difference, and the "high resolution" version is indeed usually better. Just do not confuse correlation and causation, the mastering is not _always_ different, and when it is not worse for the CD quality track, it can actually sound great, too.


----------



## UltMusicSnob

Without knowing if this is ever the case, I can imagine at least one reason for mastering differently. I recently took a track I'd created with *lots* of bass energy into a large auditorium with very poor speakers--lucky I did not destroy them with this track. They were flapping and banging for a second there, before I hit the volume and dialed it way back. 
   I have another assignment now that will have to be played in that same venue, and I'm going to pull everything below 60 Hz, and attenuate everything up to at least 120 Hz. I'll test it onsite beforehand, but I expect to have to give a little hype to the harmonics of the instruments so effected, in order to give an impression of sufficient volume, but without actually hitting the fundamental hard.
   If I had "audiophile" as my target market, I would assume as a mastering engineer that my listeners will have equipment with full extension and good detail, so I could master without these sorts of workarounds.


----------



## xnor

I don't know if you know dubstep, but there are tracks with pretty *low bass* (~35 Hz) and *LOTS *of it. I mean completely overpowering amounts. You can find such tracks on ordinary CDs.
  
  
 I would understand this for a "radio edit/version", but for normal CD botching the sound sounds like a cheap excuse to me. 
	

	
	
		
		

		
			





 Of course I'm not saying this never happens.


----------



## Marleybob217

xnor said:


> I don't know if you know dubstep, but there are tracks with pretty *low bass* (~35 Hz) and *LOTS *of it. I mean completely overpowering amounts. You can find such tracks on *ordinary CDs*.
> 
> 
> I would understand this for a "radio edit/version", but for normal CD botching the sound sounds like a cheap excuse to me.
> ...


 
 Why not? Wav only goes to 40hz? I think you are confusing mp3 to wav/flac, mp3 does cut of the inaudible frequencies if I'm not mistaken. And 35hz is very much audible.
  
 I know professional DJs that sometimes use 320kbps files in clubs. It works fine really. They have the gear to actually play those sub-bass frequencies, and it is still very much present.


----------



## stv014

Quote:


marleybob217 said:


> Why not? Wav only goes to 40hz?


 
  
 You did not read carefully enough: 
	

	
	
		
		

		
		
	


	





xnor said:


> You can find such tracks on ordinary CDs.


 
  
 Of course, CDs (and in fact anything in PCM format) can go down to 0 Hz. There is only an upper limit.


----------



## xnor

Indeed, the lower "limit" is 0 Hz = DC. In practice you won't find much content below ~20 Hz on CDs though.
  
 @Marleybob217: I was responding to UltMusicSnob's post that, if I understood it correctly, CDs are supposedly mastered for speakers that cannot reproduce sub bass properly like filtering out sub bass and boosting mid/upper bass.
  
 Besides my example above, DAPs often come with in-ears nowadays that can reproduce sub bass quite well.


----------



## UltMusicSnob

xnor said:


> Indeed, the lower "limit" is 0 Hz = DC. In practice you won't find much content below ~20 Hz on CDs though.
> 
> @Marleybob217: I was responding to UltMusicSnob's post that, if I understood it correctly, CDs are supposedly mastered for speakers that cannot reproduce sub bass properly like filtering out sub bass and boosting mid/upper bass.
> 
> Besides my example above, DAPs often come with in-ears nowadays that can reproduce sub bass quite well.


 
  
  
 Actually I was going to a purely theoretical point, that *if* a record company cared to do so, they could tailor mixes to the intended platforms. Didn't mean to imply that it's an actual practice, which actually kind of strikes me as wishful thinking. Anyway, that's a possible reason to master differently that *I* might care about.


----------



## MrMateoHead

Re-Mastering for the sake of protecting equipment seems like a good idea. On the other hand, isn't that what low- and high-pass filters are for?


----------



## Baxide

stv014 said:


> Of course, CDs (and in fact anything in PCM format) can go down to 0 Hz. There is only an upper limit.


 
 It's actually 1Hz that CD can go down to.


----------



## Baxide

xnor said:


> Indeed, the lower "limit" is 0 Hz = DC. In practice you won't find much content below ~20 Hz on CDs though.


 
 Far from it. There are loads of tunes with audio information below 20Hz.


----------



## stv014

baxide said:


> It's actually 1Hz that CD can go down to.


 
  
 There is no such restriction inherent to the format itself. It just does not make sense to record very low frequencies for the purpose of music playback.


----------



## Digitalchkn

baxide said:


> It's actually 1Hz that CD can go down to.




DC could indeed be captured showing up in the waveform plot as a constant offset from center.


----------



## Baxide

stv014 said:


> There is no such restriction inherent to the format itself. It just does not make sense to record very low frequencies for the purpose of music playback.


 
 It's not possible to record a DC signal onto a discs. What you can record is a blank space of 0 frequency. This is actually quite common on test discs and is used to measure any signal to noise ratio.


----------



## Baxide

digitalchkn said:


> DC could indeed be captured showing up in the waveform plot as a constant offset from center.


 
 Can you give me an example of a disc that has such an audio file on it? I have loads of engineering test and set up discs from CD player manufacturers. But none of them has a signal like you describe. I do have 1Hz as the lowest recorded frequency on a disc. That's as low as it goes.


----------



## stv014

baxide said:


> It's not possible to record a DC signal onto a discs. What you can record is a blank space of 0 frequency. This is actually quite common on test discs and is used to measure any signal to noise ratio.


 
  
 I can easily render DC, or a 0.1 Hz (or other very low frequency) sine wave to a 44.1 kHz stereo 16-bit PCM format sound file. Is there anything that would make it impossible to burn the data to a CD-R as an audio (Red Book) track ?


----------



## Digitalchkn

baxide said:


> Can you give me an example of a disc that has such an audio file on it? I have loads of engineering test and set up discs from CD player manufacturers. But none of them has a signal like you describe. I do have 1Hz as the lowest recorded frequency on a disc. That's as low as it goes.




Take any old waveform editor and deliberately introduce a constant offset to the waveform. Save it in 16bit/44.1K format. There you have it. The fact that you haven't seen a disc with such an offset is I would say is positive as the DC offset effectively limits the dynamic range of the recording (amongst other things.)


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## silverharbinger

Not to get too far off the subject, but this thread is an incredible resource. I know it helped me to understand that more isn't really better in audio file formats in a much better way than I was able to piece together over the years. I mean, I could never hear a difference past 16-bit in playback quality, but I just wasn't sure if it mattered. Now I know it seems we simply reached the limits of human hearing years ago, so now we're pushing formats for dogs and bats to enjoy.
  
 I wonder when we will finally get there with picture resolutions? Imagine when there's no need for another generation of TV, game system, or whatnot because you won't be able to see any difference? I guess electronics will get really boring.


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## UltMusicSnob

stv014 said:


> I can easily render DC, or a 0.1 Hz (or other very low frequency) sine wave to a 44.1 kHz stereo 16-bit PCM format sound file. Is there anything that would make it impossible to burn the data to a CD-R as an audio (Red Book) track ?


 
 An unbroken sequence of zeroes in the samples would be 0 Hz at 0 voltage. How about an unbroken sequence of, say, 0xc7f1's? That would also be 0 Hz, but at some non-zero voltage (what would the physical driver do, shift forward a millimeter and hang there???).


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## stv014

baxide said:


> Can you give me an example of a disc that has such an audio file on it?


 
  
 No claim of its commercial availability was made. There might not be such disks sold, but if that is the case, it is not because it would be technically impossible for limitations of the data format, but rather because it would not be practically useful.


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## stv014

ultmusicsnob said:


> what would the physical driver do, shift forward a millimeter and hang there???


 
  
 Yes. A high DC level could damage the driver, though, so it is best to only try it with some cheap speakers.


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## UltMusicSnob

silverharbinger said:


> Not to get too far off the subject, but this thread is an incredible resource. I know it helped me to understand that more isn't really better in audio file formats in a much better way than I was able to piece together over the years. I mean, I could never hear a difference past 16-bit in playback quality, but I just wasn't sure if it mattered. Now I know it seems we simply reached the limits of human hearing years ago, so now we're pushing formats for dogs and bats to enjoy.
> 
> I wonder when we will finally get there with picture resolutions? Imagine when there's no need for another generation of TV, game system, or whatnot because you won't be able to see any difference? I guess electronics will get really boring.


 
 In the future we'll get fitted with bionic cochlear implants that have effective response from 0.001 Hz (elephants, pigeons) to 200 kHz (bats, dolphins). Then we can start having these arguments all over again.


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## Digitalchkn

ultmusicsnob said:


> An unbroken sequence of zeroes in the samples would be 0 Hz at 0 voltage. How about an unbroken sequence of, say, 0xc7f1's? That would also be 0 Hz, but at some non-zero voltage (what would the physical driver do, shift forward a millimeter and hang there???).




And that's by definition DC. Your driver (assuming your amplification is also DC couple) would simply recenter itself. You can apply your music on top of that, but now you'd limite the range that the speaker could swing . Hasn't anyone tried to connect a AA battery to your speakers to test continuity before?


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## UltMusicSnob

stv014 said:


> Yes. A high DC level could damage the driver, though, so it is best to only try it with some cheap speakers.


 
 It's going to heat up fast, yes? The magnet will be pushing against the physical resistance of the driver material itself continuously.


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## Digitalchkn

ultmusicsnob said:


> It's going to heat up fast, yes? The magnet will be pushing against the physical resistance of the driver material itself continuously.




To that effect. You're getting into the topic of constant power vs RMS power. Let's just say for a given peak voltage or /current the RMS power of your music is going to be less than (or equal to) constant power at that peak.


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## xnor

baxide said:


> Far from it. There are loads of tunes with audio information below 20Hz.


 
  
 I didn't say there are no CDs with information below 20 Hz, just that you won't find much content (maybe should have written _energy_) below 20 Hz.
  
   
 Quote:


digitalchkn said:


> And that's by definition DC. Your driver (assuming your amplification is also DC couple) would simply recenter itself. You can apply your music on top of that, but now you'd limite the range that the speaker could swing . Hasn't anyone tried to connect a AA battery to your speakers to test continuity before?


 
 A DC coupled amp would potentially amplify the DC. Maybe you meant AC coupled? Such an amp (like the O2) doesn't output DC (well in reality every amp outputs a tiny bit of DC...).
 With such an amp your headphones are safe even if the source suddenly decides to output DC and the amp has a high gain factor.


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## silverharbinger

ultmusicsnob said:


> In the future we'll get fitted with bionic cochlear implants that have effective response from 0.001 Hz (elephants, pigeons) to 200 kHz (bats, dolphins). Then we can start having these arguments all over again.


 
  
 I had that thought too, and for people with tinnitus and all the other nasty conditions it would be amazing, but I hope it doesn't happen in my lifetime. Having to constantly update my gear *AND* my eyes, ears, tongue, etc.? Plus, just imagine all the aftermarket ear mods that will be offered on these forums? OH GAWD!


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## xnor

You better hope those implants have configurable frequency response or else you might go crazy...


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## silverharbinger

xnor said:


> You better hope those implants have configurable frequency response or else you might go crazy...


 
  
 There's an app for that.


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## Digitalchkn

xnor said:


> I didn't say there are no CDs with information below 20 Hz, just that you won't find much content (maybe should have written _energy_) below 20 Hz.
> 
> A DC coupled amp would potentially amplify the DC. Maybe you meant AC coupled? Such an amp (like the O2) doesn't output DC (well in reality every amp outputs a tiny bit of DC...).
> With such an amp your headphones are safe even if the source suddenly decides to output DC and the amp has a high gain factor.




I meant DC coupled, like headphone amps you mentioned. It's really same argument with DC coupled headphones amps damaging your headphones.


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## Headzone

When I compared 24bit and 16bit output from my pc (At 44khz) I thought for a second 24bit sounded better. My music files are all 16bit I guess. Does that mean I'm imagining things?


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## stv014

In exactly what way did they sound better ? Did you use software volume control or any other processing ?


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## Headzone

Dunno, just more clear and bit more dynamic sound I guess? I didn't do any further comparison tho,  just felt 24bit was better before I started researching on the subject. 
  
 I think the sound is coming through asio drivers, no processing or digital volume altered.


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## bigshot

headzone said:


> Dunno, just more clear and bit more dynamic sound I guess? I didn't do any further comparison tho,  just felt 24bit was better before I started researching on the subject.
> 
> I think the sound is coming through asio drivers, no processing or digital volume altered.


 
  
 Here we do level matched blind A/B tests to determine that.


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## xdog

I have to share something about 24bits; especially vinyl rips with 24bits
 Since I haven't seen a vinyl for a while (and I havent mesured it), I'll assume that the height of the vinyl row is something like 1-2mm (even less),
 that is 10^(-3)m; but the distance between atoms is something between 0.1-0,2nm which is 10^(-10) m;
 If we take (for simplicity) some organized material, like diamond, or even better graphits, the in the row we can represent
 maximum 10^7 ~= 2^23 levels (and that is a real physical obstacle);
 And if I think of 1 layer of atoms, which is insane, than I come to conclusion that at least 4 layers is maximum accuracy
 (due to random atomic movement, that everytime you touch your vinyl you nano-scratch it or pressure it, since it is not diamond),
 which means real bit-level like 2^20-21.
 And even more if you imagine the needle of you gramophone (wich is more likely NOT a atomic force microscope) and the recording process,
 than you probably get the accuracy counted in hundreths of layers of atoms (16 bit or more).
 The same applies to microphones and speakers, since 24bits is atomic distance precision.


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## skamp

You're completely ignoring the inherently high noise floor of vinyl records.


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## nick_charles

xdog said:


> I have to share something about 24bits; especially vinyl rips with 24bits
> Since I haven't seen a vinyl for a while (and I havent mesured it), I'll assume that the height of the vinyl row is something like 1-2mm (even less),
> that is 10^(-3)m; but the distance between atoms is something between 0.1-0,2nm which is 10^(-10) m;
> If we take (for simplicity) some organized material, like diamond, or even better graphits, the in the row we can represent
> ...


 
  
 here is a good link for the molecular physics involved http://www.st-andrews.ac.uk/~www_pa/Scots_Guide/iandm/part12/page2.html


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## proton007

xdog said:


> I have to share something about 24bits; especially vinyl rips with 24bits
> Since I haven't seen a vinyl for a while (and I havent mesured it), I'll assume that the height of the vinyl row is something like 1-2mm (even less),
> 
> that is 10^(-3)m; but the distance between atoms is something between 0.1-0,2nm which is 10^(-10) m;
> ...




That's interesting. In order to physically support 24 bits the vinyl grooves will have to be deeper.
This also implies that the typical 24bit vinyl rips are 16bit at best.


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## xnor

Vinyl ... 12 - 13 bits are enough.


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## spagetka

I love this thread. From time to time come back and read nice posts and believe me or not I am always smiling.
  
 Guys with super technical skills and arguments one by one - what about a nice ABX test 320vbr mp3 vs CD or even 192vbr vs CD? Remember that result 9 or better out of 10... 
  
 Edit:
 Helloooo? Nobody? I live in Prague (CZ) and Bratislava (SK) so no problem to visit Austria as it is not far away...
  
 Edit:
 Where are you?


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## reginalb

Well, this thread was an awesome read. I only made it through the first 14 pages. By then it was: 
  
 "[Scientific reasons that there is no reason to playback anything above 16-bits (although 24-bits is definitely useful during the production process)]" - Gregorio
  
 "YOU'RE AN IDIOT STOP WITH THE FACTS." - People that, like Steven Colbert, don't like books ("They're all facts, no heart")
  
 Seriously though. This thread was an _outstanding _read. I own a decent amount of 24-bit audio. I did some ABX testing after converting those files to 16/44.1, and I could hear the difference. I also learned why (dBpoweramp probably isn't the best tool) 
	

	
	
		
		

		
			




  
 I will probably still buy 96 (no more 192) or 16/44.1 from Acoustic Sounds. I like that they disclose who did the mastering, and how it was done. And that they're probably not in to the whole "loudness" thing. I think the music I get from them sounds great, but I have long bought hi-res (for some genres of music, like jazz) not because I thought the "resolution" mattered, but I hoped that the companies mastering them were taking better care of the files. 
  
 Case in point, of the many copies of Ella and Louis that I own, the one from Quality Record Pressings on 200-gram vinyl sounds better than anything else that I have heard. But I firmly believe that it's down to the master (they do all their own). That I get to play it on my sweet turntable is just a bonus 
  
 Alright, I have gone on a bit of an aside, but I do wish I Would have ventured in to the Sound Science section of the forum sooner in my first couple of years here. 
  
 Thanks Gregorio, very good read!


----------



## k3oxkjo

skamp said:


> You're completely ignoring the inherently high noise floor of vinyl records.


 
  
 If you're in digital land, that's "dither", LOL.


----------



## xnor

Yeah but the noise only becomes comparable if you're down to 13 or 12 bits...


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## Digitalchkn

xnor said:


> Yeah but the noise only becomes comparable if you're down to 13 or 12 bits...


 
 Noise characteristics of vinyl noise or tape noise are not quite the same as those of digital noise (either due to quantization, dithering, jitter etc.).
  
 But another way to look at it, if you consider including vinyl noise as part of the signal you want to capture digitally then I can argue that you need higher resolution to capture everything full detail from vinyl.


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## bigshot

The dynamic range of analogue audio fits within redbook with room to spare. And clicks, even super fast clicks on 78s, operate on a scale much larger than digital sampling. High resolution audio is totally wasted on analogue captures. All it makes is bigger files that are slower to process.


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## xnor

13 bits would already be pretty much best case vinyl during a quiet passage, so no, you wouldn't need more.
 Sure, with 16 bits you're going to be on the safe side.


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## Digitalchkn

xnor said:


> 13 bits would already be pretty much best case vinyl during a quiet passage, so no, you wouldn't need more.
> Sure, with 16 bits you're going to be on the safe side.


 
  
 Even as you throw more bits at the problem you approach the original signal+noise but never quite captured it exactly. So whether 16 bits is sufficient or not is very much subjective.


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## bigshot

I've done digital restoration of very noisy records. 16 bits is more than enough. I tried 24 bits, but it just slowed down my noise reduction filters for no audible improvement.


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## Digitalchkn

bigshot said:


> I've done digital restoration of very noisy records. 16 bits is more than enough. I tried 24 bits, but it just slowed down my noise reduction filters for no audible improvement.


 
 De-noising filters generally operate based on assumptions of known statistics of the noise. Often they don't perfectly remove just the noise. There is likelihood that some minute amount of what we consider a sound signal is removed in the process as well. The question of this balance is, again, subjective to the listener.


----------



## bigshot

That's fine in theory, but in practice, digital noise filters are extremely useful. Just another tool that can be used well or poorly depending on the person using it.
  
 Some noise filters use aspects that are beyond the range of the music itself to determine what is noise and what is signal... for instance impulse info or super audible frequency information.


----------



## aklein55

Gregorio - thank you for this incredibly compelling and helpful discussion.  I'm not a sound engineer nor do I have any technical expertise - with that, two questions:
  
 1.  Why are companies migrating to high resolution audio (see Sony, for example) when the real problem in consumer audio reproduction is the mass sale/distribution of compressed audio.  I own a bunch of CDs - but it is a pain to have to go buy CDs and then rip those into lossless audio files.  The alternative is to go with HD Tracks - but no one is selling CD quality lossless for consumer sales outside of physical CDs in any volume.  Seems crazy to me that the non-compression market distributors aren't saying - hey, you really only need a lossless version of the CD.  Just very frustrating.  Also, I take your point that higher resolution doesn't actually "resolve" anything that a listener can actually hear - and that the quality of the sound recording is the real determinant factor for playback.  With that - why not market the quality of the sound recording in a lossless format at 16 bit - and not fool consumers into buying "high resolution" snake oil (which also consumes a lot more memory and processing power).
  
 2.   Re 16 bit v 24 bit, is there something that the music industry can do to improve audio reproduction now that we all own 3+ terabit tank hard-drives and enough processing power to resolve higher resolution files.  In other words, what IS a good idea in terms of the development of audio reproduction (now that I know that added bit depth is useless).
  
 Thanks - this is a great discussion.
  
 Adam


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## xnor

1) People have been told that higher resolution sounds better due to the format, so they ask for that format. Companies are just serving what people are asking for.
 No company wants to be stuck at some "good enough" point. For example there is not much Ultra HD (4k) material out yet and companies are already working on 8k. They always try to sell you more expensive stuff which is supposedly better.
  
 2) "Improving" reproduction will not change the limits of human hearing.
 Sampling rate: An infamous analogy is trying to blind yourself with an IR remote. Do you need monitors that reproduce colors outside the visible band?
 Bit depth: Do you need someone whispering a few meters away followed by a jackhammer next to your ear?
  
 The most important improvements are still happening in headphones themselves. With research being done by Harman, NRC .. there's finally an attempt of making headphones that have a more accurate frequency response, lower distortion.
 Oh and of course audio processing technologies ranging from simple crossfeeds to more complex room simulation with head tracking and your personal HRTFs.


----------



## aklein55

Thanks Xnor - so what about up sampling, over sampling and the other "fancy" DAC processing tricks that companies advertise - is that all snake oil as well.  Given Moore's Law - did audiophile processing power hit the wall - with NO appreciable improvement with continuing advances in computer processing power?


----------



## xnor

Some techniques are needed to make high quality reconstruction possible and less expensive. For example oversampling in a ΣΔ converter is used so that you don't need expensive high-precision analog anti-aliasing filters.

  
  
 Sample rate conversion, no matter if done in software or hardware, doesn't hurt if done right. It can only add redundant information but there may be some gains in performance in certain configurations. Whether those differences are even close to being audible is another story.
  
  
 I don't think companies will stop at 96 kHz and 24 bit. Not long ago HDMI 2.0 was announced with:


> Up to *32 audio channels* for a multi-dimensional immersive audio experience
> Up to *1536kHz* audio sample frequency for the highest audio fidelity


 
 That sample rate may "only" be 32 channels with 48 kHz each (32*48=1536) but even so .. 32 is quite a big number of channels. Maybe it also means that audiophiles can resample their good old 2.0 stereo files to 768 kHz for that extra placebo .. ehh audio fidelity?


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## aklein55

Um - what's at 48 kHz or 96 kHz - even my dog couldn't hear that.  Audio for mice?


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## nick_charles

aklein55 said:


> Um - what's at 48 kHz or 96 kHz - even my dog couldn't hear that.  Audio for mice?


 
  
 A sampling rate of 48khz allows you to accurately capture signals up to about 24Khz (fs/2)


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## aklein55

Right fs/2.  After that - it's cats and mice.
  
 The behavioral audiograms of two cats were determined in order to establish the upper and lower hearing limits for the cat. The hearing range of the cat for sounds of 70 dB SPL extends from 48 Hz to 85 kHz, giving it one of the broadest hearing ranges among mammals. Analysis suggests that cats evolved extended high-frequency hearing without sacrifice of low-frequency hearing.


----------



## xnor

Well, turning Fs/2 around: you need *more than* 2 samples per cycle to reconstruct the signal.
 44.1/22.05 = 2 samples per cycle for a 22.05 kHz tone sampled at 44.1 kHz
 In practice, most DACs roll-off between about 20 kHz to 21 kHz with a 44.1 kHz sampling rate, so around 2.2 to 2.1 samples per cycle.
  
  
 Hearing high frequencies really is very problematic for humans. First there are less and less hair cells that are tuned for high frequencies. As a result, the hearing limit is rising to a point where you need to play the tone at extremely dangerous levels to just barely hear it:

  
 That's why 20 kHz is the generally accepted upper limit of hearing.
  
 Secondly, that limit degrades with age and there are many people who cannot even hear a clean 18 kHz tone anymore. Elders usually have troubles hearing stuff above ~16 kHz.
  
 Thirdly, if you look at the spectrum of real music you will see a downward slope. The higher the frequency, the less energy. I don't know of any natural musical instrument that produces mainly treble with lots of energy.
 Even if you could hear a _clean _20 kHz tone at 100 dB SPL you would have to turn up the music so loud, that the spectrum of the music would be pushed to far above 150 dB SPL at 100 Hz.  (Having heard a jet engine at ~150 m away I cannot recommend anyone hearing it at a distance of only 30 m, which according to Wikipedia would be roughly 150 dB SPL.)


----------



## gregorio

Interesting to see this thread is still alive and well 4 years after I started it! Some of the questions and responses are also interesting, as is the fact that some posters can't seem to get their head's around the concepts in my OP. Generally this appears to stem from the fact that they have an entrenched concept which they are unwilling or unable to to question, namely that more resolution = more detail. One can't entirely blame them for this entrenched position as it has been created and reinforced by the marketing of all those companies who sell higher bit depth files or equipment. The very term "Hi Rez" itself is a purely marketing term rather than an accurate description because if we are talking about resolution in terms of detail (of the sound which comes out of your DAC), 24bit has no more resolution than 16bit or indeed than 1bit! I saw one truly bizarre set of posts in this thread from someone who thought it was simple common sense both in theory and in practice that 24bit had more detail and sounds better than 16bit and that anyone suggesting otherwise must be essentially insane. This poster then held up SACD as the highest audio quality but of course SACD uses just 1bit and therefore, according to his common sense argument, SACD should sound many times inferior to the humble 16bit CD.
  
 Some of the points raised in this thread (and others on Head-Fi) could do in my opinion with a little more explanation, so here goes:
  
*Resolution*
  
 To help those still struggling with the concept that resolution does not equal detail, try turning your logic around and think of it this way: Exchange the word resolution with the words "less error". More bits = more accuracy and of course more accuracy = less error, hence higher resolution (more bits) therefore means less error. The next leap in understanding this question of resolution is understanding the fact that error in digital audio manifests as noise when it is converted back into an analogue signal. More resolution therefore results in less noise compared to lower resolution, the detail (fidelity) is always there and always the same (at any bit depth) but the more bits you use the less noise will accompany that detail. By the time we get to 16bit, the noise (due to error) is already many times below the noise which is already present in the recording due to other unrelated music production factors. Using noise shaped dither further reduces the audibility of that noise to well below the noise produced by even the very finest DAC, amp and headphone or speaker system. In other words you cannot hear this dither noise! So, if you already cannot hear the errors (noise) introduced by 16bit what is the point of increasing the resolution (to 24bit) and reducing the noise even further. In other words, what do you possibly imagine could be gained by taking inaudible noise and making it quieter? If this is true, why does 24bit even exist, what point is there? This brings me on to recording:
  
*Recording*
  
 When recording in 16bit, we want to create the cleanest recording possible and that means making sure that the noise caused by our 16bit errors are below the noise of our microphones, mic pre-amps and the noise floor of our recording studio. This means setting our microphone pre-amps so that the loudest parts of our recording are somewhere near the maximum (0dBFS) level of our system. The problem with this is that we don't know what the loudest part of our recording is actually going to be until after we've recorded it. Sure, we can (and do) do a sound check but even the very finest artists cannot reproduce a performance exactly, so a sound check is only a very rough guide. Performers can create peaks 12dB higher or even more during recording compared to the sound check. So our problem is how to record loud enough to ensure our digital noise (errors) does not encroach on our recording but not too loud so that the loudest parts of the performance exceeds 0dBFS. In some recording situations this can be a quite small window of opportunity to hit. Recording in 24bit though ensures that our digital error noise is so far below the noise of our recording equipment and environment that we don't need to think about it. With 24bit we can afford to set our mic pre-amps so that the loudest part of our recorded signal is nowhere near 0dB, even -20dB is fine in 24bit, so it doesn't really matter how loud the performer performs, we're not going to ruin that brilliant and unrepeatable performance because we've hit the limit (0dBFS). We don't use 24bit for recording because it provides more detail, better quality or higher fidelity, a 24bit recording will have the same fidelity as a 16bit recording made within that window of opportunity, it just makes that window of opportunity far easier to hit! In other words, the advantages of 24bit for recording are all about workflow and nothing to do with quality or fidelity. Of course, when we mix/master and create the distribution mix, we already have the recorded material, already know where it is going to peak and already know what that peak value is, so we don't need that spare dynamic range, can go right up to -0.1dBFS without ruining our mix and are going to be so far away from the 16bit digital noise (errors) that it's guaranteed to be many times below the threshold of audibility. This brings up the question of mixing:
  
*Mixing*
  
 When we mix music (or a film or TV program) we could be dealing with anything from a few channels of sound up to over 1,000 channels. On each of these channels we may have anywhere from zero processors, up to as many as 8-10, (EQ, compression, reverb and a wealth of others). Each of these processors process the audio, which in digital audio means runs a series of mathematical algorithms (calculation processes) and each of these algorithms is likely to introduce an error in the least significant bit (LSB). The LSB can only hold a 1 or a 0 but our algorithm (calculation) could easily result in say 0.6, so we set the LSB to "1" but we've introduced an error (noise). You're not going to hear this error in 16bit and obviously you're not going to hear it in 24bit but what happens when we've got say 80 channels of sound, each with say 2 processors and therefore a total of say 200 algorithms, the results of which are all being summed together? You will certainly hear the result of these combined errors at 16bit and even probably at 24bit. This problem is overcome by operating the processors and mixing at far greater bit depths, the errors in the LSB are now are so minuscule that even summing thousands of them together does not introduce anything remotely audible. It's common for many years for processors to be operating at 48bit, with mixing at 56bit and becoming even more common today to run everything (processors and mixing) at 64bit float. Once the mix has been made it can be brought back down to 16bit which is more than enough for every playback situation. I hear some audiophiles screaming: "I want the recording at it's original resolution". Well, you can have it! We record at 24bit (for reasons explained in the paragraph above) and we mix at various different bit depths, which generally we cannot print. The 56bit and 64bit float mixing environments common today are just for internal processing, we cannot actually write (record) files at these bit depths because they are useless, nothing can play them back and even if something could play them back you wouldn't be able to hear what was in the least significant 50bits or so (of a 64bit file) anyway. So when you see companies advertising "24/96 or 24/192 hi rez as it was created by the studio", that's a double lie! 24/96 (or 24/192) is not hi rez and is not as it was created by the studio. Which brings us on to mastering:
  
*Mastering*
  
 Mastering is the process of taking a mix created in a recording/production studio and processing it so that it sounds good on the target audience's playback equipment, rather than sounding good only in the recording/production studio where it was made. This raises a number of important questions such as;what genre is the music, who is the target audience, what is their playback equipment and related to this, what format are we distributing? This obviously requires making assumptions and generalisations but if for example we are distributing jazz or classical on SACD we are most probably looking at an older target audience, who most probably take their music listening seriously (otherwise they wouldn't have bought an SACD player) and who are therefore the most likely demographic to have high or very high quality playback system/environment. So, we are far more likely to record, mix and master to a high standard and with a wide dynamic range. Instead of burning SACDs from this high quality master we could just convert it to 16/44.1, put it on a CD and this CD will sound absolutely identical to the SACD. For the record company there are two problems with this though: Firstly, this wide dynamic range, high quality recording might not be suitable for many playback systems/environments and secondly, how do you justify a significantly higher price for a "hi-rez" SACD which contains a recording indistinguishable from the much cheaper CD version? Of course there's an easy solution, you make a different 16/44.1 version which is distinguishable from the "hi rez" version! This brings us on to:
  
*Dynamic Range*
  
 For practical purposes, dynamic range can be defined as difference between the highest energy in a signal (recording) and the lowest. As explained earlier in this thread, 16bit is capable of containing more dynamic range than you can safely listen to and even the best and most dynamic of SACDs have a dynamic range of no more than about 60dB and most recordings have a dynamic range of less than 40dB. To put this into perspective, 16bit is capable of roughly 1,000 times more dynamic range than even the most dynamic of SACDs. I can't understand this attitude from some audiophiles of wanting even more dynamic range than 16bit provide, enough dynamic range in fact to kill them if it were actually possible to use 24bits of dynamic range. Not only is this desire for 24bits (144dB) of dynamic range literally suicidal, it makes no sense in many cases to increase dynamic range of even some of the crushed music which many audiophile abhor. If you are listening to a recording in a quiet environment with very low environmental noise then yes, a decent dynamic range is a good thing and will allow the recording to sound breath and sound more alive but listen to that same recording say in a car while you're driving along the interstate and the high ambient noise will mean that you won't be able to hear half of recording without turning the volume up and nearly deafening yourself when a loud piece of the music comes along. I saw a thread on Head-Fi earlier where someone seemed desperate to get 24/192 playback from a galaxy smartphone. A quick look at the specs shows that at the headphone outputs this phone has a dynamic range of 92dB, well above the dynamic range of any commercial recording but well below the dynamic range possible with 16bit, why then is he wanting 24/192 playback? Even if his phone can handle 24bit format files it can't actually play more than about 15bits of them and for all intents and purposes ignores the other 9bits.
  
 Coming back to mastering, why is it that many audiophiles seem to spend so much time, effort and money obsessing about aspects of their equipment which cannot possibly be heard but relatively little time and effort understanding and appreciating good mastering? I've heard the response to this question, which is "well the mastering is fixed on the recording and there's nothing we can do about it but we can do something about the equipment we use". Sorry, but this is horsesh*t, for two reasons: 1. If, as some fanatical audiophiles contend, power cables, digital inter-connects and ridiculously expensive speaker cables do actually make an audible difference, then every choice they make changes the mastering! For example, if an extremely expensive cable makes their system sound brighter, the mastering engineer almost certainly used relatively cheap copper cables and made the master exactly as bright as he/she intended, the expensive cable is changing the master! and 2. If these audiophiles spent more time on something which is easily audible instead of on what is patently not audible, they would soon learn what good mastering actually is and if they only purchased well mastered music, the record companies would soon take note and make sure their mixes/masters were of a higher standard to cater for this market! Let's not forget that mastering engineers are human beings, you can have an album mastered for $40 a song through the internet from a young newbie with little knowledge, experience or facilities or you can have an album mastered in one of the top mastering facilities by one of the world's great mastering engineers for $20k. Which album would you think is likely to sound better and which would you choose to buy if the price were the same or nearly the same?
  
*In Summary*
  
 Can you hear a difference between 16bit recordings and 24bit ones? Yes, most definitely you can, I certainly have! There is however only 3 possible explanations 1. You are imagining the difference or 2. Some serious mistake has been made during mastering or the most likely 3. You are actually listening to different recordings/masters. 16bit, 24bit, 1bit (SACD) are just containers, what you put in those containers defines the quality of what you are hearing, not any inherent quality of the container itself. It's like trying a drink from a square bottle, liking it more than a previous drink you had from a round bottle and therefore deciding that you'll only drink from square bottles in future. In reality of course it's the drink which is in the bottle which makes you like the taste or not, not the shape of the bottle it's in.
  
*One final thought*: We've already covered that 16/44.1 is more than will ever be needed but going the other way, is there any potential problems with listening to the same recording at 24/96 or 24/192 or even 32/384? As far as bit depth is concerned, the answer is "no" there's nothing to be gained from higher bit depths but there's also nothing to loose, except storage space. This isn't necessarily true of the higher sample rates however! Most amps and speakers are not designed to reproduce any frequencies above about 20kHz, feeding them any significant amounts of frequency outside this range can cause them to create inter-modulation distortion (IMD), which is spurious tones or sounds within the range of human hearing. It's possible that some audiophiles might actually like this distortion or feel it is in some way "better" but for most sane people unexpected and unpredictable distortion is something to be avoided! Furthermore, once we get to 192k sampling rates and beyond it is impossible to correctly filter them according to the demands of the Nyquist Theorem. While it's extremely unlikely that this will result in any audible problems, it is in theory at least, lower digital fidelity. So don't get sucked into the marketing hype that 24/192k (or even worse 24/284) is somehow higher quality or higher fidelity than say 96k or 44.1k!
  
 G


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## xnor

Well, bit depth/resolution with PCM can be understood as detail since higher bit depth means less quantization noise and lower noise floor which potentially masked low-level details before, BUT people don't understand that the noise floor is quite low with 16 bit to begin with! And yeah, they seem to think that higher resolution will also improve loud parts of the signal.


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## spaark

Great post, gregoria. There are a couple of things I'd like to add about resolution, though. When you quantise an analogue signal, the noise introduced can actually be non-linear. This distortion _can be audible_. To prevent this, we can randomise the quantisation errors so that the noise is ergo random and spread uniformly across the frequency spectrum. This way, the noise introduced (called white noise) is linear. This process is known as _dithering_. Going even further, we can can distribute this noise at varying amounts for different frequencies. This exploits the fact that we're more sensitive to some frequencies than others, so the frequencies that we're more sensitive to should have less noise than those that we're less sensitive to. The total amount of noise is still the same; only the distribution is different. This process is known as _noise shaping_. At this point, you can expect the (white) noise to be well below the threshold of hearing at sane volumes.


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## xnor

He mentioned noise shaping and dither, but that part could indeed be a bit more detailed.
  
  


> When you quantise an analogue signal, the noise introduced can actually be non-linear.


 
 This sounds confusing.
  
 The problem is correlation between the input signal and quantization noise resulting in non-linear distortion. Dither is used to de-correlate the quantization noise from the input signal, resulting in low-level white noise instead of low-level distortion products.


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## Don Hills

gregorio said:


> ... Furthermore, once we get to 192k sampling rates and beyond it is impossible to correctly filter them according to the demands of the Nyquist Theorem. While it's extremely unlikely that this will result in any audible problems, it is in theory at least, lower digital fidelity. ...


 
  
 I think it was a very good post. There is one part I am struggling with, quoted above. How is it impossible to filter a very high sampling rate according to the demands of the Nyquist Theorem? 
  
 In any case, we don't go to 192 KHz in order to try and record signals up to 96 KHz. We do so in order that we can use a simpler, lower order, low-pass anti-aliasing filter.


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## spaark

xnor said:


> He mentioned noise shaping and dither, but that part could indeed be a bit more detailed.


 
 Yes, and also that quantisation on its own introduces non-linear noise. The point is there's more to it than "the noise in 16-bit audio is inaudible".
  
 Quote:


xnor said:


> This sounds confusing.
> 
> The problem is correlation between the input signal and quantization noise resulting in non-linear distortion. Dither is used to de-correlate the quantization noise from the input signal, resulting in low-level white noise instead of low-level distortion products.



 What's confusing about it? We can indeed say that the quantisation noise is non-linear.


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## xnor

spaark said:


> What's confusing about it? We can indeed say that the quantisation noise is non-linear.


 
  
 What is non-linear noise? What's the opposite?


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## Digitalchkn

xnor said:


> What is non-linear noise? What's the opposite?


 
 People sometimes refer white noise as linear due to it's flat power spectral density - a line.  Noise is neither "linear" nor "non-linear". It is characterized by it's statistics. Otherwise, it's not strictly speaking noise because then we would simply could remove it.


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## skamp

gregorio said:


> even the best and most dynamic of SACD have a dynamic range of no more than about 60dB




Hmm, that doesn't sound right (no pun intended). Surely a noise floor at -60dB would be audible. Or am I misunderstanding?






> The SACD format is capable of delivering a dynamic range of 120 dB from 20 Hz to 20 kHz […] With appropriate low-pass filtering, a frequency response of 20 kHz can be achieved along with a dynamic range of nearly 120 dB, which is about the same dynamic range as PCM audio with a resolution of 20 bits.




http://en.wikipedia.org/wiki/Direct_Stream_Digital


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## gregorio

xnor said:


> He mentioned noise shaping and dither, but that part could indeed be a bit more detailed.


 
  
 When I started this thread, the idea was to make as simple an explanation as possible, so it could be understood by Head-Fi'ers who may not have much interest in the deep technical detail. My last post followed the same vein. There's a broad spectrum of people on this site and it's a tough ask to write an article which covers all of them. In my last post I tried to balance as little detail as possible with enough detail to try and avoid rendering what I wrote too inaccurate.
  
 Quote:


spaark said:


> When you quantise an analogue signal, the noise introduced can actually be non-linear. This distortion _can be audible_. To prevent this, we can randomise the quantisation errors so that the noise is ergo random and spread uniformly across the frequency spectrum. This way, the noise introduced (called white noise) is linear. This process is known as _dithering_. Going even further, we can can distribute this noise at varying amounts for different frequencies. This exploits the fact that we're more sensitive to some frequencies than others, so the frequencies that we're more sensitive to should have less noise than those that we're less sensitive to. The total amount of noise is still the same; only the distribution is different.


 
  
 I'm not sure I would use the term linear or non-linear in this context, correlated or de-correlated would perhaps be more accurate and less confusing. The act of dithering is the act of de-correlating the quantisation error, which is essentially just a fancy and more precise way of saying the errors are turned into random noise. If we're going into the finer detail, I'm not sure the last sentence I've quoted of yours is entirely accurate; generally noise shaped dither would actually slightly increase the total amount of noise but of course significantly decrease the amount of audible noise. However, modern mastering dither processors allow the mastering engineer to select the amount of dither applied as well as (and independently from) the amount of re-distribution (noise shaping) of that noise, so the answer to this point is not entirely clear cut. Another factor to consider is where we can (or should) apply noise shaped dither. Where and how dither is applied in ADCs and processors can get pretty complex and difficult to understand, not least because the designers and the companies they work for tend not to want to divulge this information. When mixing/mastering though we generally do not want to noise shape the dither of every channel of sound in the mix, because if we start summing many channels of sound together all with that dither noise concentrated in the same frequency band we may introduce unwanted audible artefacts in broadcast limiters and even when converting to lossy codecs. Generally, when dither is required during mixing only standard TDPF (non-noise shaped) dither is used and as it's spread evenly across the spectrum there is no build up or concentration of noise in any one frequency band, just a 3dB increase in noise for every dither summed channel, we then we apply a noise shaped dither as the very final mastering process. In most mixing environments today though the bit depths are so high that you don't need to apply dither while mixing because even the correlated noise from truncating is still massively below audibility. It's generally only when reducing to 16bit where correlated noise errors are of any potential concern. It should be noted that even when truncating to 16bit there's relatively little evidence that this correlated noise (truncation error) can be heard at normal listening levels. So the application of noise shaped dither is to some extent a "playing it safe, just in case" approach.


don hills said:


> There is one part I am struggling with, quoted above. How is it impossible to filter a very high sampling rate according to the demands of the Nyquist Theorem? In any case, we don't go to 192 KHz in order to try and record signals up to 96 KHz. We do so in order that we can use a simpler, lower order, low-pass anti-aliasing filter.


 
  
 Nyquist demands that the signal is band limited. This means applying anti-alias and anti-imaging filters to remove the error signal above the Nyquist Point (fs/2). In the case of 16/44.1 it's relatively trivial to accomplish 120dB or more of attenuation in the stop band (the range of frequencies above the Nyquist Point) and therefore reduce anti-aliasing to below the digital noise floor. But with 24/192 we have a great deal more processing to accomplish but no additional time in which to accomplish it. At these very high sample rates and bit depths we start hitting the limits of the laws of physics in how fast we can perform the calculations required to implement a filter which reduces anti-aliasing to below the digital noise floor. The only way this is likely to change is with a new paradigm in processing, for example quantum computing could in theory solve the problem! All professional ADCs initially sample at incredibly high rates (many megahertz) but they do so with a greatly reduced bit depth, 5 bits or so generally. In other words, you either have more bandwidth OR more accuracy but not both! This is borne out in tests and in manufacturers' specifications; generally at 24/192 anti-alias attenuation is only accomplished down to around -80dB which results in distortion across the entire frequency spectrum, including the audible band! It's unlikely (but not impossible) that this failure to achieve sufficient anti-aliasing to fully satisfy the Nyquist Theorem is going to be audible but nevertheless, this additional distortion does mean that in theory at least 24/192 is lower fidelity than 24/96. For the same reason, 24/384 and 32/384 performs even worse than 24/192 and is even lower fidelity. Given the choice, no knowledgeable music recording engineer would ever record at anything higher than 24/96 but they are sometimes not given the choice by the record companies employing them. Unfortunately the audiophile world is driven by marketing more than by fidelity!
  
 G


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## MrTechAgent

Always came across , never read (I DON'T KNOW WHY ?)
 Great write up


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## gregorio

skamp said:


> Hmm, that doesn't sound right (no pun intended). Surely a noise floor at -60dB would be audible. Or am I misunderstanding?


 
  
 No, you're not misunderstanding, that's exactly the point when mixing and mastering, or at least one of the points! While you don't want listeners to specifically hear the noise floor, you do want them to hear the details of the recording all the way down to the noise floor. This means that the noise floor of the recording is hopefully roughly the same as the noise floor of the audiences' listening environment. Given that the average home listening environment is very roughly about 50dB, a recording which has a dynamic range of 60dB would mean the loudest peaks of the music would be at 110dB, which is extremely loud and would be uncomfortable for most people. In reality most people would play the music back quieter and simply not hear any of the details in the recording near the recording's noise floor because they would be considerably quieter than the noise floor of their environment. That's why very few recordings have a dynamic range as wide as 60dB and a 40dB or less dynamic range is so is much more common. Listening to music in something like a moving car, the ambient noise floor is way higher than an average home listening environment and therefore reducing the dynamic range of the recording even more is a good thing.
  
 You've quoted the wiki article but you are confusing the container with it's contents! Yes, SACD like CD is capable of 120dB of dynamic range but for the reasons explained above, no one has or ever would release a recording with a dynamic range of 120dB, as even 60dB dynamic range is too much in the majority of cases!
  
 G


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## skamp

gregorio said:


> You've quoted the wiki article but you are confusing the container with it's contents!




Well, you made it sound like SACD was technically inferior by quite a large margin, which it isn't. The dynamic range of _recordings_ has nothing to do with the medium, be it SACD, CD, DVD-Audio or BluRay, since all of them are technically able to accomodate the largest dynamic range present in recordings.


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## skamp

It's really the way you put it that is confusing. Let me quote you again:



gregorio said:


> even the best and most dynamic of SACD have a dynamic range of no more than about 60dB




That's still not technically true. The dynamic range is the difference between the loudest signal on the recording (often 0dBFS) and the actual noise floor of the recording, i.e. the noise that's present even when there's no signal (silence), which can easily be the noise floor of the medium itself (-96dB for CD-DA, -120dB for SACD) when the music is generated electronically, for instance.

When the end of a track is fading into silence, I expect that it doesn't abruptly stop at -60dB: that would be audible and would completely defeat the purpose of the effect.


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## skamp

gregorio said:


> you do want them to hear the details of the recording all the way down to the noise floor




I really, really don't want to hear the noise floor. "Quiet" is way above 16 bit quantization noise, let alone 20 or 24 bit. It sounds like you're the one who's confused.


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## gregorio

skamp said:


> That's still not technically true. The dynamic range is the difference between the loudest signal on the recording (often 0dBFS) and the actual noise floor of the recording, i.e. the noise that's present even when there's no signal (silence), which can easily be the noise floor of the medium itself (-96dB for CD-DA, -120dB for SACD) when the music is generated electronically, for instance.


 
  
 If we are talking about what is technically true then CD has a dynamic range of about 98dB and SACD has a range of about 6dB! However, both use noise shaped dither to achieve a perceptual dynamic range of 120dB. It maybe possible to achieve a recording noise floor of 120dB with purely electronically generated music but it wouldn't be easy, mainly due to the nature of generating electronic music and the processing which is usually applied to it. There would be no point in trying to achieve this though as at any vaguely sensible listening level a recording noise floor at -120dBFS would be many times below the noise floor of the listening environment.
  
 Quote:


skamp said:


> I really, really don't want to hear the noise floor. "Quiet" is way above 16 bit quantization noise, let alone 20 or 24 bit.


 
  
 Of course you don't but presumably you also don't want to miss any of the details in the music because it's below the noise floor of your listening environment? This is why recordings never attempt to get anywhere near the dynamic range limitations of the recording format, as I explained earlier. Sorry if you're still confused, it seems to have been caused by a typo, I intended to write "even the best and most dynamic of SACD*s* have a dynamic range of no more than about 60dB".
  
 G


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## stv014

gregorio said:


> Sorry if you're still confused, it seems to have been caused by a typo, I intended to write "even the best and most dynamic of SACD*s* have a dynamic range of no more than about 60dB".


 
  
 That is the difference between the loudest and quietest part of the music, but noise can be heard even if it has a lower overall RMS level than the signal (but not by an extreme amount). Not least because it has a different spectral distribution, and not all of it is masked. Otherwise, those high quality SACDs could be quantized down (without noise shaping) to 11 bits at 44100 Hz, and sound the same, which can be proven false with ABX testing (see also this post).


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## spaark

xnor said:


> What is non-linear noise? What's the opposite?


 
 Non-linear noise is noise produced by a non-linear system. If we're being pedantic, I suppose it's wrong to say the noise is non-linear, because it's the system that is.
  


gregorio said:


> I'm not sure I would use the term linear or non-linear in this context, correlated or de-correlated would perhaps be more accurate and less confusing. The act of dithering is the act of de-correlating the quantisation error, which is essentially just a fancy and more precise way of saying the errors are turned into random noise. If we're going into the finer detail, I'm not sure the last sentence I've quoted of yours is entirely accurate; generally noise shaped dither would actually slightly increase the total amount of noise but of course significantly decrease the amount of audible noise.


 
 Less confusing (or rather more insightful) perhaps, because the correlation of quantisation errors is the reason behind non-linearity, but not more accurate. I'm just getting to the crux of the matter. The reason dithering improves fidelity is because it eliminates the non-linearity i.e. the distortion, which is much more audible than white noise of the same energy.
  
 I think you're right than it slightly increases the amount of noise as far as real noise shaping techniques are concerned, but we'd like an ideal one to simply distribute it.


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## sarasa

I've just came across this thread...
  
 Quote:


> [...]The result is that we have an absolutely perfect measurement of the waveform (2*) plus some noise.


 
  
 From my understanding, I think that statement isn't correct, of course the Nyquist Theorem is right, but conditions to use the theorem weren't met since the measuring errors aren't expected, I do agree that the new waveform probably isn't that far away from the original one... but how far away? in most of these threads I usually don't find those errors expressed as a amplitude quantization function and someone who shows the math in order to really show if something is really improved or not...
  
 I would expect that a higher frequency / bit accuracy could improve the relation to the original signal... I don't have any problem in being wrong, though...


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## stv014

> Originally Posted by *sarasa* /img/forum/go_quote.gif
> 
> From my understanding, I think that statement isn't correct, of course the Nyquist Theorem is right, but conditions to use the theorem weren't met since the measuring errors aren't expected, I do agree that the new waveform probably isn't that far away from the original one... but how far away? in most of these threads I usually don't find those errors expressed as a amplitude quantization function and someone who shows the math in order to really show if something is really improved or not...
> 
> I would expect that a higher frequency / bit accuracy could improve the relation to the original signal... I don't have any problem in being wrong, though...


 
  
 In the digital part of the reconstruction filter, image rejection for a 20 kHz tone at 44.1 kHz sample rate can be 120 dB or better without major difficulty (using a FIR filter with a length of <= 2 ms). There is some low-ish level imaging far outside the audio band after the analog reconstruction of the oversampled signal, though. However, this (other than being unlikely to be an audible issue) actually might not be improved by using a higher sample rate if it is still oversampled to the same rate anyway.
  
 Dithered quantization simply adds noise to the signal. The amount of noise in the audio band depends on the bit depth (every additional bit improves it by 6.02 dB, but analog hardware usually becomes a limiting factor above about 20 bits, and 16 bits should normally be enough in practice for music), noise shaping - if any, and to some extent on the sample rate (because increasing it moves more of the same total noise power into the ultrasonic range).


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## xnor

sarasa said:


> From my understanding, I think that statement isn't correct, of course the Nyquist Theorem is right, but conditions to use the theorem weren't met since the measuring errors aren't expected, I do agree that the new waveform probably isn't that far away from the original one... but how far away? in most of these threads I usually don't find those errors expressed as a amplitude quantization function and someone who shows the math in order to really show if something is really improved or not...


 
  
 It's quite simple actually.
  
 You quantize an analog value x to xq. This will add some quantization error eq.
  
 xq = x + eq    with an SNR = 20*log10(xrms / erms)
  
 The error signal cannot escape the sampling theorem, so it is low-level noise from DC to Fs/2. You can theoretically perfectly reconstruct the signal including low-level noise given by all xq values.


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## gregorio

sarasa said:


> From my understanding, I think that statement isn't correct, of course the Nyquist Theorem is right, but conditions to use the theorem weren't met since the measuring errors aren't expected, I do agree that the new waveform probably isn't that far away from the original one... but how far away?


 
 That's impossible to say because it depends on the design and construction of each individual ADC and DAC. In general though, the actual conversion process to and from digital is perfect or at least perfect to orders of magnitude more than the ear could possibly detect. So much so infact that the limiting factor of every ADC and DAC on the market is the analog input stage (in and ADC) and the analogue output stage (of a DAC) rather than the actual conversion process itself and this is true of even the best ADCs and DACs, employing the finest of components. Even very cheap ADCs and DACs these days have virtually linear responses and of course there are only a handful of companies which manufacture the actual digital conversion chips.
  
 Quote:


sarasa said:


> I would expect that a higher frequency / bit accuracy could improve the relation to the original signal...


 
 To effectively implement the Nyquist Theorem, up to a point it can! But beyond that point the accuracy actually deteriorates simply due to the fact that the higher the sample rate the less time there is to process each sample. So, you can have high sample rates and few bits or lots of bits and a low sample rate but not both without compromising accuracy! In practise, modern ADCs take the former approach and initially sample at a frequency of many megahertz but with only a handful or so of bits. This initial sampling is then decimated down to the sample rate/bit depth selected by the user. Sample rates/bit depths from 16/44 up to 24/96 appear to fall into the optimum window for accuracy but with sampling rates beyond 24/96, accuracy deteriorates. I explained in a bit more detail at the end of this post.
  
 G


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## Don Hills

gregorio said:


> ...  At these very high sample rates and bit depths we start hitting the limits of the laws of physics in how fast we can perform the calculations required to implement a filter which reduces anti-aliasing to below the digital noise floor.  ...


 
  
 Thank you for your explanation.


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## Matvei

So, I am about to buy the Fostex HP-A4 DAC which comes out next week in Japan, and I had some questions relating to this thread in terms of PCM vs DSD. For reference, the DAC supports up to 24bit 192kHz in PCM, and up to 5.6MHz!!! in DSD

It comes with a new application called Fostex Audio Player to allow playback of these file types on the fly.

If I understand this thread correctly, 24 bit PCM is useless, as are 96kHz and 192kHz sampling rates. Therefore, HD Audio is snake oil. 

I'm not quite sure I understand DSD. It was said in this thread and others as well by Giorgio that although SACD shouldn't sound any better than redbook, they often do because they have better masters. In that case, wouldn't that mean the only use of the DSD feature is to play back (mostly illegally) ripped SACDs? And what on earth is the difference between 2.8MHz and 5.6MHz DSD? Is there any point to using this crap or are these features all extraneous?


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## gregorio

matvei said:


> So, I am about to buy the Fostex HP-A4 DAC which comes out next week in Japan, and I had some questions relating to this thread in terms of PCM vs DSD. For reference, the DAC supports up to 24bit 192kHz in PCM, and up to 5.6MHz!!! in DSD
> 
> If I understand this thread correctly, 24 bit PCM is useless, as are 96kHz and 192kHz sampling rates. Therefore, HD Audio is snake oil.


 
 If all you are doing is playing commercially released recordings, then yes, 24bit is useless. It's not useless if you are recording or mixing music though. Again, 96kHz may have some uses during recording and mixing but not so much for playback. 192kHz is useless for virtually everything, including recording, mixing and playing back commercial recordings!
  


matvei said:


> I'm not quite sure I understand DSD. It was said in this thread and others as well by Giorgio that although SACD shouldn't sound any better than redbook, they often do because they have better masters. In that case, wouldn't that mean the only use of the DSD feature is to play back (mostly illegally) ripped SACDs? And what on earth is the difference between 2.8MHz and 5.6MHz DSD? Is there any point to using this crap or are these features all extraneous?


 

 DSD uses 1bit rather than 16bit or 24bit. The downside of this is huge amounts of unwanted noise but with very high sampling rates there is plenty of space above the audible spectrum to move this noise, thereby making it inaudible. The SACD standard has a sampling rate of 2.8mHz and therefore a theoretical audio range of 1.4mHz, which is way more than enough to account for any difficulties related to reconstruction filters or other required process. This DAC obviously has a mode which exactly doubles that and indeed there are apparently some recordings available in this format and one or two DAC manufacturers are now offering a quad DSD oversampled rate of 11.2mHz. It appears that these oversampled DSD rates which have appeared are purely for marketing (snake oil) purposes. On the basis that if something has a higher number in it's specs, it's easier to market it as better, even if it isn't or even if it's actually worse!
  
 Your basic premise is correct, look for the quality of the recording rather than for the distribution format. In some cases the best you will find will be a standard CD (16/44) in other cases it might be SACD or 24/96. You can always take a good master which may happen to be in 24/96 format, convert to a lossless 16/44 format to save space on say your smartphone and be secure in the knowledge that you're not loosing anything audible and are therefore listening to the highest quality available!
  
 G


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## MrMateoHead

I shared this link in a different thread:
  
http://www.npr.org/blogs/therecord/2013/09/11/219727031/what-does-a-song-that-costs-5-sound-like
  
 You can get some of the history of DSD from that article, and sense of why it is being revived today. The conclusion would appear that people are willing to pay more for "high quality", though what is and is not high quality is the hard part. I, for one, don't want to pay $5 for a file download and $50 for an album, but that's just me. The file sizes are atrocious, and there is no guarantee that a 'warm analog like' sound will be heard on each listen. The analog / digital snake oil bid drives me nuts.
  
 When I complain about "sound quality" sometimes, I am not complaining about a specific format. What I am complaining about is clipped recordings, or recordings that seem stripped of dynamic range or for whatever reasons are harsh and nasty. An Mp3 has the same subjective properties (to my ears) as the CD it was ripped from, so I am not convinced that new formats and new equipment = better listening. Good headphones have made me aware just how far my recordings range (from amazing to disappointing). Are there ANY standards or guidelines for making good recordings these days?


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## xnor

> Are there ANY standards or guidelines for making good recordings these days?


 
  
 Standards don't help if your "boss" tells you to "make it louder" and work at a faster pace.


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## Copperears

I remember years ago someone writing about the fact that even an unplugged resistor sitting quietly inside a desk drawer is generating self-noise of about 2-3db.

Per resistor.

I decided at that point (1992?) that all the folderol about any playback above 16-bit/44kHz was just that, folderol, especially as I've never lived in an anechoic chamber entirely sealed from street/refrigerator/lightbulb/HVAC noise, any of which probably squash the dynamic listening range to about 35-40db.

And when I've gotten close to that, with IEMs, there's the - what - 20db of white noise that is my blood coursing through my veins and arteries.

Vampires may not experience that, but then again I've heard tell they have a terrible time with tinnitus.




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## Copperears

Absolutely fascinating!

http://youtu.be/j6PpDQ6miBg




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## xnor

copperears said:


> I remember years ago someone writing about the fact that even an unplugged resistor sitting quietly inside a desk drawer is generating self-noise of about 2-3db.
> 
> Per resistor.


 
  
 Ahhh.. that is what raised the noise floor in my system from -100 dB to over +1000 dB!!! Now I can rest in peace.


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## Copperears

Did I say something ludicrous, again?! Damn......

Resistors were bigger back then...... I guess they're smaller and quieter now..... hmmm.... I'll keep trying....

Perhaps I misremembered what I read? Impossible.....


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## bigshot

I use Magic Pebbles to combat my noise floor problem.


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## xnor

copperears said:


> Resistors were bigger back then...... I guess they're smaller and quieter now..... hmmm.... I'll keep trying....


 
 Yeah, I also think it's the size that matters. A bigger passive component, unconnected, in a desk drawer definitely generates more noise, especially the black ones (they absorb more quantum energy and that energy has to go somewhere ... so it's being radiated into the room and therefore raises the noise floor ... Q.E.D.)!
  


bigshot said:


> I use Magic Pebbles to combat my noise floor problem.


 
 My floor is made entirely of magic pebbles. Somehow the product doesn't work as advertised though because each step makes a helluva lot of noise!


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## Copperears

Clearly you are not using quantum-noise-negating pebbles. They're blue and also resistant to black holes.


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## dotrunghieu

_Excellent_ thread!


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## Copperears

Something like this is probably what I was dimly thinking of:

http://www.analog.com/static/imported-files/rarely_asked_questions/moreInfo_raq_resistors.html

Applicable only to analogue circuits but hey, it could spawn a new round of hi-end hi-fi pseudo-scientific marketing frenzy!

Always happy to help..... 


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## Copperears

My fave line in that article:

"It is not possible to change Boltzmann's Constant because Professor Boltzmann is dead."



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## Copperears

The Imp of the Perverse strikes again:

http://en.wikipedia.org/wiki/Boltzmann_constant

I Love Math.



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## Copperears

Even better!

http://scitation.aip.org/content/aip/journal/apl/100/20/10.1063/1.4717462




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## Copperears

.... And for those of you interested in nano wires:

http://en.wikipedia.org/wiki/Nanowire

They _are_ the future!



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## Copperears

Also:

http://en.wikipedia.org/wiki/Nanocircuitry

My bet is on quantum dot cellular automata, systems that will self-replicate in the quantic realm in response to input. But then there's the problem of what to do with automata overpopulation and death. Perhaps some sort of sub-quantum recycling process will be discovered that opens the doorway between known and dark matter, to solve that problem....


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## krtzer

copperears said:


> Also:
> 
> http://en.wikipedia.org/wiki/Nanocircuitry
> 
> ...


 
 Quantum dots are the way of the future, guys,
  
 http://en.wikipedia.org/wiki/Quantum_dot
  
 In all seriousness though, quantum dots are have some interesting properties. It seems like we don't exactly know what to do with them yet.


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## Copperears

Yes, I was joking around but hopefully you humans will figure this out correctly in time...... 



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## Lespectraal

So I read the whole thread, all 96 pages of it. Boy have I never been so interested in digital signal processing. I'd like to thank gregorio, xnor, D. Lundberg, Skamp, Jaddie and possibly other members who have actively posted in this thread and inputting valuable insight and knowledge on this subject matter. I will use this information with good intention and make my life a better one starting from now on. I shall be more cautious with future marketing tricks and make sure my wallet is intact.
  
 I used to think that higher sampling rate and bit depths equates to better sounds, but it is not the case. Well I actually had a feeling that it was so because before this I had many recording of different bit depths and sampling rates, all of them are capable of great sound. So I thought maybe this whole hi-rez 24/192 is just a bunch of hoohaa. But once I stumbled upon this thread, now I am better informed than I was. Now I know the real deal.

 Again, thank you for this wonderful thread.

 *listens to some John Mayer in 16bit/44.1kHz*


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## CantScareMe

lespectraal said:


> So I read the whole thread, all 96 pages of it. Boy have I never been so interested in digital signal processing. I'd like to thank gregorio, xnor, D. Lundberg, Skamp, Jaddie and possibly other members who have actively posted in this thread and inputting valuable insight and knowledge on this subject matter. I will use this information with good intention and make my life a better one starting from now on. I shall be more cautious with future marketing tricks and make sure my wallet is intact.
> 
> I used to think that higher sampling rate and bit depths equates to better sounds, but it is not the case. Well I actually had a feeling that it was so because before this I had many recording of different bit depths and sampling rates, all of them are capable of great sound. So I thought maybe this whole hi-rez 24/192 is just a bunch of hoohaa. But once I stumbled upon this thread, now I am better informed than I was. Now I know the real deal.
> 
> ...


 
  
_So I read the whole thread_
 Funny - I was about to do the same thing. I read the first post and skipped to the last. Bad habit nowadays 
  
 I've come across this resolution stuff before and the math/validation here is solid in my opinion. With two files of the exact same recording/mastering, my ears have never been able to tell a difference.


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## bunner

So after reading several pages of this thread it does strike me as interesting. While I agree most users, whether trained professional ears or not, could be fooled by an ABX of varied differences between bit-rates or 16/24 bit sample sizes.. But what is interesting to me is saying its pointless to achieve or use higher sample rates because we can't possibly hear anything above a certain frequency. The goal of sampling music or images in our surrounding life is to be able to recreate it the best we can; however, we haven't possibly understood how everything in this world works, the best we can do to capture it, is to sample it by recording it with our best means possible, but it still isn't the same as the moment you experienced it with your own eyes or ears. Then why wouldn't you want the highest possible bit-rate and depth to capture it?
  
 Questions to consider:
  
 1) Even if you can't hear a frequency by limitations of the ear, have we proved we can't feel it? 
 2) A 4000 pixel video looks more lifelike (almost said to be 3D, seems near real) on a 4K monitor compared to a 1920x1080 pixel blu-ray; couldn't oversampling just be that much more closer to the real life event when it happened? 
 3) Listening to the live music of the actual band sounds more lively/real than any recording sampled and recreated with any equipment I have ever heard... So the question begs, why haven't we been able to recreate that? Perhaps there is more work to be done on this audio capturing and recreation, in my opinion, rather than settling for 16-bit just because it sounds just like a 24-bit capture. 
  
 Maybe this is more rhetorical but I just don't think we have mastered, or completely understood anything/everything as a human species.


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## esldude

bunner said:


> So after reading several pages of this thread it does strike me as interesting. While I agree most users, whether trained professional ears or not, could be fooled by an ABX of varied differences between bit-rates or 16/24 bit sample sizes.. But what is interesting to me is saying its pointless to achieve or use higher sample rates because we can't possibly hear anything above a certain frequency. The goal of sampling music or images in our surrounding life is to be able to recreate it the best we can; however, we haven't possibly understood how everything in this world works, the best we can do to capture it, is to sample it by recording it with our best means possible, but it still isn't the same as the moment you experienced it with your own eyes or ears. Then why wouldn't you want the highest possible bit-rate and depth to capture it?
> 
> Questions to consider:
> 
> ...


 

 Maybe this is rhetorical, but since we haven't understood everything we by definition don't understand anything?  Please not this tired old canard. 
  
 You say listening to live music no recording matches that.  I agree.  Will it match it suddenly at 32 bit or 48 bit? You do realize it is near impossible to exceed a noise floor in the analog world beyond 20 bit don't you?
  
 4K video does provide pixels of a size within the resolving power of our eyes at the correct distance.  192khz sampling rates aren't analogous to that.  And even 4k pictures from too far away look no better than 2k pictures.  This happens when they fall below the resolution of the eye in angular terms.  192 khz sampling is more analogous to saying video displays with ultraviolet light output beyond human vision look better even though we cannot see it.  More than 24 bit in audio would be analogous to having a range from total darkness to brighter than our eye can stand without being blinded (or ears deafened).
  
 Now the reason live music is not fully re-created has to do with soundfields imposed upon a different room from the recording.  24 bit allows definition in terms of loudness from total quiet to louder than we could stand with enough fineness there is no issue.  48 khz or so sampling covers the frequencies we can hear.  The complete recreation of the soundfield at our ears is another matter and not limited by the basic resolution of digital audio at normal sample rates.  Any advance is in manipulating the signals we get to our hears, not in the basic fidelity of the digital audio medium.  That might mean more channels and/or different type miking and some more advanced DSP.  Those higher bits and sample rates get us no closer to a solution for those problems.  Stereo is a convincing illusion or trick anyway.


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## bunner

> Maybe this is rhetorical, but since we haven't understood everything we by definition don't understand anything?  Please not this tired old canard. ​


 
 So by your definition, there is no reason to understand everything completely as we already understand enough to prove what we already think we know. I'd prefer to look forward, or we'd still be listening to AM radio. 
  


> You say listening to live music no recording matches that. * I agree*.  Will it match it suddenly at 32 bit or 48 bit? You do realize it is near impossible to exceed a noise floor in the analog world beyond 20 bit don't you?


 
 By agreeing, then by definition, you realize our method of capturing and recreating is flawed. So sample rate at this point doesn't matter. Which is my point, we have yet to recreate the sound and feel of the music that was created by the instruments and voices of the original performer. 
  


> 4K video does provide pixels of a size within the resolving power of our eyes at the correct distance.  192khz sampling rates aren't analogous to that.  And even 4k pictures from too far away look no better than 2k pictures.  This happens when they fall below the resolution of the eye in angular terms.  192 khz sampling is more analogous to saying video displays with ultraviolet light output beyond human vision look better even though we cannot see it.  More than 24 bit in audio would be analogous to having a range from total darkness to brighter than our eye can stand without being blinded (or ears deafened).
> 
> Now the reason live music is not fully re-created has to do with soundfields imposed upon a different room from the recording.  24 bit allows definition in terms of loudness from total quiet to louder than we could stand with enough fineness there is no issue.  48 khz or so sampling covers the frequencies we can hear.  The complete recreation of the soundfield at our ears is another matter and not limited by the basic resolution of digital audio at normal sample rates.  Any advance is in manipulating the signals we get to our hears, not in the basic fidelity of the digital audio medium.  That might mean more channels and/or different type miking and some more advanced DSP.  Those higher bits and sample rates get us no closer to a solution for those problems.  Stereo is a convincing illusion or trick anyway.


 
 I appreciate your write up here, it is in depth. But by just accepting 48khz sampling as the final answer, just because it covers our ears frequencies seems disappointing.. because--even by your confirmation above-- that we have yet to match/capture a live music experience, seems to prove we have to be missing something.  I still think we have a long way to go in the recreation of the full visual and audio recreation of what this real world has to offer. Is there a way to truly capture it? Perhaps not, but you cannot deny the magnificence of what is real. To go back to color as an example. Color is relative, and a dog's eyes have more rods than cones to see better in darkness that which they see in this world. Humans have 3 (RGB) cone receptors. Butterflies have 5 (we have yet to determine what they fully see), Mantis Shrimp have 16 (This is beyond our comprehension). The point is, we really don't know all the answers in life, and have a long way to go in being able to recreate it. Far from a 4000 pixel image or 16/24 bit+ audio sample in comparison.


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## esldude

bunner said:


> So by your definition, there is no reason to understand everything completely as we already understand enough to prove what we already think we know. I'd prefer to look forward, or we'd still be listening to AM radio.
> 
> By agreeing, then by definition, you realize our method of capturing and recreating is flawed. So sample rate at this point doesn't matter. Which is my point, we have yet to recreate the sound and feel of the music that was created by the instruments and voices of the original performer.
> 
> I appreciate your write up here, it is in depth. But by just accepting 48khz sampling as the final answer, just because it covers our ears frequencies seems disappointing.. because--even by your confirmation above-- that we have yet to match/capture a live music experience, seems to prove we have to be missing something.  I still think we have a long way to go in the recreation of the full visual and audio recreation of what this real world has to offer. Is there a way to truly capture it? Perhaps not, but you cannot deny the magnificence of what is real. To go back to color as an example. Color is relative, and a dog's eyes have more rods than cones to see better in darkness that which they see in this world. Humans have 3 (RGB) cone receptors. Butterflies have 5 (we have yet to determine what they fully see), Mantis Shrimp have 16 (This is beyond our comprehension). The point is, we really don't know all the answers in life, and have a long way to go in being able to recreate it. Far from a 4000 pixel image or 16/24 bit+ audio sample in comparison.


 

 You do realize the test has been done more than once.  No difference between 176 khz and 44khz.  No difference between 24 bit and 16 bit was detected.  With several hundred people involved in one such test.  Maybe we can  gain something from 24 bit if other processing goes on.  Though for final playback probably not.  Going further isn't getting us one bit closer to your goal of recreating life music.  We are already beyond what human ears can ear.  The general" hey more is better" idea you are portraying and a feeling of faith it gets us closer isn't rational.  We don't know everything, we know more than we used to and yes some of what we know does show limits for what makes any sense.  5 megahertz audio bandwidth is no better than 1 mhz is no better than 20 khz bandwidth if we are using human ears to hear it.   Whatever stands between us recreating life music is not lacking in these basic parameters. 
  
 So no, increased sample rate doesn't matter.  Same for going beyond 24 bit.  Accepting 48 khz as the answer because it covers our ears frequency range may disappoint you, but if we are making this for humans it simply is enough.  The idea of lets just up it and hope it is better is not science.  This is the Sound Science forum you are in here. 
  
 You also are either twisting or misunderstanding me to say we know all we need to know and not to know more is worthwhile.  I specifically said we don't know.  However, that as an excuse to ignore what is known is simply irrational.  I can go over many things, but perhaps it makes more sense for you to go back and read the rest of this thread.  No sense in me repeating what is already here.


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## hogger129

I've never felt 24 bit was necessary for anything other than mastering.


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## bunner

esldude said:


> You do realize the test has been done more than once.  No difference between 176 khz and 44khz.  No difference between 24 bit and 16 bit was detected.  With several hundred people involved in one such test.  Maybe we can  gain something from 24 bit if other processing goes on.  Though for final playback probably not.  Going further isn't getting us one bit closer to your goal of recreating life music.  We are already beyond what human ears can ear.  The general" hey more is better" idea you are portraying and a feeling of faith it gets us closer isn't rational.  We don't know everything, we know more than we used to and yes some of what we know does show limits for what makes any sense.  5 megahertz audio bandwidth is no better than 1 mhz is no better than 20 khz bandwidth if we are using human ears to hear it.   Whatever stands between us recreating life music is not lacking in these basic parameters.
> 
> So no, increased sample rate doesn't matter.  Same for going beyond 24 bit.  Accepting 48 khz as the answer because it covers our ears frequency range may disappoint you, *but if we are making this for humans it simply is enough.  The idea of lets just up it and hope it is better is not science.  This is the Sound Science forum you are in here. *
> 
> You also are either twisting or misunderstanding me to say we know all we need to know and not to know more is worthwhile.  I specifically said we don't know.  *However, that as an excuse to ignore what is known is simply irrational.*  I can go over many things, but perhaps it makes more sense for you to go back and read the rest of this thread.  No sense in me repeating what is already here.


 
 I grant you this, you are fully correct on this for a cold one-dimensional sound as it pertains to an acceptable (seemingly) real recreation to human ears. I am not trying to argue with you, I am just curious is all.
  
 But you cannot deny that 48khz will be obsolete when future studies show there is more to humans perceiving sound than just the audible spectrum. Lower frequencies can be felt, it is a science fact. Higher subsonic frequencies are yet to be determined. Given that ultrasonic frequencies operate in the 60khz range, would require a sample rate of 120khz. Just because we don't audibly hear them doesn't mean they shouldn't exist or cannot be recreated with the intention of reproducing a real life sound.
  
http://en.wikipedia.org/wiki/HyperSonic_Sound#HyperSonic_Sound
  
 We also don't understand energy and how it pertains to real life human beings. Do aura's exist? Does emotion have a real energy as the performer is singing it? Why do animals sense emotions of human being's moods? Why can some dogs detect cancer in an individual and other dogs/cats know or are able to predict when someone is about to die? If you could prove a soul existed, it would be the biggest discovery of all time. It's easier to laugh off and call ideas we don't understand as crazy.. but my argument still stands; we have yet to recreate a life-like recreation of a live musical event. So if this is a sound science forum, then we should at least entertain the possibility that we don't know all of the answers for what could make sound recreation better; just that we DO understand the one that our limited minds accept as fact because it fools us enough to accept it. Science is about discovery and imagination, and then proving theories, not about accepting an outcome as indisputable fact. Smarter people than you or I are probably already working on these studies, but I personally want to hear a better recreation of what is real, and our current model of accepting 48khz DOES NOT do this.


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## bunner

esldude said:


> You do realize the test has been done more than once.  No difference between 176 khz and 44khz.  No difference between 24 bit and 16 bit was detected.  With several hundred people involved in one such test.  Maybe we can  gain something from 24 bit if other processing goes on.  Though for final playback probably not.  Going further isn't getting us one bit closer to your goal of recreating life music.  We are already beyond what human ears can ear.  The general" hey more is better" idea you are portraying and a feeling of faith it gets us closer isn't rational.  We don't know everything, we know more than we used to and yes some of what we know does show limits for what makes any sense.  5 megahertz audio bandwidth is no better than 1 mhz is no better than 20 khz bandwidth if we are using human ears to hear it.   Whatever stands between us recreating life music is not lacking in these basic parameters.
> 
> So no, increased sample rate doesn't matter.  Same for going beyond 24 bit.  Accepting 48 khz as the answer because it covers our ears frequency range may disappoint you,* but if we are making this for humans it simply is enough.  The idea of lets just up it and hope it is better is not science.  This is the Sound Science forum you are in here. *
> 
> You also are either twisting or misunderstanding me to say we know all we need to know and not to know more is worthwhile.  I specifically said we don't know. * However, that as an excuse to ignore what is known is simply irrational.  *I can go over many things, but perhaps it makes more sense for you to go back and read the rest of this thread.  No sense in me repeating what is already here.


 
 I grant you this, you are fully correct on this for one-dimensional sound as it pertains to acceptable seemingly real recreation to human ears. I am not looking for an argument, I just refuse to accept an end game of 1 fact, compared to what and where we could possibly go.
  
 But you cannot deny that 48khz will be obsolete when future studies show there is more to humans perceiving sound than just the audible spectrum of our ears. Lower frequencies can be felt, it is a science fact. Higher subsonic frequencies are yet to be determined. Given that ultrasonic frequencies operate in the 60khz range, would require a sample rate of 120khz. Just because we don't audibly hear them doesn't mean they shouldn't exist or cannot be recreated with the intention of reproducing a real life sound. You can look up hypersonic sound and its creation from Elwood Norris on your own; I am not allowed to share external links. I did do a search of this and haven't seen it addressed.
  
 We also don't understand energy pertaining to humans and animals. Do aura's exist? Sometimes animals and people can sense emotion. Is there an emotion given off with live music that cannot be recreated? It's easier to laugh off and call ideas we don't understand as crazy.. but my argument still stands; we have yet to recreate a lifelike recreation of a live musical event. So 48khz does NOT completely represent a full sampling of recreating live sound. Just the common accepted belief of representing a one-dimensional signal to fool our ears capability of 20hz-20khz. Sure we can live with this cold fact, or someone can do better to figure out beyond what we currently accept; and possibly recreate the real experience. I am hoping for the latter. I'm not expecting you to debate every single item stated above, as being in a position to prove what neither of us know is pointless. I just think its foolish for all of us to ignore the unknown, and claim its irrational to want to understand it.


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## esldude

bunner said:


> I grant you this, you are fully correct on this for one-dimensional sound as it pertains to acceptable seemingly real recreation to human ears. I am not looking for an argument, I just refuse to accept an end game of 1 fact, compared to what and where we could possibly go.
> 
> But you cannot deny that 48khz will be obsolete when future studies show there is more to humans perceiving sound than just the audible spectrum of our ears. Lower frequencies can be felt, it is a science fact. Higher subsonic frequencies are yet to be determined. Given that ultrasonic frequencies operate in the 60khz range, would require a sample rate of 120khz. Just because we don't audibly hear them doesn't mean they shouldn't exist or cannot be recreated with the intention of reproducing a real life sound. You can look up hypersonic sound and its creation from Elwood Norris on your own; I am not allowed to share external links. I did do a search of this and haven't seen it addressed.
> 
> We also don't understand energy pertaining to humans and animals. Do aura's exist? Sometimes animals and people can sense emotion. Is there an emotion given off with live music that cannot be recreated? It's easier to laugh off and call ideas we don't understand as crazy.. but my argument still stands; we have yet to recreate a lifelike recreation of a live musical event. So 48khz does NOT completely represent a full sampling of recreating live sound. Just the common accepted belief of representing a one-dimensional signal to fool our ears capability of 20hz-20khz. Sure we can live with this cold fact, or someone can do better to figure out beyond what we currently accept; and possibly recreate the real experience. I am hoping for the latter. I'm not expecting you to debate every single item stated above, as being in a position to prove what neither of us know is pointless. I just think its foolish for all of us to ignore the unknown, and claim its irrational to want to understand it.


 

 I guess I agree with one statement you made.  Debating what neither of us can know is pointless.  You speak about future studies that will obsolete 48khz as if you believe and have faith it will happen.  That kind of faith and wishful thinking is not science.  There is no rational debate about imagined things that might occur.  Get back to us when those studies happen that obsolete 48 khz.  The refusal to accept facts is anything except scientific.  The rest of your babble about aura's and emotion is not scientific.  Nor rational.  If you think you have made damning criticism, I fear you don't understand the issue.  So far you seem determined to not understand it if it conflicts with your evidence-less feelings of how things might work.  You won't progress very far that way.


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## bigshot

There have already been studies on the effect of super audible frequencies. Frequencies beyond the range of human hearing have no impact on the sound quality of recorded music at all. Frequency response is pretty much fully understood. And high bitrate sound is indeed primarily for mixing and mastering, not playing back music in the home. Redbook covers that completely.
  
 If you want an area that has potential for learning new things about improving recorded sound quality, that would be multichannel sound and synthesized acoustic environments, not bigger files with lots of information we can't hear. But the real trick here isn't scientific as much as it is aesthetic. Engineers need to learn how to apply the tools we already have.


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## bunner

esldude said:


> I guess I agree with one statement you made.  Debating what neither of us can know is pointless.  You speak about future studies that will obsolete 48khz as if you believe and have faith it will happen.  That kind of faith and wishful thinking is not science.  There is no rational debate about imagined things that might occur.  *Get back to us when those studies happen that obsolete 48 khz. * The refusal to accept facts is anything except scientific.  The rest of your babble about aura's and emotion is not scientific.  Nor rational.  If you think you have made damning criticism, I fear you don't understand the issue.  So far you seem determined to not understand it if it conflicts with your evidence-less feelings of how things might work.  You won't progress very far that way.


 
  
 I already addressed this. Ultrasonic/Hypersonic frequencies has already rendered 48khz obsolete. Feel free to live in the past, and hang on to the 48khz theory. It just doesn't do a full recreate of life like sound. It's easy to dismiss something as babble, but they are real rational questions that prove flaws in the inability for 48khz to recreate sound naturally as they were originally produced or felt. Just because you refuse to accept something doesn't mean it does not exist.  If you really think what I am talking about is out there and nonsense, you are free to think whatever you want; google the studies of ultrasonic frequencies. I feel I am done here debating you, as you cannot provide answers that can substantially disprove what I have already addressed.


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## bigshot

They did that test in the AES. The results were that although super high frequencies might be perceivable by some people as sound pressure in test tones at high volumes, they didn't improve the sound quality of music at all.


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## Digitalchkn

Another point is that spectrum analysis of all 192K and even 96K (at least ones I looked at) production files doesn't really reveal anything but noise in the upper registers of the recording spectrum. Basically tells me that either a) there is not much up there to record and/or b)   the signal chain (transducers, amps, cables, mixers, so on) is band limiting the recording well before the sampling rate limitations kick in.


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## esldude

Yeah, seriously, if you use Earthworks mikes or some such you have the extraordinary bandwidth of about 40 khz.  Generally no mikes respond further, and in fact few highly regarded mikes go even that far.  So much for extended bandwidth.  There simply won't be any signal to work with.  88 or 96 khz sampling is simply all you could possibly offer up in any manner with available microphones. 
  
 Excepting of course the faith that future results will show human response to extreme frequencies (which we already know for a fact, yes an actual fact, will not involve perception by the ears), and future mikes that have wide bandwidth response.  Like response more than 40 khz man.  Check your aura and all that.
  
 Peace brother....................................


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## Digitalchkn

esldude said:


> Yeah, seriously, if you use Earthworks mikes or some such you have the extraordinary bandwidth of about 40 khz.  Generally no mikes respond further, and in fact few highly regarded mikes go even that far.  So much for extended bandwidth.  There simply won't be any signal to work with.  88 or 96 khz sampling is simply all you could possibly offer up in any manner with available microphones.
> 
> Excepting of course the faith that future results will show human response to extreme frequencies (which we already know for a fact, yes an actual fact, will not involve perception by the ears), and future mikes that have wide bandwidth response.  Like response more than 40 khz man.  Check your aura and all that.
> 
> Peace brother....................................


 
  
 I am tempted to think there probably just isn't much signal energy up there anyway.  Since we are not bats we naturally designed things to occupy the spectrum that gives us most visceral impact. As a consequence, things that we make noise with are mechanically large enough and likely can't resonate with large amounts of energy at extreme harmonics.  For instance, violin strings which are smallest of stringed instruments, fundamentally resonate up to only few kilohertz and I think have any measurable harmonics maybe to ~30KHz.  They just physically can't resonate faster.


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## ferday

digitalchkn said:


> I am tempted to think there probably just isn't much signal energy up there anyway.  Since we are not bats we naturally designed things to occupy the spectrum that gives us most visceral impact. As a consequence, things that we make noise with are mechanically large enough and likely can't resonate with large amounts of energy at extreme harmonics.  For instance, violin strings which are smallest of stringed instruments, fundamentally resonate up to only few kilohertz and I think have any measurable harmonics maybe to ~30KHz.  They just physically can't resonate faster.




I'm tempted to agree with this!

Since we are surrounded by ultrasonic frequencies all the time, I can see how some would propose it as something needed to 'complete' the signal to make it more natural. 

However in my mind it's the fault of the transducers (not to even mention the recording process itself) that a recording can't match the wild untamed noise fronts of a live performance, stereo or multichannel is still just an illusion anyways. Interesting to me is at a rock concert you are essentially hearing the sound from multiple transducers where an orchestra is more of a natural presentation...


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## bigshot

The other thing that isn't apparent looking at numbers without context is that from 20kHz to 40kHz is just one octave. It's just a sliver of sound anyway.


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## Digitalchkn

> ...Since we are surrounded by ultrasonic frequencies all the time, I can see how some would propose it as something needed to 'complete' the signal to make it more natural...


 
  
 I saw an article just yesterday (can't remember where) that sighted a study done in the mid-70s where the conclusion is that our acuity could resolve on the order of 1 degree phase offset of a 250Hz test tones, translating to something like nearly 10uS timing resolution. So the conclusion was we need sampling rate that covers that kind of bandwidth. But that's not actually correct. We only would need sampling rate that sufficiently covers the 250Hz. In other words, don't expect some signal to suddenly show up at 100KHz.


----------



## ferday

i'm not saying i think ultrasonics matter, but here's just one (of many) articles (here's another one that i don't really like) that hint that perhaps noises outside of our hearing range can in fact be detected and processed, just not as "sound".  heck, we all know that though we can't _hear_ 10Hz, if it's played loud enough there is no one that won't know it's there.  there is at bare minimum, a basis to be looking into these types of things.


----------



## bigshot

The point is music. The only place super audible frequencies exist in music is in upper harmonics of cymbal crashes. And even if we could hear it (which we can't) it would probably be masked by frequencies we can hear.
  
 If someone wants to look into sound for the purposes of improving reproduction of music, they would use their time much more profitably by focusing on sound that can actually be heard by humans.


----------



## ab initio

bigshot said:


> The point is music. The only place super audible frequencies exist in music is in upper harmonics of cymbal crashes. And even if we could hear it (which we can't) it would probably be masked by frequencies we can hear.
> 
> If someone wants to look into sound for the purposes of improving reproduction of music, they would use their time much more profitably by focusing on sound that can actually be heard by humans.


 

 With all due respect, you're forgetting those of us who are more machine than man...
  

  
 I suspect 24bit/320kHz is a tad more important for those who've succumbed to the dark side.
  
  
 Cheers


----------



## Don Hills

monstermunch said:


> Hi-res audio is the perfect concept for robbing all these poor buggers at the same time.


 
 This leads to an internal conflict... whether to attempt to warn and educate them, or to fleece them.


----------



## jonbernard

monstermunch said:


> I have long suspected that there is a subsect of the audiophile community which contains individuals who actually don't like music very much, who are caught up in an obsessive compulsive disorder of sorts, whereby the search for technical improvement has become the means and the end in and of itself.


 
  
 There are reviewers here on Head-Fi who admit they aren't interested in music very much, and whose interest is purely in audio reproduction for its own sake.
  


> These are the people who will always be suckered by claims of advances and developments by manufacturers.


 
  
 But this is certainly not true, judging by the reviews of the these same people.


----------



## Rob49

Has anyone listened to cd's from record label, rockcandy, that state "fully remastered sound shaped from 24bit digital technology" ?
 Are these recording's anything special, or is it just a marketing gimmick ??


----------



## skamp

rob49 said:


> marketing gimmick


----------



## Rob49

skamp said:


>


 

 .....that was perhaps my thoughts.....Rock Candy are releasing Toto's first three albums, I'm a huge fan, but I think i'll save my money & not bother buying them ?


----------



## ericr

rob49 said:


> .....that was perhaps my thoughts.....Rock Candy are releasing Toto's first three albums, I'm a huge fan, but I think i'll save my money & not bother buying them ?




Maybe buy just one to compare it to the original CD, then report back what you think. As a fan you'll know the original well enough to tell if the remaster is better, worse, or just hype.

I would appreciate knowing and think many others here would too!


----------



## headwhacker

Really takes a while to catch up on this topic. But I think, it's worth reading all the arguments. I made several tests myself (limited by my equipment and listenming skills), A very well mastered recording in hi- res format (24 bit) when downsampled properly indeed sounds identical to me ( I use SoX for downsampling files). The same recording compared to the original CD I bought years back sounded a bit different. (e.g, higher background noise and somewhat lacking in minute details).

This tells me that the method used in mastering accounts most of the difference between 16 bit and 24 bit files than the bitness and the sampling rate. Comparing the spectrum analysis chart of both hi-res file and downsampled version show the amount of data looks the same at 22Khz and below. The downsampled image show it just cuts off everything beyond that. 

I will still buy hires music but only to store my music, since remastered records nowadays are in hires format. However, everything going to my DAP will only be 16/44 or 16/48.


----------



## sonitus mirus

Google Play Music All Access (they really should do something about the name) has the Grateful Dead "Compete Studio Albums Collection" available to stream.  They only provide up to 320 kbps mp3 streams, but the remastered versions sound fantastic and definitely sound different from the original studio albums.  There are high rez versions available to purchase at 192kHz and 96kHz, both at 24 bits, although I seriously doubt there is any noticeable difference from those expensive versions and the streaming version I have access to as part of my subscription.


----------



## Otodynia

gregorio said:


> In some respects this thread doesn't help head-fiers because there is not much concrete advice. 24bit over 16bit offers no advantages but sampling rates higher than CD may, under certain circumstances, offer marginal improvement.


 
  
 Hello.  I'm still reading through this thread, and was hoping to get completely up to date before posting, but the statement above compels me to chime in.  I realize it was written in 2009, but as someone just getting into the world of high-end headphones, DAPs, and computer audio, this information is invaluable: _gregario, you've saved me from spending untold amounts of money buying high res music just for the sake of it being high res_.  I've been liberated from format chauvinism, so to speak.  I'll still buy high res if I learn that a particular recording has better mastering than the CD, or can't find it on physical media, but I'm not going to re-purchase my collection just because of the bits and sampling frequency.  
  
 To turn a popular head-fi-ism on its tail, thank you for my wallet.
  
 I also feel somewhat relieved, because so far I've been hard pressed to tell the difference between 16/44.1 and the higher resolution content, at least in terms of any night-and-day differences.  I own several versions of a favorite album, and I have to say that the best sounding of all is the K2HD-mastered version, which of course is 16-bit, 44.1-kHz.  I like it better than the 24/176 version I recently purchased online.  
  
 I don't want to derail the thread, but it seems to me -- having read only up to page 17 so far -- that the "chain of custody" of the audio data is more important than the final format.  By "chain of custody," I mean from the moment the microphone transducer sends a signal to the release of the finished product.  I've nothing to do with the world of pro audio and my ignorance is profound, but would you be able to describe -- perhaps in a new thread -- what happens (or should happen) to the audio data from recording to public release?  What are the best procedures, and where do people mess up?  When a recording is remastered, who decides who gets to do the remastering, who decides it's worthy of release?  (In my example above, who would decide what albums get the K2HD treatment?  Why those albums?  Etc.)  I'm suddenly insatiably curious about the subject, and feel like this is way more important than the bit counts and sampling frequencies...
  
 Of course, if a whole new thread is out of the question, a few helpful link recommendations would be welcomed...  
	

	
	
		
		

		
			




  
 Back to catching up...


----------



## bigshot

Now that you've had the scales pulled off your eyes when it comes to 24 bit, would you like us in Sound Science to turn you on to just as surprising truths about high bit rate lossy files, headphone amps and external DACs?


----------



## Elladan

The PONO appears to be a decent deal at its current kickstarter price of $300, and with just a few days left, I need to decide yea or nay.  

 I believe that there could be a hearable -- or "experiencable," perhaps -- difference between 16/44.1 and 24/96(192, whatever). But when people say things like "man, if you can't hear the difference between an SACD and a CD, you're deaf," I get a little frustrated.  An SACD is almost certainly going to be a remaster, right?  And during remastering, an engineer is going to be trying to achieve the best sound, using the best tech available.  I would expect a new remaster to sound better than an unremastered CD.  (Otherwise, why release it at all?)
  
 It seems to me that the test would be to take a CD encoded at 24/96 or "better" , downmix (is that the right word?) it to 16/44.1, and do a blind comparison.  That removes the remaster/remix as a factor...has anyone done this in a rigorous way?


----------



## bigshot

Yes, careful comparisons between high bitrate files and redbook bouncedowns have been done many, many times. i've done it myself. When these tests are done with sloppy controls, expectation bias intrudes. But I've never seen any careful test show any audible difference at all. That really shouldn't be surprising, because if you understand how digital audio works, it's obvious that the differences are completely inaudible.

The fact that people keep imagining that they can hear a difference tells you a lot about how sloppy people are about their thinking. Whenever I see someone talking about how inaudible frequencies make music sound better and claiming to hear things they clearly can't hear, I know that I don't really need to listen to anything else they say.


----------



## esldude

elladan said:


> The PONO appears to be a decent deal at its current kickstarter price of $300, and with just a few days left, I need to decide yea or nay.
> 
> I believe that there could be a hearable -- or "experiencable," perhaps -- difference between 16/44.1 and 24/96(192, whatever). But when people say things like "man, if you can't hear the difference between an SACD and a CD, you're deaf," I get a little frustrated.  An SACD is almost certainly going to be a remaster, right?  And during remastering, an engineer is going to be trying to achieve the best sound, using the best tech available.  I would expect a new remaster to sound better than an unremastered CD.  (Otherwise, why release it at all?)
> 
> It seems to me that the test would be to take a CD encoded at 24/96 or "better" , downmix (is that the right word?) it to 16/44.1, and do a blind comparison.  That removes the remaster/remix as a factor...has anyone done this in a rigorous way?


 

 Yes, it has been done.  E. Brad Meyer, and David Moran did this in 2007 and wrote an AES paper about it.  Using SACD, and DVD-A, they inserted a redbook A/D/A step into the middle of it, and it wasn't detected above chance levels. 
  
 http://www.drewdaniels.com/audible.pdf
  
 60 members of the Boston Audio Society were the testees.  276 correct choices out of 554 choices made.  49.82% or effectively guessing results.


----------



## ferday

esldude said:


> Yes, it has been done.  E. Brad Meyer, and David Moran did this in 2007 and wrote an AES paper about it.  Using SACD, and DVD-A, they inserted a redbook A/D/A step into the middle of it, and it wasn't detected above chance levels.
> 
> http://www.drewdaniels.com/audible.pdf
> 
> 60 members of the Boston Audio Society were the testees.  276 correct choices out of 554 choices made.  49.82% or effectively guessing results.


 
  
 thanks for posting that, hadn't seen it yet.
  
 554 trials is certainly well beyond many normal ABX tests including in the medical field.  case closed!


----------



## bigshot

Oh no! The case is never closed. Every few days someone new stumbles into the sound science forum and we start from scratch all over again!


----------



## Elladan

bigshot said:


> ...if you understand how digital audio works, it's obvious that the differences are completely inaudible.


 
 Yeah, I wouldn't claim to, though a higher sampling rate intuitively seems like it would be more accurate.  I have absolutely no doubt that there is enormous confirmation and/or selection bias in listening -- I've fooled myself, before -- so I don't really trust those who say "of course you can hear it, anyone who says you can't is an idiot."
  
 That said, it's no better to blindly trust those who flatly reject the notion of audible difference at higher bit and sampling rates when the many audio engineers who insist that they do hear a difference, especially given that those engineers typically do master in 24-bit.  
  
Could you possibly link to any published results?  I'd really like to evaluate methodology and judge for myself, and my google searches have been unenlightening.  Even a few distinctive keywords might help.

 EDIT: Sorry, I didn't see theabove post from esldude, which offered just what I wanted!


----------



## esldude

elladan said:


> Yeah, I wouldn't claim to, though a higher sampling rate intuitively seems like it would be more accurate.  I have absolutely no doubt that there is enormous confirmation and/or selection bias in listening -- I've fooled myself, before -- so I don't really trust those who say "of course you can hear it, anyone who says you can't is an idiot."
> 
> That said, it's no better to blindly trust those who flatly reject the notion of audible difference at higher bit and sampling rates when the many audio engineers who insist that they do hear a difference, especially given that those engineers typically do master in 24-bit.
> 
> ...


 

 At least you admit you didn't know how digital audio works.  And yes, higher rates, more bits seems intuitively like if fits more is better.  If you knew a little about how the digital works higher sample rates get you one thing.  Wider bandwidth.  That is all.  The extra bits don't encode more accurately than lower rates at the 20 khz and lower frequencies.  The higher rates encode higher frequencies.  24 bit vs 16 bit gets more signal to noise.  Effectively no electronics are quiet enough to reproduce at better than 20 bit resolution. 
  
 http://www.xiph.org/video/vid2.shtml
  
 If you haven't seen this video, it is well worth the 24 minutes of time to watch.  If more audiophiles watched this, there would be far less confusion about things digital.  This link has been posted before, even in this thread I believe.  Again I repeat, very much worth watching.  If you have a fuzzy idea how digital works this should be very helpful.  It is simple and easy to understand while having considerable explanatory power. 
  
 One of the good things about this video, is Monty uses some quite high quality analog signal generators, analog o-scopes and analog spectrum analyzers to show the resulting digital AD and DA conversions.


----------



## bigshot

elladan said:


> Yeah, I wouldn't claim to, though a higher sampling rate intuitively seems like it would be more accurate.  I have absolutely no doubt that there is enormous confirmation and/or selection bias in listening -- I've fooled myself, before -- so I don't really trust those who say "of course you can hear it, anyone who says you can't is an idiot."




When you understand how sound works, you realize how silly it is to claim to hear improvements with 24 bit. What if I told you that I saw protozoa crawling around on your face with my naked eye, or that I could feel the Earth spinning around the Sun? No human can detect stuff on that scale. It's the exact same thing with high bit rate audio. Human's can't hear frequencies beyond the range of human hearing, or noise at levels so low human ears can't hear it.

The resolution of high res audio in the range of human hearing is EXACTLY the same as regular old CD quality. The only advantages of hi res are BEYOND the range of human hearing.

By the way, engineers don't claim they can hear the difference in hi bit rate/sampling rate. They know there is no audible difference and bounce the mix down to redbook when they are done. The people who claim to hear a difference are audiophiles who haven't a clue how digital audio works.


----------



## Elladan

> http://www.xiph.org/video/vid2.shtml
> 
> If you haven't seen this video, it is well worth the 24 minutes of time to watch.


 
 Ha! Esldude, I actually came across that on my own through an article by the same guy, and had come back here to post it.  Fascinating stuff.
  
 Long story short, for increased quality, I should be looking for improved amplification and speakers/'phones.  But I don't want to carry an amplifier beyond Fiio E5 size, dammit!  In your opinion, what players have the best amplifier sections?


----------



## ferday

elladan said:


> Long story short, for increased quality, I should be looking for improved amplification and speakers/'phones.  But I don't want to carry an amplifier beyond Fiio E5 size, dammit!  In your opinion, what players have the best amplifier sections?




Why 'improved amplification'? A nice little fiio is cheap and well constructed, and audibly transparent.


----------



## Elladan

ferday said:


> Why 'improved amplification'? A nice little fiio is cheap and well constructed, and audibly transparent.


 
 I do have a Fiio E5, as well as a cMoy and a Meier Headsix.  The cMoy just doesn't sound good to me; also, leaving one on a plane once got the bomb squad involved.  (True story.)  The Meier sounds great, but its slightly curved case and the same security-scaring concerns prevent me from travelling with it.  I'd like a player with a really good amplifier section (and battery, obviously) so I can simplify my habit.  From this perspective, perhaps a Pono does make sense, though they're not exactly forthcoming with specs on the Kickstarter site.  The Fiio X-es seem awfully primitive, and while the iBasso D90 seems nice, the Pono seems a bit slicker, though the shape is hardly ideal.


----------



## bigshot

The simplest solution would be just to get headphones that don't require amping. Dragging an amp around for portable is a pain.


----------



## Otodynia

bigshot said:


> Now that you've had the scales pulled off your eyes when it comes to 24 bit, would you like us in Sound Science to turn you on to just as surprising truths about high bit rate lossy files, headphone amps and external DACs?




I'm sorry, was your post in response to my comment?


----------



## bigshot

Yup!


----------



## Otodynia

Well, it seems like a lot of what you mentioned is outside the scope of this thread, so I'll keep looking around with regards to the topics you mentioned.  One of my big takeaways so far, other than that 16/44.1 is "good enough," has been to look at the SNR of any prospective DAC purchase.  
  
 That said, to keep somewhat on the topic of consumer-level bit-depth, what about HDCD?  Having read this thread through from beginning to end, I noticed that someone from Linn Records stopped by and mentioned high res audio vs. HDCD, and since I don't recall a great deal of anti-HDCD sentiment being expressed at the time, does that mean HDCD might have some positive benefits?  Or has that topic already been done to death and nobody around here even bothers to roll their eyes anymore?


----------



## castleofargh

otodynia said:


> Well, it seems like a lot of what you mentioned is outside the scope of this thread, so I'll keep looking around with regards to the topics you mentioned.  One of my big takeaways so far, other than that 16/44.1 is "good enough," has been to look at the SNR of any prospective DAC purchase.


 
 SNR if given with no information on how it's done is mostly a commercial trick. from what I've seen about it, you can use several methods that won't give the most flatering results, or you can just filrter out the frequencies of shame to have those incredible results we like so much.


----------



## Otodynia

castleofargh said:


> SNR if given with no information on how it's done is mostly a commercial trick. from what I've seen about it, you can use several methods that won't give the most flatering results, or you can just filrter out the frequencies of shame to have those incredible results we like so much.


 
 I recall mention of manufacturer's using A-weighting to get better numbers; is that along the lines of what you're talking about?  Out of curiosity, I looked up the specs of a fair number of DACs last night -- from cheap to expensive -- and have to admit that the results were surprising: the Benchmark (SNR stated as 123 and 126, A-weighted and unweighted, respectively) looked far better than a premium unit that costs $22K (for one channel!).  But even so, does SNR matter?  All the music I have seems to stay far enough away from the noise floor that even SNR specs may be a bit of a red herring in terms of a useful metric.  
  
 I've been reading a thread in the hardware forums where user Gary in MD auditioned 14 different DACs and (as of thread page 60/89, which is as far as I've gotten) had a tough time finding any sonic difference between most of the contenders.  People on the sound science side will argue about the validity of his testing methodology, but since it was never intended to be a rigorous scientific evaluation I'm not going to comment.  
  
 But...
  
 Isn't the fact that he couldn't tell the difference a reasonable result?  The DAC is just supposed to take the 1s and 0s and turn them into an analog waveform, and it's not exactly a digital task that requires bleeding edge technology.  As a mechanical designer, I've had to employ the services of our test and measurement guys every so often, and I've yet to hear one of them say they've taken a new piece of equipment but can't use it until it's "burned in," or that any particular component "colors" the resulting data.  It seems like any reasonable implementation ought to get the job done, and perhaps Gary might have been better served if he's just tried to eliminate the units that sounded "bad."  
  
 Which brings me to this question: these days, is there really such as a thing as a "bad" DAC?  I mean, one that absolutely gets things wrong, that doesn't convert the bits to waveforms properly?
  
 I won't say my mind's completely made up, but if the amp section reviews well I'm probably just going to get the Oppo HA-1 when it comes out: it's a one-box solution for my _music enjoyment_ needs.  I suppose I'm not a diehard audiophile if I don't want to chase down the last 0.1% of perceived performance from my gear, and I like the fact that the Oppo will play pretty much any format thrown at it (storage is cheap, and I don't want to mess with resampling any high resolution files that might come my way).  Oppo also seems to make good quality components available for a fair price.
  
 Doesn't anyone just enjoy listening to their music without being hypercritical about the equipment used?  In the vast majority of cases, the consumer wasn't there when the music was recorded, so who are we to say that our systems are reproducing it accurately?  I'll say it again: I'm glad Gregorio started this thread, because I'll not be buying 24-bit music simply for the sake of a extending a dynamic range that was already adequate at 16/44.1.  
  
 Am I missing something?  Is there a flaw in my thinking?  I'm always happy to be educated.


----------



## bigshot

Do you know the threshold of audibility for this spec? How much is enough?


----------



## yadako

writing this up just to appreciate the very 1st thread! thanks for the info


----------



## Otodynia

bigshot said:


> Do you know the threshold of audibility for this spec? How much is enough?


 
 I'm not sure I understand your question, but I can try to explain my reasoning (which I think has been shaped by earlier discussions in this thread):
  
 If the full 96dB dynamic range of the 16/44.1 format is sufficient to blow out my eardrums, why should I worry about whether a DAC itself has a noise floor of 105 dB or 120 dB?  (I've shifted my thinking a bit since my last response to you, apparently.)
  
 Was that what you were asking?  If not, then I will ask you back: what is the audibility threshold of the spec?  Is my thinking wrong?
  
 The question from my response to castleofargh still stands, though: is there really such a thing as a "bad" DAC these days?  I'm not talking about the more esoteric botique brands (since I'm not ever going to spend that much money), but it seems that the larger, established companies can hire people who really know what they're doing when it comes to the digital-to-analog conversion process.  Microscopic evaluation of the output waveforms might reveal this or that, but am I going to really _hear_ the difference?  Even if I can, how am I, the consumer, supposed to know which microscopic difference is right and which is wrong?


----------



## bigshot

That's exactly what I was driving at.

And an even better question than your "bad DAC" question is, are most players audibly transparent even without an external DAC?


----------



## Otodynia

bigshot said:


> That's exactly what I was driving at.
> 
> And an even better question than your "bad DAC" question is, are most players audibly transparent even without an external DAC?


 
 That's a question that my limited knowledge can't answer.  What's your opinion?  
  
 Having thought about why I was going to look at DAC SNR specs, I can say that I was using at it more from the standpoint of "engineering quality" (i.e., whether the manufacturer did a good job or not) rather than "sound quality."  
  
 Really, though, it just doesn't seem to matter that much.  At this point, I'm more inclined to just compare features (how many formats are decoded, does it have a USB input, etc.) to determine if a DAC/DAP/streamer will meet my needs rather than nitpick the technical specs.  Is that misguided?
  
 Or, if you'll tolerate another question, if _you_ were shopping for a DAC, what would _you_ look for?


----------



## bigshot

Just about all solid state electronics (DAPs, CD/DVD/Blu players, DACs / amps) are audibly transparent. This means they all sound the same. There are differences in the way they measure, but who cares if you're listening to music?

Do you need a high end DAC? Do you need 24 bit? Do you need to spend a lot of money? Nope.

If I was shopping for a DAC, I would look for a player that didn't require an external DAC. The same goes for amps. I's look for headphones that didn't require amping. I'd take the money I saved and spend it on better headphones.


----------



## cjl

Many (but certainly not all) portable digital players are audibly transparent, including many cell phones. There are a couple of common problems to look for though. The two I have personally noticed with some players are an audible noise floor and significant output impedance. An audible noise floor is easy to detect - play a silent audio file, or an audio file with a silent passage at the maximum volume you would ever listen to the player using your most sensitive set of headphones. If you can't hear any noise or hiss, it isn't a problem. As for the output impedance, that can be harder to find. With some headphones, this isn't an issue (dynamic IEMs don't tend to have a problem with this, for example), but with other headphones, it can be a significant problem. With balanced armature IEMs, a high output impedance can cause significant distortions in the frequency response, due to the significant variation in the impedance of the IEM with frequency, and with some low-impedance headphones that use electrical damping to control the driver at low frequencies (for example the older Denon AH-Dx000 series), a high output impedance can cause the bass to become boomy and distorted.
  
 Unfortunately, unless you have equipment like an oscilloscope or at least a good AC voltmeter, you have to detect this problem either by ear (by comparing its output to a known good source) or by googling your player and hoping someone else has measured the output impedance. If you do have access to a good AC voltmeter or scope though, you can put a test signal through your device and measure the output voltage with no load. Then you can put a 16 ohm resistor (or similar) across the output of the device and measure the voltage again. If it has a very low output impedance, the level should be basically identical for both the unloaded and loaded case. If your device has a 16 ohm output impedance though (as an example), it will put out half as much voltage with a 16 ohm resistor across its output as it will when unloaded, and this could cause audible issues with some (but not all) headphones.


----------



## bigshot

Or just get headphones where impedance isn't an issue.


----------



## cjl

bigshot said:


> Or just get headphones where impedance isn't an issue.


 
 Sure, except that I haven't found ones I like as much as my D5000s, and they have an issue with a high source impedance. Besides, any competently designed headphone output should have a low impedance (but sadly, many aren't competently designed). I end up just running an O2 amp/USB dac most of the time, since I primarily listen to headphones at work, and my work computer has the most pitiful headphone output you've ever heard (full volume is rather quiet with my D5ks, which aren't exactly power hogs, and it has a fair amount of hiss too). The O2 has an inaudible amount of noise and an output impedance of significantly below 1 ohm, so for my purposes, it's audibly perfect.


----------



## bigshot

Amping isn't as much of an inconvenience for a home or office system I suppose. Worse for a portable system.


----------



## castleofargh

otodynia said:


> castleofargh said:
> 
> 
> > SNR if given with no information on how it's done is mostly a commercial trick. from what I've seen about it, you can use several methods that won't give the most flatering results, or you can just filrter out the frequencies of shame to have those incredible results we like so much.
> ...


 

 I said that because you were going from 16/24bit to SNR, my point being that usually however good the specs read, any audible noise will be because of the sound system, not because it's "only16bit" and the noise floor is higher than 24bit.
  
  
 does SNR matter? I would be tempted to say that it goes as a pack, if the sound system ends up with ungly distortions only 50db under the music, it will probably not matter much that the SNR is only -80db.
  
 to me noise values that matters are those I can hear. that's all. I listen to music very very quietly(most of the time) in a very quiet environment on very sensitive IEMs, so what reads on graphs: music @ 0db and noise @ -70db, might end up being me listening @ -30db and noise only 40db under my music if it's not volume dependent.
 I've tried to link values from dacs or amp to the actual hiss I can sometimes get, and I can't seem to find a relation between them, so I have concluded that manufacturer specs are often phony, or that those specs didn't tell about something I can hear.
 the only thing I could relate to, came from http://www.markuskraus.com/RMAA/rmaa%20complete%20-%20html.html as he makes the measurements from the IEM, it gives the actual idea of what is happening. so those measurements gave me actual usable intels about hiss and showed what I heard on the daps I've owned. not like all noises are actual hiss, but it's the only one that matters to me if it's the only one I can hear ^_^.
 sorry if it's a weird answer, but that's all I've got.
  
 about DACs, I believe that some are better than others in an audible way. but at the same time, I believe that a bad dap already does an OK job.
 a bad headphone is bad, a bad amp can be pretty messy(bad or not with the intended kind of headphones), a bad dac is making ok sound. worst case is usually a smaller soundstage, some kind of roll off due to bad filtering? even with a poor hifimediy I didn't feel like the sound was bad. I preferred the odac or even my daps with a line out, but it wasn't bad. sound differences were audible but not really obvious, and it was more "different" than better or worst. telling who had the truth would have been a lot less obvious if I didn't know one was a cheap little usb crap. I really believe you could trick a few guys in a double blind test with different dacs.
 now that's just my opinion based on my still small experience. I've read some stuff that would suggest a lot of ways a dac could go so very slightly wrong. like the fact that the moment the dac is triggered by voltage entering (signal=1) doesn't always start exactly the moment the tension start going through. something about taking the point with maximum voltage as the trigger, and the signal overshooting or undershooting at some places along the duration of the signal 1.  possibly creating some matter of jitter.
  
 for most questions, I feel like someone like ClieOS would be better fitted than humble nooby me.


----------



## bigshot

Sounds to me like you've got a piece of equipment in your chain that isn't working properly. None of that makes any sense.


----------



## cjl

bigshot said:


> Amping isn't as much of an inconvenience for a home or office system I suppose. Worse for a portable system.


 
 True, and the rare times I use a portable systems, I'm using my Senn IE80s, which don't seem to care about source impedance much (so I drive them straight out of my MP3 player without any problem at all). When I'm at home, I tend to use speakers rather than headphones, since they sound so much fuller and more natural than any headphones I've tried. Because of this, the only time I'm left listening to headphones that care at all what is driving them is at work, where having one extra little box sitting on my desk is no big deal (and if anything, I prefer the physical volume knob on the O2 to software volume control, so even from a convenience standpoint, it is an improvement).


----------



## castleofargh

bigshot said:


> Sounds to me like you've got a piece of equipment in your chain that isn't working properly. None of that makes any sense.


 
 I can't let you say that !!!!!
 at least I got that part right


> for most questions, I feel like someone like ClieOS would be better fitted than humble nooby me.


----------



## bbmiller

*Irrespective of there being any reason for recordings with a 24-bit depth to be better than ones with a 16-bit depth, do you think they are because the producers/audio engineers know when they produce a recording with 24-bit depth they are catering to people who care about sound and thus create better mixes/audio engineering for people who care about better sound?*


----------



## esldude

bbmiller said:


> *Irrespective of there being any reason for recordings with a 24-bit depth to be better than ones with a 16-bit depth, do you think they are because the producers/audio engineers know when they produce a recording with 24-bit depth they are catering to people who care about sound and thus create better mixes/audio engineering for people who care about better sound?*


 

 Could be.  Remasterings are at least half the time much better.  Then again, I always did the best job I could in my career regardless of whom it was for.  I was grateful for those times I was allowed to go all out with the resources I needed, and do understand even if that is your desire your employer may not allow it.  Just to be clear my career was not in the music industry.


----------



## headwhacker

I thought historically digital recording has always been done at 24-bit. Then after mastering, mixes and everything else are done it is re-sampled to 16 bit  for CD release.


----------



## bigshot

If I remember, 24 bit started around 2000. Maybe a bit before. It was all pretty much 16 bit before that.


----------



## bigshot

bbmiller said:


> *Irrespective of there being any reason for recordings with a 24-bit depth to be better than ones with a 16-bit depth, do you think they are because the producers/audio engineers know when they produce a recording with 24-bit depth they are catering to people who care about sound and thus create better mixes/audio engineering for people who care about better sound?*


 
  
 24 bit allows for more flexibility in the mix. If you want to boost or compress something as you mix, you can do it without the noise floor coming up with it.


----------



## 7ryder

lots of information on 16 bit vs 24 bit and whether or not a remastered album recorded prior to the early 80's can be considered hi-res if it was recorded via analog tape (hint, they aren't) can be found here: http://www.realhd-audio.com/


----------



## cjl

bigshot said:


> 24 bit allows for more flexibility in the mix. If you want to boost or compress something as you mix, you can do it without the noise floor coming up with it.


 
 It also allows for performances to be recorded farther below 0dBFS, allowing for more headroom (and still maintaining an inaudible noise floor), in case the performer hits substantially higher levels in the performance than they did in the sound check.


----------



## Captain Duck

Ah, but there is a (small) problem. Good analog disk mastering equipment effectively has more than 96db of dynamic headroom.
  
 First example of hardware: the Neumann disk mastering console of the 1980's had a s/n of about 114db. The difference of it sonically in a fade-out was demonstrated by Bob Ludwig at a NY AES chapter meeting in the late 80's. He did a fade on the Neumann console and then from the Sony PCM 1630. The digital playback got quieter (and grainier) and then POOF!!. All gone. Suddenly. The analog fade continued until it was awash in the noise floor.
  
 Next is the problem of how s/n is measured vs. the way we hear.  Noise floor measurements are done either 'A' or 'C' ( or 'B' too) weighted depending on whether flat or LF rolled off response is desirable for the purpose. Either way, you are calculating the sum of 10 octaves. Music, especially quiet music, has a much narrower bandwidth. In analog what happens is that you can get a n/s ratio where the broadband noise is of a higher value than the instrument(s) playing, but their output is above the noise floor over their smaller bandwidth.
  
 This is what Mr. Ludwig so elegantly demonstrated.


----------



## cjl

captain duck said:


> Ah, but there is a (small) problem. Good analog disk mastering equipment effectively has more than 96db of dynamic headroom.
> 
> First example of hardware: the Neumann disk mastering console of the 1980's had a s/n of about 114db. The difference of it sonically in a fade-out was demonstrated by Bob Ludwig at a NY AES chapter meeting in the late 80's. He did a fade on the Neumann console and then from the Sony PCM 1630. The digital playback got quieter (and grainier) and then POOF!!. All gone. Suddenly. The analog fade continued until it was awash in the noise floor.
> 
> ...


 
 16 bit digital audio (interestingly enough) also has a s/n in narrow frequency bands of about 110-120dB (or even more) with proper dither. You can also encode a waveform (with dithering) that has an amplitude that is less than half of your least significant bit (LSB). Because of this, with dither, you would not get that sudden dropout in the fade. Instead, you'd get the exact effect you described with the analog system: it would continue to get quieter until it is lost in the noise.
  
 In addition, if you can even hear the "dropout point" at all on a non-dithered 16 bit digital system, that means that a signal with an amplitude of 1 LSB would have to be around 20-30dB in any normal room. That means that a full amplitude signal (0dBFS) would be upwards of 110-120dB, which is painfully, eardrum-damagingly loud. If you have the volume turned up this loud on your system, chances are you won't hear the dropout at the end of the fade anyways, since your ears will still be ringing from the volume of the peaks. Sure, you can intentionally record something 40dB down from full scale, and then you do have a problem, but the fix is simple: record closer to the full scale level.
  
 I know this video has been posted a lot here recently, but it is worth watching, especially the part about dither starting around 12 or 13 minutes in. Notice how the noise floor is around -120dB when he enables dither with a 16 bit signal, and he can encode a 1/4 bit amplitude sine wave perfectly fine (-103dBFS or so): http://xiph.org/video/vid2.shtml


----------



## Captain Duck

Hadn't looked at it that way (narrow band, encoding with dither). I doubt that the Sony PCM-1630 did anything more than straight encoding. Good video, too.


----------



## Notus

First i have to say that i am noob in all related to audio, at least high end audio. I have never understood the reasoning behind 24bit / 96khz for audio listening.
 I read at some point that CD supports only 16bit / 44.1khz and its from 1hz - 22khz the limits of human hearing. As far as i know 99% of all music in the world is CD quality.
 I think there are physical limitations to the existing recordings? Would re-sampling the masters do anything to them sound wise?
  
 I have always felt the faults in sound quality is not because of the digital ( cd quality ) limitations, but in the mixing and mastering of the actual music. Also the Hardware used to record the music. I have heard a lot of music in my life, majority of music is poorly recorded, poorly mixed and i don't think that increasing the sample rates will fix that. At least that's how i feel.
 I have some great recordings and they sound great even as MP3, but poorly recorded, mixed / mastered ones sound bad no matter what you do to them. I don't think any amount of pixie dust is going to make them better. This is a huge threat and i managed to only read the first 22 pages, barely understanding what people were saying. Hard to follow when you don't have deeper understanding on the subject.
  
 Lets say that in theory the 24bit / 96khz would have better sound quality. Ok it becomes the "standard" and we start to demand music in that form. All the old music would gain nothing from this as they have the physical limitations? Newer produced music and recorded music would only "gain" in sound quality. 99% of our music would still be physically limited to the quality of "16bit / 44.1khz". I mean the physical limitations of hardware and the existing recordings we currently have. Vinyl is bellow CD in quality, at least according to some.
 Its possible to re-master the older recordings that are in 24bit / 96khz form. But they have been recorded with limited gear (frequency limited to 20khz)?
 So what does the consumer gain from the higher sample rates?
  
 The downsides are far greater, we would have to buy all the music in DVD all over again with no evidence to support it sounds any better (at least nothing they could prove with empirical evidence). Also the file size would grow quite a lot. I am not sure how much but its possible it would be near 500mb for 1 song or even higher?
  
 Since the release of CD and digital music piracy and downloading songs from internet has become a real bane for the music business.
 When i write this i get the feeling the reason behind pressing the 24bit / 96khz onto the market is to try and kill piracy. At least thats what popped into my mind.
  
 Sorry if this came out as just some noob ramble that makes no sense?


----------



## MarcadoStalker7

Head-Fi sucks


----------



## bigshot

No audible difference between the two.


----------



## MarcadoStalker7

Head-Fi sucks


----------



## bigshot

That or dog brains!


----------



## kh600rr

Save


----------



## bbmiller

bigshot said:


> No audible difference between the two.


 
 Is it necessary to have a DAC with 24-bit capability even though there is no audible difference between 16 bits and 24 bits just to decode 24-bit's most effectively?


----------



## castleofargh

bbmiller said:


> bigshot said:
> 
> 
> > No audible difference between the two.
> ...


 

 if your dac handles only 16bit, then the source(computer?) will have to go back to 16bit before sending data. seeing how it works, I guess it's only cutting the end of each sample. it shouldn't do anything to the first 96db of music.


----------



## ab initio

bbmiller said:


> Is it necessary to have a DAC with 24-bit capability even though there is no audible difference between 16 bits and 24 bits just to decode 24-bit's most effectively?




Having a 24bit DAC is much more important than having 24bit recordings. I say this because having 24 bit output (effectively 20-22) gives you a lot of headroom to use software volume control without any loss of sound quality, nor noticeable increase in noise. This is true regardless if youve got 24bit or 16bit recordings

Cheers


----------



## skamp

ab initio said:


> (effectively 20-22)




20-22 bits is really high end hardware (and I'm talking about real high end measurable stuff like the Benchmark DAC1/DAC2, not snake oil hardware). With good consumer hardware, it's more like 18-20 bits.


----------



## ab initio

skamp said:


> 20-22 bits is really high end hardware (and I'm talking about real high end measurable stuff like the Benchmark DAC1/DAC2, not snake oil hardware). With good consumer hardware, it's more like 18-20 bits.




Either way, every additional Effective Number of bits in the output hardware allows an additional 6dB of headroom for software volume adjustment without detriment from the noise floor

Cheers


----------



## Dark_wizzie

ab initio said:


> Having a 24bit DAC is much more important than having 24bit recordings. I say this because having 24 bit output (effectively 20-22) gives you a lot of headroom to use software volume control without any loss of sound quality, nor noticeable increase in noise. This is true regardless if youve got 24bit or 16bit recordings
> 
> Cheers


 
 I turn volume to max in software and adjust volume via volume knob on my dac/amp combo.


----------



## cjl

dark_wizzie said:


> I turn volume to max in software and adjust volume via volume knob on my dac/amp combo.


 
 That works fine, but if you have relatively sensitive headphones and a fairly high-output dac/amp, that could put you in the low volume range of the volume pot, where there can be channel imbalance issues. This can be avoided by turning down the software volume and increasing volume on the amp, but with a 16 bit dac, this can cause an audible noise floor. With a 24 bit dac, there is no problem with doing this (unless you use a huge amount of software attenuation).


----------



## ab initio

dark_wizzie said:


> I turn volume to max in software and adjust volume via volume knob on my dac/amp comjbo.




That's fine but it doesn't matter (see here). Like cjl said, as long as you keep the noise floor down, you can set set software volume to whatever is convenient for you/ your amp/ your headphones/etc.

Cheers


----------



## Dark_wizzie

ab initio said:


> That's fine but it doesn't matter (see here). Like cjl said, as long as you keep the noise floor down, you can set set software volume to whatever is convenient for you/ your amp/ your headphones/etc.
> 
> Cheers


 
  
  


cjl said:


> That works fine, but if you have relatively sensitive headphones and a fairly high-output dac/amp, that could put you in the low volume range of the volume pot, where there can be channel imbalance issues. This can be avoided by turning down the software volume and increasing volume on the amp, but with a 16 bit dac, this can cause an audible noise floor. With a 24 bit dac, there is no problem with doing this (unless you use a huge amount of software attenuation).


 
 I don't hear any noise until I turn gain on and crank up volume to 50-100%. It's inaudible without gain no matter the volume. And I only need 15-40% volume depending on what I'm doing. So I should be getting all 16 bits, right?


----------



## cjl

dark_wizzie said:


> I don't hear any noise until I turn gain on and crank up volume to 50-100%. It's inaudible without gain no matter the volume. And I only need 15-40% volume depending on what I'm doing. So I should be getting all 16 bits, right?


 
 Yes, though if the 15-40% is the volume range on your amp, you could get channel imbalance issues at the low end of that range (as I mentioned before). Many (if not most) volume pots have channel imbalance issues at the very low end of their range, so you can sometimes get substantially better performance by setting software volume level to ~50% or so and using the 30-80% range on your volume pot rather than setting software to 100% and using the low end of the volume pot. In addition, most DACs have slightly increased distortion when fed a 0dBFS signal, so some software volume attenuation prevents this as well.
  
 Of course, the chances are that all of the flaws I just mentioned are near-inaudible anyways, so it doesn't really matter much. Out of curiosity (maybe you already mentioned and I didn't notice), what amp/dac are you using?


----------



## Dark_wizzie

cjl said:


> Yes, though if the 15-40% is the volume range on your amp, you could get channel imbalance issues at the low end of that range (as I mentioned before). Many (if not most) volume pots have channel imbalance issues at the very low end of their range, so you can sometimes get substantially better performance by setting software volume level to ~50% or so and using the 30-80% range on your volume pot rather than setting software to 100% and using the low end of the volume pot. In addition, most DACs have slightly increased distortion when fed a 0dBFS signal, so some software volume attenuation prevents this as well.
> 
> Of course, the chances are that all of the flaws I just mentioned are near-inaudible anyways, so it doesn't really matter much. Out of curiosity (maybe you already mentioned and I didn't notice), what amp/dac are you using?


 
 O2/Dac.
 I don't notice any channel imbalance issues and didn't know it was a thing until a few days ago when I read about it.
  
 Although I think the Objective has different gain settings, one for default and another for 'extra gain' button. So in theory if we decrease the gain even further, lower than default without the gain switch, I would turn the volume knob up a little bit to decrease channel imbalance and prevent higher noise from having higher gain. But then again, neither of these issues are noticeable to my ears, and having sound play at 50% volume with gain switch on is ear suicide whether by speakers or headphones. But I can actively look for channel imbalance issues, lol.


----------



## cjl

dark_wizzie said:


> O2/Dac.
> I don't notice any channel imbalance issues and didn't know it was a thing until a few days ago when I read about it.
> 
> Although I think the Objective has different gain settings, one for default and another for 'extra gain' button. So in theory if we decrease the gain even further, lower than default without the gain switch, I would turn the volume knob up a little bit to decrease channel imbalance and prevent higher noise from having higher gain. But then again, neither of these issues are noticeable to my ears, and having sound play at 50% volume with gain switch on is ear suicide whether by speakers or headphones. But I can actively look for channel imbalance issues, lol.


 
 I would definitely leave it in low gain unless you're driving some hard-to-drive headphones that really require the high gain level. If you don't notice any channel imbalance though, then you should be good to go. I use an O2/ODAC as well, and I do notice a slight channel imbalance (right channel is a bit louder) at extremely low volume levels (around the 8-o-clock position on the dial), but I have my software volume set so that my listening level is always higher than that on the analog dial. Usually, I leave my windows volume control around 50%, my volume control on my O2 at around 50%, and my gain switch on low gain.


----------



## Dark_wizzie

cjl said:


> I would definitely leave it in low gain unless you're driving some hard-to-drive headphones that really require the high gain level. If you don't notice any channel imbalance though, then you should be good to go. I use an O2/ODAC as well, and I do notice a slight channel imbalance (right channel is a bit louder) at extremely low volume levels (around the 8-o-clock position on the dial), but I have my software volume set so that my listening level is always higher than that on the analog dial. Usually, I leave my windows volume control around 50%, my volume control on my O2 at around 50%, and my gain switch on low gain.


 
 Can't really change the default gain now though. I'd have to open it up and stuff.
 I still can't hear channel imbalance at 7-8oclock position.


----------



## cjl

dark_wizzie said:


> Can't really change the default gain now though. I'd have to open it up and stuff.
> I still can't hear channel imbalance at 7-8oclock position.


 
 I'm not saying change the default gain - just leave the gain switch on the low gain setting. No real reason to go monkeying around with the internals if there's nothing wrong with the amp. As for the lack of channel imbalance - maybe you got a volume pot that happens to be really good, even at low levels. Not all volume pots have channel imbalance issues at low gain (and if you got a good one, that means there's even less reason to go mess with the internal gain settings of the amp).


----------



## Dark_wizzie

cjl said:


> I'm not saying change the default gain - just leave the gain switch on the low gain setting. No real reason to go monkeying around with the internals if there's nothing wrong with the amp.


 
 Yup, I don't touch the gain button.


----------



## Logsah

cjl said:


> Usually, I leave my windows volume control around 50%, my volume control on my O2 at around 50%, and my gain switch on low gain.


 
  
  
 Jesus, I usually have Win vol 100% / iBasso Boa 80% & high gain on..........listening to Dubstep!!!  
	

	
	
		
		

		
			




  
 I'm going to die aren't I lol?
  
  
 Ok clearly I need to re-look at my listening levels doh!


----------



## cjl

What headphones do you have? You might just have much, much less sensitive headphones than I do (which is very plausible, since I use Denon D5000s, which are fairly low impedance and definitely on the more sensitive side as audiophile headphones go).
  
 EDIT: Oh, and you're also using a completely different amp. 80% on one amp is completely different from 80% on another amp, so you could be listening at the same level as I am for all we can tell from the information given. Looking at the specs on the Boa, it doesn't look like it has nearly as much power as the O2, so you aren't listening nearly as loud as you would be on an O2 with the high gain setting on at 80% (which would be pretty loud on the vast majority of headphones out there).


----------



## Logsah

cjl said:


> What headphones do you have? You might just have much, much less sensitive headphones than I do (which is very plausible, since I use Denon D5000s, which are fairly low impedance and definitely on the more sensitive side as audiophile headphones go).
> 
> EDIT: Oh, and you're also using a completely different amp. 80% on one amp is completely different from 80% on another amp, so you could be listening at the same level as I am for all we can tell from the information given. Looking at the specs on the Boa, it doesn't look like it has nearly as much power as the O2, so you aren't listening nearly as loud as you would be on an O2 with the high gain setting on at 80% (which would be pretty loud on the vast majority of headphones out there).


 
  
 Audio Technica ES700 with *Forza* AudioWorks re-cable & ESW9 pads. [impedance 36 ohms]
 +
 iBasso D2+ Boa Amp/Dac / Output power：Up to 100mW+100mW into 32Ω / Gain: +3dB/ +10dB (AMP) 
  
  
  
 Aww no dude I'm deffo listening to my music way too loud (raver child of the 90's here lol) 
	

	
	
		
		

		
		
	


	



  
 For example before I had the V-Moda M80's and with everything turned up 100% and at 24 seconds into
 this song for example the 'Pop or Bok' sound would feel like a tiny knome with a hammer inside my head hitting
 me behind the bridge of the nose/eyes area lol 
	

	
	
		
		

		
		
	


	




 bad headaches afterwards etc.
  

  
  
 The Youtube version doesn't quite go loud enough but there's a zippy download if you wanna
 temporarily listen to it thru your player of choice with EQ etc. http://www49.zippyshare.com/v/42536243/file.html
  
 I used Foobar with the 'Punch & Sparkle' EQ preset found here:
 http://www.sevenforums.com/sound-audio/2477-foobar-2000-equalizer-presets.html
  
  
 Any advice welcome'd yall 
	

	
	
		
		

		
		
	


	



  
 (sry for mini thread hijack)


----------



## Dark_wizzie

You use HD800 with O2 at max volume and gain, your ears will liquify.


----------



## cjl

Well, those will be about 6dB quieter at the same voltage level than my D5000s, and as I already said, the O2 can output significantly more power than the Boa from what I can find. The Boa claims 125mW into 16 ohms, and assuming that's voltage limited, that means it can swing about 1.4V RMS at full power (and this is probably underestimating a bit - especially if your stat of 100mW into 32 ohms is accurate, since that's more like a 1.7V RMS signal). Into the ES700s, that would be 120dB for a 0dBFS signal at full power. However, at 80% volume and (probably) not a full scale input signal to the amp from the dac, you might be down at more like 100-105dB. That's still very, very loud, and potentially damaging to your hearing.
  
 For comparison though, the O2, which is what I use, can swing about 4V into 25 ohms (my D5000s), which would be a full-power level of 134dB (!!!). It can also push  around 6V into 56 ohms (your ES700s), which is about 132dB. You would never want to listen this loud - it would almost immediately permanently damage your hearing (and possibly blow your headphones).
  
 (For what it's worth, my D5000s are fairly loud on my O2 at 50% volume and low gain, with 50% software volume and playing that youtube video as the source. They are VERY loud (uncomfortably so in parts) with the O2 on high gain, still 50% volume both on the knob and in software I can't imagine turning it up to full software volume/80% amp volume on high gain, but as I said, that would be quite a bit louder on my setup than on yours)


----------



## cjl

dark_wizzie said:


> You use HD800 with O2 at max volume and gain, your ears will liquify.


 
 I would believe it - the O2 could push a bit over 7V RMS into the HD800, which is the better part of 200mW and 125dB.


----------



## Dark_wizzie

Yeah, even running the volume knob to halfway without gain is going to damage my ears. I've seen some graphs showing the HD800 having impedence shoot up to ~640 ohms at 100hz. 
 http://www.innerfidelity.com/images/SennheiserHD800B.pdf
  
 But O2 can drive that?


----------



## Logsah

cjl said:


> Into the ES700s, that would be *120dB for a 0dBFS signal at full power*.
> However, at *80% volume* and (probably) not a full scale input signal to the amp from the dac, *you might be down at more like 100-105dB*.


 
  
 Good man cheers, yup I'm still too loud but these are figures I can work it out from


----------



## SilverEars

dark_wizzie said:


> Yeah, even running the volume knob to halfway without gain is going to damage my ears. I've seen some graphs showing the HD800 having impedence shoot up to ~640 ohms at 100hz.
> http://www.innerfidelity.com/images/SennheiserHD800B.pdf
> 
> But O2 can drive that?


 
 True, although the specs say 300ohms, it's not true.  It varies and can shoot up to 600ohms.  You need to take care of that as the worse case scenario as it will clip if there isn't enough.  When I used O2 with it, it didn't sound right, not sure if it clipped.  I know my other amp it clipped, and it drives my 650 fine, and it can have something to do with the varying impedance as the 650's impedance peak is less than the 800's.


----------



## Dark_wizzie

Here is the impedence graph for the HD800 (from Innerfidelity):
 http://www.innerfidelity.com/images/SennheiserHD800.pdf
  
 The guy from NW talk about O2 seems like it is designed to be able to run Beyer DT880 600ohm version as worst case scenario.
  


> *THD+N vs OUTPUT & MAX POWER ON AC:* At 1 Khz with both channels driven here’s the distortion versus output on AC power into 15, 33, 80, 150 and 600 ohms. At 150 & 600 Ohms the output voltage was essentially the same at about 7.3 volts RMS. And even at about 200 mW of output into any of the loads the distortion is still below about 0.0025%! Maximum power is about 640 mW at 80 ohms. The power limits shown below exceed the power requirements established for the assumed worst case headphones (HiFiMan planars and 600 ohm version of the Beyer DT880):


 
 The Hifiman planars... they don't require a lot of ohms, they just have low sensitivity. But the DT880 600ohm graph here 
 http://www.innerfidelity.com/images/BeyerdynamicDT880600ohm.pdf
  
 shows that the DT880 is definitely harder to drive than HD800.
 The thing is though, that the impedence requirement spike to 640 ish? is at 100hz, which is a bass frequency. I'm just wondering if this can have an effect on the bass. One guy on headfi replied saying, maybe it's the rapid changes in required impedance that could cause problems? But I have no idea whether his idea has an merit.
  


> *CLIPPING PERFORMANCE:* Some amps become unstable when pushed to clipping for many reasons. Some op amps, for example, are prone to phase reversal when clipped where the output violently slams into the _opposite_supply rail. Other amps exhibit ultrasonic oscillation when clipped. The O2 is completely clean into any load I tried and also exhibits very close to symmetrical clipping. This is one of those tests everyone should always run, and not just with a soundcard “scope”, so you can see any ultrasonic/RF problems. Here the O2 hits +/- 20 volts peak-to-peak at 10 Khz into 600 ohms on AC power on a 100 Mhz scope:


 
 Based on what the guy from NW says I think he believes the O2 can drive regular 600ohm headphones, so a HD800 should really be covered no matter the frequency with no problems at all. 
  
 Also I do not know of any other impedance graph for HD800s, neither do I know how accurate InnerFidelity's measurements are.
 Personal experience with O2 is that the O2 can drive the HD800 effortlessly. But seeing the huge variation in impedance from frequency to frequency it's a little hard to know how the 100hz section fares but I don't think it's bad enough to be a problem, in my uneducated opinion.


----------



## cjl

silverears said:


> True, although the specs say 300ohms, it's not true.  It varies and can shoot up to 600ohms.  You need to take care of that as the worse case scenario as it will clip if there isn't enough.  When I used O2 with it, it didn't sound right, not sure if it clipped.  I know my other amp it clipped, and it drives my 650 fine, and it can have something to do with the varying impedance as the 650's impedance peak is less than the 800's.


 
 As far as an amp is concerned, a higher impedance load is an easier load. This is also why pretty much every preamp out there can drive a 10kOhm input impedance into an amp easily, but you have to have some pretty serious professional (or high end) amplifier to push any kind of substantial voltage swing into a 2 ohm loudspeaker. Since the amp is trying to act as a voltage source, it isn't going to clip if the impedance of the headphones is higher in a particular area - it just doesn't have to source as much current at those frequencies for the same voltage swing. The O2 can easily drive the HD800 well past any volume you'd ever want to listen at, and it will do so with zero audible distortion and perfect frequency response. As I mentioned earlier, the O2 can push around 7V RMS into a high-impedance load like the HD800 (and if you look at the designer's graph, you might notice that its voltage output vs THD plots look almost identical when driving an 80 ohm, a 150 ohm, and a 600 ohm load, and its THD drops and peak output voltage increases for higher impedance loads). I know people in high-end audio like to believe that a $150 amp couldn't possibly drive a $1500 pair of headphones correctly, but the measurements disagree in this case. I will admit though that my prior calculation is wrong - I forgot that Sennheiser specifies sensitivity in dB/1VRMS, not dB/mW like everyone else, so a 7.3V signal would actually only push them to 119dB, not 125 as I stated above. Of course, 119dB is still way more than enough.
  
 As for the HD800 vs HD650? Assuming the manufacturer's specified sensitivities are correct, there really shouldn't be much difference between the difficulty of driving the two of them. Sure, the HD650 is a bit more sensitive, so it should be a bit easier to drive, but the difference should be on the order of a dB or two. Unless you were already really, really borderline with the 650, the 800 shouldn't be a problem for any amp that can drive the 650. How are you determining that it is clipping, out of curiosity?
  
 Finally, as far as the amp is concerned, the most difficult load to drive well is actually a low impedance, low sensitivity headphone, such as the HE-6. To drive those properly, you need a large voltage swing and a LOT of current, and that is where you could well see a problem with the O2 distorting. It does just fine on the LCD-XC in my opinion, which is the hardest headphone to drive that I have ever tried my O2 with. The HE-6 is even harder to drive though, so I could envision it having some problems there. I don't have access to a pair to try them out, so this is purely theoretical. Actually, in many ways a moderate sensitivity, high-impedance headphone is the easiest to drive (which is probably part of why many professional monitoring headphones fall into this category). A super sensitive, low-impedance design like an IEM is really sensitive to noise in the amplifier circuit (which can be hard to eliminate entirely), a low impedance, low sensitivity design requires a lot of current (which can be hard to source without distortion), but a moderate sensitivity, high impedance design is not going to have a noise problem (unless you have a truly appalling amount of noise in your electronics), is not going to require much current to drive (which means things like output impedance and trace resistance on the board become unimportant), and the only real downside is that it takes a bigger voltage swing (and +/- 20V or so is easy to do when you have access to an AC power supply - it only really is a problem if you have to run off batteries in a portable player or something like that). So, if you know you'll pretty much always be using a desktop amp with high voltage power sources available, the high impedance, moderate sensitivity design is kind of a no brainer. I don't know where people got this idea that high impedance = hard to drive, but it's really not even close to correct (sadly, otherwise I'd be running my 8 ohm speakers off a tiny little headphone amp 
	

	
	
		
		

		
		
	


	




)


----------



## Dark_wizzie

cjl said:


> As far as an amp is concerned, a higher impedance load is an easier load. This is also why pretty much every preamp out there can drive a 10kOhm input impedance into an amp easily, but you have to have some pretty serious professional (or high end) amplifier to push any kind of substantial voltage swing into a 2 ohm loudspeaker. Since the amp is trying to act as a voltage source, it isn't going to clip if the impedance of the headphones is higher in a particular area - it just doesn't have to source as much current at those frequencies for the same voltage swing. The O2 can easily drive the HD800 well past any volume you'd ever want to listen at, and it will do so with zero audible distortion and perfect frequency response. As I mentioned earlier, the O2 can push around 7V RMS into a high-impedance load like the HD800 (and if you look at the designer's graph, you might notice that its voltage output vs THD plots look almost identical when driving an 80 ohm, a 150 ohm, and a 600 ohm load, and its THD drops and peak output voltage increases for higher impedance loads). I know people in high-end audio like to believe that a $150 amp couldn't possibly drive a $1500 pair of headphones correctly, but the measurements disagree in this case. I will admit though that my prior calculation is wrong - I forgot that Sennheiser specifies sensitivity in dB/1VRMS, not dB/mW like everyone else, so a 7.3V signal would actually only push them to 119dB, not 125 as I stated above. Of course, 119dB is still way more than enough.
> 
> As for the HD800 vs HD650? Assuming the manufacturer's specified sensitivities are correct, there really shouldn't be much difference between the difficulty of driving the two of them. Sure, the HD650 is a bit more sensitive, so it should be a bit easier to drive, but the difference should be on the order of a dB or two. Unless you were already really, really borderline with the 650, the 800 shouldn't be a problem for any amp that can drive the 650. How are you determining that it is clipping, out of curiosity?
> 
> ...


 
 I am inclined to agree with you - What you say seems to correlate with NW person's writings.
 How hard are LCD3s or HE500s to drive, I'm sure they are easier than HE6.
  
 If higher ohms mean easier to drive, then why can't all amps just drive 600 ohms?


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## SilverEars

Well, I have a tube hybrid amp I use with the 650 and when I used it with the 800 it started to crackle at a certain volume level, so my guess was that it was starting to clip.  I had no problems like that with my 650.  I've also tried the 800 with my O2, and I recall it sounded worse than my 650(don't recall if it clipped or not), but I don't like listening to my 650 with the O2.  
  
 I've tried O2 and my Beta 22 with LCD2 and HE-6.  There is noticeable difference in the sound, and I favor the Beta 22 as both of them sound tighter and has better dynamics.


----------



## ab initio

dark_wizzie said:


> If higher ohms mean easier to drive, then why can't all amps just drive 600 ohms?




Because you need enough volts to get loud enough. As a general rule, the higher the impedance, the higher the voltage swing needed to get the same loudness. Many sources like phones or ipods Peter out at 1.5 or 2 volts RMS which may not be loud enough.

Cheers


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## SilverEars

ab initio said:


> Because you need enough volts to get loud enough. As a general rule, the higher the impedance, the higher the voltage swing needed to get the same loudness. Many sources like phones or ipods Peter out at 1.5 or 2 volts RMS which may not be loud enough.
> 
> Cheers


 
 This of course depends on the sensitivity of the headphones.  Even at 600ohms, if the sensitivity is high enough, it will be loud at 1.5 or 2 volts. In the case of Beyers, it's not the case so yeah, it need higher voltage, so the puny DAPs are not providing enough juice.
  
 I think people generally think low impedance as easier to drive because lots of portable headphones are low impedance and they get loud, not because of the low impedance, but because of the high sensitivity.  
  
 This is why HE-6 at 50ohms is still hard to drive because the sensitivity is low, which is quite different from most headphones specs.  That probably throws people off.


----------



## ab initio

silverears said:


> This of course depends on the sensitivity of the headphones.  Even at 600ohms, if the sensitivity is high enough, it will be loud at 1.5 or 2 volts. In the case of Beyers, it's not the case so yeah, it need higher voltage, so the puny DAPs are not providing enough juice.
> 
> I think people generally think low impedance as easier to drive because lots of portable headphones are low impedance and they get loud, not because of the low impedance, but because of the high sensitivity.
> 
> This is why HE-6 at 50ohms is still hard to drive because the sensitivity is low, which is quite different from most headphones specs.  That probably throws people off.




While sensitivity plays a role ( e.g. He-6 are like 50 ohms, but terribly inefficient) physics works against high impedance drivers being loud, given a fixed voltage and driver efficiency. 

The electrical power available to convert into acoustic power for a voltage amplitude of V and a speaker impedance R is V^2/R. Therefore the the power decreases with increasing impedance. 

Cheer


----------



## cjl

dark_wizzie said:


> If higher ohms mean easier to drive, then why can't all amps just drive 600 ohms?


 
 Most well designed desktop amps should be able to drive 600 ohms. As ab initio said though, usually, with 600 ohm headphones, an amp will run out of voltage before it runs out of current, so the limiting factor tends to be the output op amp (or discrete amp, if the designer so chooses), and the voltage supplied to that op amp by the power supply (you need a higher supply voltage to have a higher output, unsurprisingly). Most mobile devices can only go to 1 or 2 volts RMS before running out of voltage, usually because they are driven off a fairly low voltage battery, but any decent desktop design should be able to hit the 7-8V RMS needed to drive some of the less sensitive 600 ohm models.


----------



## cjl

silverears said:


> Well, I have a tube hybrid amp I use with the 650 and when I used it with the 800 it started to crackle at a certain volume level, so my guess was that it was starting to clip.  I had no problems like that with my 650.  I've also tried the 800 with my O2, and I recall it sounded worse than my 650(don't recall if it clipped or not), but I don't like listening to my 650 with the O2.
> 
> I've tried O2 and my Beta 22 with LCD2 and HE-6.  There is noticeable difference in the sound, and I favor the Beta 22 as both of them sound tighter and has better dynamics.


 
 Hmm. Given that it's a tube hybrid amp, it probably wasn't designed with low distortion and accuracy in mind, but I wouldn't have guessed that it would crackle or have any issues with the 800 if it plays the 650 just fine. The 800 is higher impedance at all frequencies (which, as I said, makes it easier on the amp), and it is only 1dB less sensitive at the same voltage (102dB at 1V, vs 103dB at 1V) by Sennheiser's spec. I don't know if there's much more to be said on the subject without measurements of the amp playing into each headphone though.
  
 As for the O2 vs Beta22, have you done a level-matched blind test? I haven't seen a detailed measurement of the B22 from an independent source, but I would expect it to measure near perfect based on AMB's spec, so I would expect both it and the O2 to sound identical with the LCD2. With the HE-6, I would expect them to sound identical at low to moderate volume, but at very high listening levels, the O2 may hit its current limiter, which would be very audible (and make it sound quite different from the B22).


----------



## cjl

ab initio said:


> While sensitivity plays a role ( e.g. He-6 are like 50 ohms, but terribly inefficient) physics works against high impedance drivers being loud, given a fixed voltage and driver efficiency.
> 
> The electrical power available to convert into acoustic power for a voltage amplitude of V and a speaker impedance R is V^2/R. Therefore the the power decreases with increasing impedance.
> 
> Cheer


 
 Sure, but increasing the voltage swing of an amp is pretty straightforward within the range that headphones require - it's often as simple as just increasing the supply voltage (and possibly heatsinking a couple parts). Most amplifiers have lower distortion into higher impedance too, even at a higher voltage level. High current with low distortion is more difficult to achieve, and for a fixed driver efficiency, the power input to a speaker is I^2*R, so the higher the impedance, the lower the current required for a given power level. This is why a low impedance, low efficiency speaker/headphone is so hard to drive.
  
 For some numerical examples, to drive an HD800 to 110dB, you will need about 2.6V RMS, and (depending on frequency) around 4-9mA of current, which is 10-20 milliwatts of power. To drive a Beyerdynamic T1 to the same level requires just 2V RMS and only 1.5-3.5 mA, which is only 3-7mW (again, depending on frequency). So, even though the Beyer is higher impedance, it actually takes a bit less voltage and less than half the power and current to drive it to the same level. This means that the HD800 is harder to drive in every sense than the T1, and any amp that can drive the 800 should comfortably be able to drive the T1. Now, to look at the other end of the impedance spectrum, an Audeze LCD-2 would require 1.8V RMS to hit 110dB (nearly as much voltage as the Beyers) and around 25mA of current, resulting in 47mW of power. So, an amp for the LCD-2 would need just about as much voltage swing as an amp for the T1, but ten times the current, making them harder to drive. Compared to the HD800, the LCD-2 requires almost as much voltage, but around 2-3 times the current, so again, I would consider the LCD-2 the harder headphone to drive between the two (though there could potentially be some corner cases where an amplifier can just barely supply enough voltage for the LCD-2, but has plenty of current capability. This hypothetical amp would drive the LCD-2 just fine, but clip on loud passages with the HD800). Finally, let's look at the HE-6. With a 50 ohm impedance and an 83.5dB/1mW sensitivity (among the lowest I've ever seen for a headphone), it requires 5V RMS to hit 110dB, and in the process, pulls 100mA and about half a watt of power. For the same sound pressure level, it takes twice the voltage swing of an HD800, 10-20x the current, and 25-50x the power. This is why many headphone amps would have trouble with these headphones.
  
 Interestingly, the O2 should actually be able to achieve this 110dB SPL with the HE-6, since at 50 ohms, it can achieve a voltage swing of about 5.5-6V. This doesn't leave a lot of headroom over the 5V required for 110dB though, and if you are listening to very dynamic music, this could still be inadequate. That's why I consider the O2 to be borderline for the HE-6. All of the other numbers I calculated above though are easily within the O2s capability with a huge amount of headroom (and within the capability of many other amps - 2-3V RMS and <50mA of current should be within the abilities of nearly every headphone amp on the market).


----------



## SilverEars

cjl said:


>


 
 I agree you.  Interesting thing about my HE-6 is that I can drive it to loud volume with my O2.  Also, I've been reading about variance to HE-6 impedance by various people.  Some people say they measure 40ohm and others say 60.  Not sure if HM goofed and labled HE-500 as HE-6, but I thought it was quite interesting.


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## Dark_wizzie

cjl said:


> Sure, but increasing the voltage swing of an amp is pretty straightforward within the range that headphones require - it's often as simple as just increasing the supply voltage (and possibly heatsinking a couple parts). Most amplifiers have lower distortion into higher impedance too, even at a higher voltage level. High current with low distortion is more difficult to achieve, and for a fixed driver efficiency, the power input to a speaker is I^2*R, so the higher the impedance, the lower the current required for a given power level. This is why a low impedance, low efficiency speaker/headphone is so hard to drive.
> 
> For some numerical examples, to drive an HD800 to 110dB, you will need about 2.6V RMS, and (depending on frequency) around 4-9mA of current, which is 10-20 milliwatts of power. To drive a Beyerdynamic T1 to the same level requires just 2V RMS and only 1.5-3.5 mA, which is only 3-7mW (again, depending on frequency). So, even though the Beyer is higher impedance, it actually takes a bit less voltage and less than half the power and current to drive it to the same level. This means that the HD800 is harder to drive in every sense than the T1, and any amp that can drive the 800 should comfortably be able to drive the T1. Now, to look at the other end of the impedance spectrum, an Audeze LCD-2 would require 1.8V RMS to hit 110dB (nearly as much voltage as the Beyers) and around 25mA of current, resulting in 47mW of power. So, an amp for the LCD-2 would need just about as much voltage swing as an amp for the T1, but ten times the current, making them harder to drive. Compared to the HD800, the LCD-2 requires almost as much voltage, but around 2-3 times the current, so again, I would consider the LCD-2 the harder headphone to drive between the two (though there could potentially be some corner cases where an amplifier can just barely supply enough voltage for the LCD-2, but has plenty of current capability. This hypothetical amp would drive the LCD-2 just fine, but clip on loud passages with the HD800). Finally, let's look at the HE-6. With a 50 ohm impedance and an 83.5dB/1mW sensitivity (among the lowest I've ever seen for a headphone), it requires 5V RMS to hit 110dB, and in the process, pulls 100mA and about half a watt of power. For the same sound pressure level, it takes twice the voltage swing of an HD800, 10-20x the current, and 25-50x the power. This is why many headphone amps would have trouble with these headphones.
> 
> Interestingly, the O2 should actually be able to achieve this 110dB SPL with the HE-6, since at 50 ohms, it can achieve a voltage swing of about 5.5-6V. This doesn't leave a lot of headroom over the 5V required for 110dB though, and if you are listening to very dynamic music, this could still be inadequate. That's why I consider the O2 to be borderline for the HE-6. All of the other numbers I calculated above though are easily within the O2s capability with a huge amount of headroom (and within the capability of many other amps - 2-3V RMS and <50mA of current should be within the abilities of nearly every headphone amp on the market).


 
 Can you do a for dummies version of your HD800 calculation? How did you get 2.6vRMS, is that the average case or the worst case scenario? So if we want to calculate the worst case scenario for HD800/O2 instead we should look at minimum impedance because that is hardest to drive? Then that would be 350 impedance which is a larger number than the 300 ohms listed on the headphones itself.
  
 So you believe O2 will easily drive HD800, T1, LCD2/3.
  
 And, higher impedance = requires more voltage, less current. Lower impedance = less voltage, more current. So in a way, having either a very high impedance or very low impedance can be an issue for an amp.


----------



## cjl

dark_wizzie said:


> Can you do a for dummies version of your HD800 calculation? How did you get 2.6vRMS, is that the average case or the worst case scenario? So if we want to calculate the worst case scenario for HD800/O2 instead we should look at minimum impedance because that is hardest to drive? Then that would be 350 impedance which is a larger number than the 300 ohms listed on the headphones itself.
> 
> So you believe O2 will easily drive HD800, T1, LCD2/3.
> 
> And, higher impedance = requires more voltage, less current. Lower impedance = less voltage, more current. So in a way, having either a very high impedance or very low impedance can be an issue for an amp.


 
  
 So, with Sennheiser headphones, it's pretty easy. They specify their sensitivity at 1V RMS, so you don't actually have to worry about their impedance at all (unless you want to calculate power required or current). The HD800 are rated at 102dB at 1VRMS. From there, you can calculate the voltage needed for 110dB as shown here:
  
 1) How much gain do we need compared to the reference? We want 110dB, reference specified is 102dB, so we want 8dB more SPL
 2) For 8dB gain, 8 = 20*log(V2/V1)
 3) V1 = 1 volt (specified reference level)
 4) Rearranging the equation, 8/20 = log(V2)
 5) 10^(8/20) = 10^log(V2) = V2
 6) V2= 2.512
  
 If you want to calculate the current or power requirements, you can then look up the impedance at the desired frequency, and use P = V^2/R and V = I*R to figure them out. If you want the worst case power and current, use the minimum impedance to do these calculations.
  
 If you had a pair that specified in dB/mW, you would have an extra step involved. Let's try it with the HE-6:
  
 1) How much gain do we need? Desired = 110, reference = 83.5, gain = 26.5dB
 2) 26.5 = 10*log(P2/P1)    <----- Note the change in the leading constant - for voltage, use 20*log(V2/V1), for power, use 10*log(P2/P1)
 3) P1 = 0.001W (1mW)
 4) 26.5/10 = log(P2/0.001)
 5)10^(2.65) = 10^log(1000*P2) = 1000*P2
 6) 447 = 1000*P2 ---> P2 = 0.447W
 7) P = V^2/R, R = 50, P = 0.447
 8) V = sqrt(0.447*50)
 9) V = 4.73
  
 (and yes, I believe the O2 will easily drive the T1, HD800, and any of the LCD series)
  
 As for your last statement, that is true - either very high or very low impedance can mean that a given amp will be unable to drive the headphones, and in either case, a very low sensitivity will increase the difficulty of driving the headphones.


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## briskly

> Originally Posted by *Dark_wizzie* /img/forum/go_quote.gif
> Can you do a for dummies version of your HD800 calculation? How did you get 2.6vRMS, is that the average case or the worst case scenario? So if we want to calculate the worst case scenario for HD800/O2 instead we should look at minimum impedance because that is hardest to drive? Then that would be 350 impedance which is a larger number than the 300 ohms listed on the headphones itself.
> 
> So you believe O2 will easily drive HD800, T1, LCD2/3.
> ...


 
 Voltage sensitivity is taken at some reference frequency, typically 1khz. The frequency response chart shows you the output of the headphone though a range frequencies given a constant voltage. The current draw can be concluded from the impedance graph of the headphone.
  
 The O2 should be able to power those headphones to near defeaning levels.
  
  
 Interestingly enough, the HE-6 as measured by Tyll is far less sensitive, with an output of 77dB/mW and a 90dB output at 1 volt.


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## cjl

briskly said:


> Interestingly enough, the HE-6 as measured by Tyll is far less sensitive, with an output of 77dB/mW and a 90dB output at 1 volt.


 
 90dB at 1V? That's even worse than my numbers above imply - that means to hit 110dB, you need 10Vrms and nearly 2 watts, well beyond the capability of the O2 (and the majority of other headphone amps out there).


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## warmer

There is a huge difference in the detail of transients between 16bit and 24bit.  I can absolutely hear it with my own ears.  Mixes never sound as good when I bounce them to cd, and it's not because I'm playing from a different source.  The files themselves sound less alive and bouncey in 16bit and when I'm mixing in 24bit.  I used to agree with much of what was stated in the above analysis, but that was before I had enough ear time with the differences to really tell. 
  
 I'm sure I'm not alone in my sentiment.  There is also a very subtle difference between 24bit 48khz and 96khz perceptibly speaking.  I can't any tell any difference difference between 96 and 192.


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## cjl

warmer said:


> There is a huge difference in the detail of transients between 16bit and 24bit.  I can absolutely hear it with my own ears.  Mixes never sound as good when I bounce them to cd, and it's not because I'm playing from a different source.  The files themselves sound less alive and bouncey in 16bit and when I'm mixing in 24bit.  I used to agree with much of what was stated in the above analysis, but that was before I had enough ear time with the differences to really tell.
> 
> I'm sure I'm not alone in my sentiment.  There is also a very subtle difference between 24bit 48khz and 96khz perceptibly speaking.  I can't any tell any difference difference between 96 and 192.


 
 Ever done a proper double blind ABX?


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## MrMateoHead

*delete*


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## Dark_wizzie

cjl said:


> 90dB at 1V? That's even worse than my numbers above imply - that means to hit 110dB, you need 10Vrms and nearly 2 watts, well beyond the capability of the O2 (and the majority of other headphone amps out there).


 
 With something like that, I'd just skip HE2 altogether. There are other good options that do not require this type of amplification.


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## Danz03

Although I do agree with the article about the 24bit myth, 24bit/96Hz do have some uses for some listeners, that is, if they were to use some digital plugins like EQ, pitch change, tempo change, noise reduction etc. 



warmer said:


> There is a huge difference in the detail of transients between 16bit and 24bit.  I can absolutely hear it with my own ears.  Mixes never sound as good when I bounce them to cd, and it's not because I'm playing from a different source.  The files themselves sound less alive and bouncey in 16bit and when I'm mixing in 24bit.  I used to agree with much of what was stated in the above analysis, but that was before I had enough ear time with the differences to really tell.
> 
> I'm sure I'm not alone in my sentiment.  There is also a very subtle difference between 24bit 48khz and 96khz perceptibly speaking.  I can't any tell any difference difference between 96 and 192.


----------



## Digitalchkn

cjl said:


> So, with Sennheiser headphones, it's pretty easy. They specify their sensitivity at 1V RMS, so you don't actually have to worry about their impedance at all (unless you want to calculate power required or current). The HD800 are rated at 102dB at 1VRMS. From there, you can calculate the voltage needed for 110dB as shown here:
> 
> 1) How much gain do we need compared to the reference? We want 110dB, reference specified is 102dB, so we want 8dB more SPL
> 2) For 8dB gain, 8 = 20*log(V2/V1)
> ...


 
  
 Just want to make a quick comment here difference RMS and peaks (voltage or currents or power). Some care must be taken when talking about specifying these numbers. Manufacturers are not always clear on this.


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## cjl

digitalchkn said:


> Just want to make a quick comment here difference RMS and peaks (voltage or currents or power). Some care must be taken when talking about specifying these numbers. Manufacturers are not always clear on this.


 
 Manufacturer specified sensitivities are almost always in RMS power/voltage. You're right that you do need to be careful though.


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## Digitalchkn

cjl said:


> Manufacturer specified sensitivities are almost always in RMS power/voltage. You're right that you do need to be careful though.


 
 Also, and more importantly the interesting metric is the peak output voltage or current . Because it, not the RMS value, is the determining factor whether a particular amp can drive distortion free.


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## cjl

digitalchkn said:


> Also, and more importantly the interesting metric is the peak output voltage or current . Because it, not the RMS value, is the determining factor whether a particular amp can drive distortion free.


 
 It depends on how the amp is measured/specified. A lot are specified in RMS, but some are specified in peak (since the number is higher). As long as you know which one you're working with, either can be used.


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## icesenshi

Here's an article from the register, affirming that 16bit is enough, and a few other things.
  
 http://www.theregister.co.uk/2014/05/13/apple_beats_and_fools_with_money
  
 "Even worse is that in the hi-fi world, that vacuum is filled by pseudoscience, dogma and fruitloopery to the extent that it resembles a fundamentalist religion. And when fundamentalism gains a hold, science leaves on the next boat – and progress ceases."


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## bigshot

great article


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## Tuco1965

That was a good read.


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## sonitus mirus

It reads great, to a point; but, it's clear to me that the information has been garnered from reading and not from direct experience.  
  
 "Some time ago, Young expressed concerns about what was left of his music after being downloaded by the consumer. Those concerns are entirely justified, as MP3 is a lossy compression scheme and at limited bit rates – such as the 320kbit/sec of Beats Music – does its best to preserve the dominant sounds by neglecting ambience and reverberation."


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## bigshot

A simple ABX test puts the lie to that.


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## icesenshi

I cannot tell the difference between 320 and flac or other lossless formats, so I think the author is wrong about that part. But the rest of what he says I pretty much agree with.


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## headwhacker

icesenshi said:


> I cannot tell the difference between 320 and flac or other lossless formats, so I think the author is wrong about that part. But the rest of what he says I pretty much agree with.


 
  
 That is why 16 bit is already overkill. I think people think 16 bit and 24 bit audio has the same correlation with 16 bit and 24 bit on video in terms of number of colors can be produced. Hence they refer to 16 bit audio being lower resolution compared to 24 bit.


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## Baxide

icesenshi said:


> Here's an article from the register, affirming that 16bit is enough, and a few other things.
> 
> http://www.theregister.co.uk/2014/05/13/apple_beats_and_fools_with_money
> 
> "Even worse is that in the hi-fi world, that vacuum is filled by pseudoscience, dogma and fruitloopery to the extent that it resembles a fundamentalist religion. And when fundamentalism gains a hold, science leaves on the next boat – and progress ceases."


 

 A terrible propaganda piece against high resolution. The writer is obviously living in denial and/or has a terribly bad audio system.


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## kraken2109

baxide said:


> A terrible propaganda piece against high resolution. The writer is obviously living in denial and/or has a terribly bad audio system.


 

 Please tell us more about this denial


----------



## limpidglitch

kraken2109 said:


> Please tell us more about this denial


 
  
 Obvious satire, intentionally or not.


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## cjl

baxide said:


> A terrible propaganda piece against high resolution. The writer is obviously living in denial and/or has a terribly bad audio system.


 
 Please, tell us more about these double blind tests where you've clearly demonstrated an ability to tell the difference between 16/44.1 and "high-resolution" formats...


----------



## esldude

Well like cameras or video you have your resolution products.  With audio 44.1 khz and 16 bit would be 22050 possible frequencies at 65K possible levels.  A total resolution product number of frequency and levels of 22,050x65,536=1,445,068,800.
  
 Then your 192khz 24 bit is good for 16,777,216x96,000=1.61x10 to the 12th power. 
  
 As doubling sample rates to 384 khz only doubles the resolution product. I think more bits is better.  Adding just one bit gets you twice as much.  Much more efficient way to expand the resolution product. 192/32 bit would give 8.25 x10 to the 14th power.  Far more resolution product that way for just a few bits extra. 
  
 This evaluation of resolution also gives lie to the idea that DSD is high resolution.  Only 50,000 different frequencies at one bit.  Okay since one choice is 0 as well as one I'll be generous and say it is 50,000x2=100,000.  Totally pathetic vs even redbook for resolution products. 
  
 So I think these increasingly high sample rates are really just marketing bull.  They intend to sell everyone 192khz and in a few years it will all be 384 khz rates and you'll need to buy your recordings all over again.  They need to make the jump directly to 32 bit and start working on 48 bit for now. 
  
  
  
  
 Yes, I know the above reasoning is ridiculous.  But it makes more sense than plenty of marketing ads explaining benefits I have seen in many cases.


----------



## bigshot

Most people will take what you typed there at face value. Your slant to what you're saying doesn't come through.


----------



## Danz03

It depends on how the tracks were mixed and mastered too. The human hearing range is about 140 dB, 24 bit gives you 144 dB of dynamic range, 16 bit gives you 96 dB, but most music nowadays, apart from classical and some jazz, has a dynamic range of around 5-7 dB, with some as little as 1-2 dB. So I don't see a lot of improvement between 24 and 16 bit unless the actual 24 bit track was mastered differently, hopefully with much less compression and limiting. A lot of the older analogue recordings including classical and jazz have very high noise floor, some as much as 60 dB, so 24 bit will not improve the sound quality unless the noise gets digitally removed. A lot of the older records from the Beatles to Pink Floyd have been remastered and remastered over the years with less and less dynamic range and with more and more enchantment through digital plugins, so the music sounds cleaner and clearer, but at the same time there are a lot of digital noises and artefacts added which weren't there in the original recording. So to me, if the 24 bit file was just an up-sampled 24/96 or a pre-down samples 24/96 version of the same master, I don't see much point in getting them than the CD version unless you're going to do your own version of remastering.


----------



## esldude

esldude said:


> Well like cameras or video you have your resolution products.  With audio 44.1 khz and 16 bit would be 22050 possible frequencies at 65K possible levels.  A total resolution product number of frequency and levels of 22,050x65,536=1,445,068,800.
> 
> Then your 192khz 24 bit is good for 16,777,216x96,000=1.61x10 to the 12th power.
> 
> ...


 
  
  
 In case it was not clear, the above is an example of making something sound like it makes sense.  Like there is a measure to resolution, when in fact what I wrote makes no sense and is misleading in several ways.  Sure it has some simple math, uses the parameters of different sample rates and bit depths, but the use of them is out of context and misleading.  Just like lots of high end marketing spiel people read in magazines and on web sites, and on audio forums.


----------



## castleofargh

danz03 said:


> The human hearing range is about 140 dB


 
 that might be the acknowledged dynamic, 1volume level at a time, one sound at a time, from ludicrously lowest perceived sound first, and then up to 140db louder when the guys started crying "plz kill me!!!!".
 it has nothing to do with actual music listening, and the noise floor is a far away problem if we don't do dumb things with our digital volume control. simply by listening at say 90db on my phone, I already by myself remove a good 50db from my maximum perception range. -140db is a nice dream for people willing to listen to music at 141db so they can hope to grasp the finest details, when our hears would have already activated the emergency safety measures to reduce percieved sounds. then you add the masking effect and you see how candid it is to believe we can hear -100db sound behind music playing above.
  
 that 140db value is a joke from the beginning when we're talking music. I just launched 2foobar sessions (2installs in different partitions, or one in a sandbox obviously not with wasapi ^_^). and played one song at max level and another at the same time 40db lower. it's already pretty fun how low the sound on the second is and how hard it is to get all the sounds when I know perfectly what I'm trying to listen to. and that's just -40db. -140db? impossibru!!!!


----------



## Steve Eddy

The instantaneous dynamic range of human hearing is about 60dB by the way.

se


----------



## Danz03

That is exactly my point, although the human hearing range is 140 dB, we don't need the 144 dB in 24 bit audio. The loudest I've had to monitor music in mixing studios was around 105 dB, and I already had to wear ear plugs to protect my hearing, that is why so many musicians like Eric Clapton and Phil Collins are having hearing problems when they got older. A lot of rock concerts can be louder than 105 dB, depending on where you're standing. It's the same with the 20kHz treble, most ppl can't even hear that anymore when they're over 20, so it seems like 192 kHz sampling rate is quite an overkill.



castleofargh said:


> that might be the acknowledged dynamic, 1volume level at a time, one sound at a time, from ludicrously lowest perceived sound first, and then up to 140db louder when the guys started crying "plz kill me!!!!".


----------



## Steve Eddy

danz03 said:


> That is exactly my point...




Yes, I know. Was just putting a number on it to help give others an idea of how poor the effective dynamic range of our hearing is.

You familiar with the two files produced by Bill Waslo? One is clean, the other has some Sousa music playing something like 70dB below it. Don't think anyone has passed the ABX that Bill intended it to be used with. If the dynamic range of our hearing approached even that of 16 bit, it would be trivially easy to tell them apart.

se


----------



## bbmiller

So how do the above responses, I mean about the last six is so, relate to trolling for the best possible music files to listen to. I have obtained some files which are said to be music video files from blue Ray at 24-bit depth. These files sound better to my ears than the ones of only 16-bit depth. So is that only because the translation from the Masters by the engineers on the 24-bit version is superior?
  
 Also I possess a DAC which is the "Little Dot DAC_I Digital to Analog Converter" which is set to have the fall 24-bit 192 kHz capability if you wanted to. I have been always utilizing this DAC in the automatically select the best operating mode mode. And it has always said on the little LED display that 44.1 kHz is selected. So are there some files I should be listening to it by using the force mode switches to achieve the best possible sound? Or do you think I can continue trusting the automatic select mode mode?


----------



## Danz03

Exactly, but unfortunately, that's the same reason why music is being mastered louder and louder until there's hardly any dynamic range left. At the moment the average for chart music is around 5-7 dB, who knows, it'll probably become less then 3 dB in a decade. Also those brick wall limiters add chipping and distortion to music, and the more they compress, the more distortions and chipping there will be. Lots of ppl think that vinyls sound better than CDs, it's not because vinyls have better sound quality, it's mainly because vinyls cannot be mastered loud, so there is much more dynamic range with vinyl mixes than CDs even though vinyls have a smaller dynamic range technically.

A lot of ppl claimed they can hear the difference between a 24/192 and a 16/44 file, I'm sure some can but it really depends on the actual recording and mix. And even if there is a difference, it doesn't always mean it's better.



steve eddy said:


> Yes, I know. Was just putting a number on it to help give others an idea of how poor the effective dynamic range of our hearing is.
> se


----------



## Danz03

As with the noise issue, no matter what ppl say, I hate it! Especially background noises! There is this old recording of a Schubert Piano Trio that I really like but I just can't stand the background noise on it. It was obviously mastered on analogue tape originally with a noise floor of around 60dB. It probably wouldn't matter much if it was on a Katy Perry song that has 5dB dynamic range but on a classical track, it does stand out a lot and is extremely irritating to me. Eventually I had to remove the noise digitally with Izotope RX2. Now since the file was digitally processed, I could have resampled it to 24/192 if I wanted to, and that would be technically a 24/192 remaster, now obviously that would sound better that the original 16 bit file. But if I had output another 16/44 file with 4X over-sampling, I don't think it would sound much different to the 24/192 version. 



steve eddy said:


> You familiar with the two files produced by Bill Waslo? One is clean, the other has some Sousa music playing something like 70dB below it. Don't think anyone has passed the ABX that Bill intended it to be used with. If the dynamic range of our hearing approached even that of 16 bit, it would be trivially easy to tell them apart.
> 
> se


----------



## Dark_wizzie

bbmiller said:


> So how do the above responses, I mean about the last six is so, relate to trolling for the best possible music files to listen to. I have obtained some files which are said to be music video files from blue Ray at 24-bit depth. These files sound better to my ears than the ones of only 16-bit depth. So is that only because the translation from the Masters by the engineers on the 24-bit version is superior?
> 
> Also I possess a DAC which is the "Little Dot DAC_I Digital to Analog Converter" which is set to have the fall 24-bit 192 kHz capability if you wanted to. I have been always utilizing this DAC in the automatically select the best operating mode mode. And it has always said on the little LED display that 44.1 kHz is selected. So are there some files I should be listening to it by using the force mode switches to achieve the best possible sound? Or do you think I can continue trusting the automatic select mode mode?


 
 Either master or placebo. And I know placebo is not on your mind, that's why placebo is still a factor. It's pretty hilarious what the human mind can trick us into sensing, and it's not really the fault of you per say, but humans in general.
  
 The NorthWesternerAudioVideo person noted that many Dacs are claiming 24bit playback but in reality do not achieve this. What I also read is you should have software volume at max and change volume via volume knob, otherwise that 24 or 16 bit goodness is getting further reduced on top of whatever is going on. But in reality, good luck hearing 24 vs 16 bit. Or even 15 vs 16 bit.
  
  
 Good old fashion 16bit 44khz is just fine. And if you want to argue technical reasons why that's no so, HydrogenAudio would love to deconstruct your claims methodically, one by one.


----------



## PDVJAM

So, if human can't hear the difference between 16 and 24bit records - is there any sense to buy players that can play 24/192? Or this is also marketing and hunting for our money?


----------



## bigshot

Not for playing back music at that bit rate. But most modern players and DACs use up sampling ability to apply a more accurate filter above the range of human hearing. Without that, the rolloff from the filter can extend into the audible range a little.


----------



## castleofargh

pdvjam said:


> So, if human can't hear the difference between 16 and 24bit records - is there any sense to buy players that can play 24/192? Or this is also marketing and hunting for our money?


 
 no a 24/96 or more dac can benefit in several ways from a 16bit track. edit: it's the other way around ^_^.
 I would tend to believe that 32bit is very much useless unless the last 8bits are used for digital volume control. but playing a 16bit file in 24bit is a good thing(or at least not a bad one). not that it might be audible, but it could be usefull on noise and volume control(digital). and by all means it has no bad effect at all, so I would always suggest you to output all your tracks as 24bit if it's ok with the dac.
 and about upsampling, well that seems to be pretty usefull for some filters as said above. but it would be done by the dac without us knowing anyway ^_^.
  
 so all in all, if the benefits might not be audible, they do no wrong this time.


----------



## harmonix

danz03 said:


> It depends on how the tracks were mixed and mastered too. The human hearing range is about 140 dB, 24 bit gives you 144 dB of dynamic range, 16 bit gives you 96 dB, but most music nowadays, apart from classical and some jazz, has a dynamic range of around 5-7 dB, with some as little as 1-2 dB. So I don't see a lot of improvement between 24 and 16 bit unless the actual 24 bit track was mastered differently, hopefully with much less compression and limiting. A lot of the older analogue recordings including classical and jazz have very high noise floor, some as much as 60 dB, so 24 bit will not improve the sound quality unless the noise gets digitally removed. A lot of the older records from the Beatles to Pink Floyd have been remastered and remastered over the years with less and less dynamic range and with more and more enchantment through digital plugins, so the music sounds cleaner and clearer, but at the same time there are a lot of digital noises and artefacts added which weren't there in the original recording. So to me, if the 24 bit file was just an up-sampled 24/96 or a pre-down samples 24/96 version of the same master, I don't see much point in getting them than the CD version unless you're going to do your own version of remastering.


 
 I get what you are saying - IMHO it's not just the recording and mastering chain either. Alot of modern day rock/pop musicians don't play with real dynamics.
 How many times have you heard musicians that can only play f, ff and fff ? Really annoying.
  
 I think electric instruments and amplification are the root cause of alot of this - maybe people should learn to play unplugged but then alot of things would really sound crappy but hey that may not be a bad thing.


----------



## groovyd

one thing to consider is most dacs and adcs have anywhere from 1 to 3 bits of error. so while a 16 bit dac may be sufficient on paper in reality it may come up a bit short.  it is like building a bridge to the minimum spec for the intended load.  a little wind and a heavy truck and you have Tacoma Narrows.


----------



## Lespectraal

Well we are subjective beings somewhat, objective measures don't measure up to our own bug dsp chips which are our brains.. Large numbers have positive connotations so that's why very large resolutions appear when they actually do not provide any actual improvement. 

Either way, economics, marketing and loads of other social and psychological factors come into play when we value anything. That is why we have DSD and DXD and whatnot because you can't just separate the ideal from the practical. Everything comes as a whole instead of discrete parts that can be scrutinized individually.

Just my two cents


----------



## esldude

groovyd said:


> one thing to consider is most dacs and adcs have anywhere from 1 to 3 bits of error. so while a 16 bit dac may be sufficient on paper in reality it may come up a bit short.  it is like building a bridge to the minimum spec for the intended load.  a little wind and a heavy truck and you have Tacoma Narrows.


 

 How many DAC chips are only 16 bits now?  Giving you the benefit of the doubt, 24 bits minus 3 bits is 21 bits to do 16 bit data.  Seems plenty sufficient.  The great majority of current DACs all the way out to the analog output exceed the 96 db needed for clean accurate 16 bit reproduction.


----------



## headwhacker

lespectraal said:


> Well we are subjective beings somewhat, objective measures don't measure up to our own bug dsp chips which are our brains.. Large numbers have positive connotations so that's why very large resolutions appear when they actually do not provide any actual improvement.
> 
> 
> 
> ...



 


The question of needing a 24-bit DAC for 16-bit playback is not subjective in this context. For DAPs which use digital volume control 24 bit DAC is essential to maintain good SNR. For USB DAC purpose, ASRC also helps to reduce jitter which requires 24 bit DAC even for 16 bit playback.


----------



## Lespectraal

headwhacker said:


> lespectraal said:
> 
> 
> > Well we are subjective beings somewhat, objective measures don't measure up to our own bug dsp chips which are our brains.. Large numbers have positive connotations so that's why very large resolutions appear when they actually do not provide any actual improvement.
> ...



I guess I was being too general there in my post.

Of course there exceptions, like the bit depth in the case or digital volume control. I'm just talking about high resolution audio in general, like needlessly high sampling rates and dual-mono dac chip arrangements and whatnot.


----------



## groovyd

agreed - was just bringing up the bit error point as a comparison against when DACs were 16 bits in saying that you may not be really getting 16 bits worth of signal.


----------



## Digitalchkn

groovyd said:


> agreed - was just bringing up the bit error point as a comparison against when DACs were 16 bits in saying that you may not be really getting 16 bits worth of signal.


 
  
 Unless you have a player from the 80's/early-90's you are likely experience a DAC that's some variation of oversampling+deltasigma. The linearity error in those tends to be effectively less one bit equivalent of 24bits. The SNR is pushing 115-120dB or so. In effect, you are getting the error floor of >16bit DAC even when you are playing 16bit data.


----------



## Danz03

I don't know, the lack of dynamics may not be a bad thing for most ppl, as most ppl nowadays listen to music while commuting, jogging, driving etc. With a small dynamic range like less than 5dB, all you need is for the music level to be 5dB over the noise level so you don't get part of it drowned out by the ambient noise. The ambient noise level in the city where I work in averages around 83dB and peaks at around 96, if I were wearing non-noise isolating ear buds I'd have to have the music level as high as 88dB to even hear it properly. Can you imagine if the dynamic range were higher, like 15dB? So I think it is a kind of dilemma how loud should music be mixed at. Probably best solution would be a low dynamic version as standard and a special high dynamic audiophile version in 24/192? Most cinema goers would know what high dynamic music sounds like, as most cinema sounds can have a 30dB + dynamic range.
  
 Quote:


harmonix said:


> I get what you are saying - IMHO it's not just the recording and mastering chain either. Alot of modern day rock/pop musicians don't play with real dynamics.
> How many times have you heard musicians that can only play f, ff and fff ? Really annoying.
> 
> I think electric instruments and amplification are the root cause of alot of this - maybe people should learn to play unplugged but then alot of things would really sound crappy but hey that may not be a bad thing.


----------



## Notus

Funny thing you mentioned movies. Everybody i know have always complained about the sounds in movies, the weaker sounds tend to drown out in the ambient noises ( often times talking ).
 But the explosions and music can be very very loud. In cinemas they crank the volume so high that it sometimes feels like you can get permanent hearing damage from the Peak sounds.
 This is actually one aspect that in my opinion should be discussed, can higher dynamic range lead to increased cases of hearing damage if 24 / 96 is made standard?
 Most adults will know to limit the volume so no damage is done, but what about the young people?
 While i think higher dynamic's can serve some people, i think those people are very small percentage of music listeners over all.
 Like Danz03 pointed out, there are situations where high dynamic range would work to hinder music listening quite a bit.


----------



## stv014

Dynamic range that would actually benefit from 24-bit resolution (and not just for marketing reasons) would indeed not be practical for most people listening at home under typical conditions. The noise floor of CD quality audio even with the simplest dithering at a loud 110 dB peak SPL is about 14 dB (A-weighted), which is normally not noticeable, especially when there is any music to mask it.
  
 The problem with the "loudness war" is not just that the dynamic range is compressed, but it is done beyond the point of diminishing returns, when the last few dB of extra loudness is achieved at the cost of severe and audible distortion.


----------



## castleofargh

notus said:


> Funny thing you mentioned movies. Everybody i know have always complained about the sounds in movies, the weaker sounds tend to drown out in the ambient noises ( often times talking ).
> But the explosions and music can be very very loud. In cinemas they crank the volume so high that it sometimes feels like you can get permanent hearing damage from the Peak sounds.
> This is actually one aspect that in my opinion should be discussed, can higher dynamic range lead to increased cases of hearing damage if 24 / 96 is made standard?
> Most adults will know to limit the volume so no damage is done, but what about the young people?
> ...


 
  
  real noisy environments as danz03 suggested, or the opposite, when you want to listen very very quietly, then music with low dynamic range might help. I can't imagine listening to some great symphonies in the subway, I would have to listen to deafening levels to hear most of the sounds,else I would miss 2/3 of the symphony.
  
 but for "usual" listening, a big dynamic is actually much less fatiguing to the ears. what ruins your ears isn't only the pressure level of the sound(unless you play it at unreasonable levels) but also the duration of that pressure. when you have some album with 8db of DR(hello metallica...), you'll listen to that level all the album, and the quiet parts will never really be quiet.
 when you have something with a lot of dynamic, you'll set the volume control for the loudest parts higher, yes! but you'll also have a lot more moments when the pressure from the music will be lower than the minimum on metallica. ups and downs(high DR) are more natural and ok than constant moderate-to-high volume levels(low DR). I'm making a general statement here so there must be some point where such DR is better for the ears than such. but overall, big DR(who are we kidding, 60DB on most tracks would be a dream come true) is considered better for the ears.
  
 and for all of the above, I'm talking actual dynamic range of the music, not dynamic range available because it's 16 or 24bit. it's not like there was a lot of tracks out there that actually use 16bit of dynamic. so I wouldn't worry too much about 24/96 becoming a standard.


----------



## blades

The purpose of 24 bit audio is to provide the music producers with some headroom for mixing and mastering.  Downsampling a recording to 16 bit red book doesn't cause any audible loss.  16 bits can handle all the dynamic range we can handle.  If anyone is interested in doing a bias controlled listeing test, there is a feature for Foobar that can make that easy for you.


----------



## Digitalchkn

blades said:


> The purpose of 24 bit audio is to provide the music producers with some headroom for mixing and mastering.


 
  
 I am curious to know why would music producers bother with 24 bits if 16 bits is sufficient?


----------



## ab initio

The extra headroom allows them to mix parts with adjusted gains and apply affects while keeping the noise floor at negligibly small levels. It also simplifies the recording process: the overall level of the recording can be kept low enough to avoid clipping while there is sufficient dynsmic range to record everything without the noise floor being a problem. With 24 bits, it is easy to keep the entire recording within the dynamic range without needing the careful setup and sound check required when using 16 bit or analog recording equipment.

 Cheers


----------



## groovyd

so basically engineering of music should be done in 24 bits so that we get a quality 16 bits out.  i would agree a well mastered 24 bit track then quantized down to 16 bits is really all a human ear can perceive.  the biggest issue in recording quality is the effects of not mastering a track properly.


----------



## Digitalchkn

By same token might as well keep things as 32 (64, oh my) bits to line up to the native word processor word boundaries. These days storage is cheap. So is processing the power. Overkill? Yes. But it's almost for free. Don't you agree?


----------



## bigshot

More data increases the odds of file corruption. If you've ever had a protools project that got corrupted, you know how bad that can be. There is something to be said for just one level of overkill, not two or three or a dozen levels.


----------



## Digitalchkn

bigshot said:


> More data increases the odds of file corruption. If you've ever had a protools project that got corrupted, you know how bad that can be. There is something to be said for just one level of overkill, not two or three or a dozen levels.


 
  
 That's more of an example of a flaw specific to protools. With modern error checking and correction mechanisms data integrity shouldn't be a primary concern.


----------



## bigshot

Anything that you are editing is subject to error occasionally, not just protools. The bigger the files you are working with, the more chance of corruption. Hard drives themselves can be the culprit through data deterioration.


----------



## groovyd

this falls into the concept of safety margin... like when you build a bridge to handle double or triple the expected worst case scenario.  In reality going from 16 to 24 or even 32 bits should not dramatically increase file size if indeed a well behaved lossy compression is being used that limits the information in the file to the true limits of human perception in which case most of the extra bits are all redundant and compressed out. in the case of lossless compression the same is also true assuming there is actually no more significant information contained in the data.  For example in the case where the recording does not use the full dynamic range indeed the most and least significant bits are all 0 anyway.


----------



## cjl

bigshot said:


> Anything that you are editing is subject to error occasionally, not just protools. The bigger the files you are working with, the more chance of corruption. Hard drives themselves can be the culprit through data deterioration.


 
 Not necessarily - you could make the file larger by adding a lot of ECC at the end of each sample, which would make the chance of corruption go down by an enormous amount.


----------



## Digitalchkn

>


 
  


bigshot said:


> The bigger the files you are working with, the more chance of corruption. Hard drives themselves can be the culprit through data deterioration.


 
  
 I think this statement is no longer as valid as it once was. While data corruption is still a real problem it has been mitigated to a high degree of reliability.  If it were not the case the modern computing and networking systems would be in perpetual state of crashing considering the amount of data that must be processed. Or at least fail far more often than we see in practice.


----------



## headwhacker

digitalchkn said:


> By same token might as well keep things as 32 (64, oh my) bits to line up to the native word processor word boundaries. These days storage is cheap. So is processing the power. Overkill? Yes. But it's almost for free. Don't you agree?




Storage may be cheap but doesn't mean you have to waste it. A personal music collection only grows bigger. In a mobile/DAP scenario it's convenient to have all you collection with you all the time. The bigger the file the harder the DAP has to work and significantly shortens battery life. Flash storage is not as cheap as mechanical. So your argument does not have as strong merit yet in the mobile space.


----------



## Steve Eddy

headwhacker said:


> Storage may be cheap but doesn't mean you have to waste it.




It's like getting people to move toward higher efficiency air conditioners to save energy. But instead of saving energy, they just use them more. 

se


----------



## bigshot

In mixing, higher bit rates tax the processor more. This means that if you add too many RT filters, frames will start to drop in monitoring. If you have a very complex mix, like I often do in TV work, with multiple music, dialogue and effects tracks all with different RT filtering applied, super high bit rates can grind the processor to a halt. The more data you can push through a processor, the more data wants to be pushed through. It pays to be efficient and not waste resources on things that just don't make a difference.


----------



## groovyd

couple more years and between the abundance of ultra cheap flash and ultra fast and omnipresent cloud storage and both lossy and lossless compression advances and I don't think file size will concern anyone anymore.  I would like to see 24/96 as the new 16/44 into the foreseeable future.


----------



## esldude

groovyd said:


> couple more years and between the abundance of ultra cheap flash and ultra fast and omnipresent cloud storage and both lossy and lossless compression advances and I don't think file size will concern anyone anymore.  I would like to see 24/96 as the new 16/44 into the foreseeable future.


 

 I think 24 bit /48 khz would be a more appropriate standard than 16/44 actually.  Certainly we don't need even 24/96, but some will want it anyway.


----------



## groovyd

yeah, bandpass filters are not perfect or ideal so having twice the number of samples then needed for the highest frequencies will do a lot to smooth the true response and give reasonable safety margin for even the most critical head-fier using single crystal gold power cords.


----------



## headwhacker

Isn't 48khz already twice the sample of up 22KHz band? All those samples can more than fit into 16 bits.


----------



## bigshot

Whenever you establish a threshold where you say "anything beyond this doesn't matter", there's always some joker in the crowd who insists that the line needs to be just a few feet further along... "just to be safe... just in case..." This happens over and over again with people worrying about "that last five percent" until the line gets pushed way over into La La Land.
  
 Redbook standards were based on human hearing thresholds. If it performs to redbook spec, it's good enough for human ears. Period. Anyone who thinks they need more are either super humans or not in touch with reality. Take a guess which group I think they belong to!


----------



## castleofargh

for frequency needs (for humans) 44khz is all we will ever need. people saying otherwise are mathematically wrong.
 now 48khz allows for better work on the low pass filter, without being uselessly big in data. so it does sound like the very reasonable best choice. not that I can really complain about 44.1.

 24bit has no defaults except size. worst case scenario,it doesn't do anything. so when possible it can actually be a cool thing. but playing 16bit as 24bit in foobar already brings most of the benefits of 24bit.
 between playing a 16bit in 24bit or having 24bit files:
 - digital volume control get the same margin
 -quantization noise from the dac will also be the same
 the only real difference is that the original file will have higher noise if in 16bit from the ADC right?  but we're talking about stuff done in pro studios, so it's reasonable to say that noise is really well under -96db and means nothing to us.

 everything I've seen pushes me to be a fan of 16/48 but I have no problem with 24/48(except size).
 wouldn't 48khz be better than 44.1 when the dac does the upsample/oversample into 96khz or 192khz as it would be a simple division by 2 or 4 operation?


----------



## Lespectraal

Imagine uploading a raw image file for use of your profile picture on Facebook, that is how I feel about these high res nonsenses. What can we do about it. Life goes on.


----------



## Digitalchkn

esldude said:


> I think 24 bit /48 khz would be a more appropriate standard than 16/44 actually


 
 ++


----------



## esldude

bigshot said:


> Whenever you establish a threshold where you say "anything beyond this doesn't matter", there's always some joker in the crowd who insists that the line needs to be just a few feet further along... "just to be safe... just in case..." This happens over and over again with people worrying about "that last five percent" until the line gets pushed way over into La La Land.
> 
> Redbook standards were based on human hearing thresholds. If it performs to redbook spec, it's good enough for human ears. Period. Anyone who thinks they need more are either super humans or not in touch with reality. Take a guess which group I think they belong to!


 

 People living in super human reality obviously!


----------



## Digitalchkn

castleofargh said:


> wouldn't 48khz be better than 44.1 when the dac does the upsample/oversample into 96khz or 192khz as it would be a simple division by 2 or 4 operation?


 
  
 Not exactly that sample. Still need to apply anti-aliasing filter prior to downsampling.


----------



## Digitalchkn

bigshot said:


> In mixing, higher bit rates tax the processor more. This means that if you add too many RT filters, frames will start to drop in monitoring. If you have a very complex mix, like I often do in TV work, with multiple music, dialogue and effects tracks all with different RT filtering applied, super high bit rates can grind the processor to a halt. The more data you can push through a processor, the more data wants to be pushed through. It pays to be efficient and not waste resources on things that just don't make a difference.


 
  
 I see it as one argument for native 32bit or 64bit word size, actually. 24bits processed on a 32bit machine is still going to eat up same cycle. There is really no advantage to trying to take advantage of a "fractional" cycle. The 8 bits just get thrown away.


----------



## esldude

digitalchkn said:


> I see it as one argument for native 32bit or 64bit word size, actually. 24bits processed on a 32bit machine is still going to eat up same cycle. There is really no advantage to trying to take advantage of a "fractional" cycle. The 8 bits just get thrown away.


 

 Well, a good many software DAWs end up doing 24 bit files in a 32 bit float format.  Allows processing that for the most part doesn't intrude on your 24 bits from rounding or truncation issues. Some software even uses 64 bits.


----------



## bigshot

digitalchkn said:


> Not exactly that sample. Still need to apply anti-aliasing filter prior to downsampling.


 
  
 Which for playback is handled by DACs that upsample to apply the anti aliasing filter, then back to redbook. You don't need to use up sampling native to get around that problem.
  
 None of this discussion matters, because it just plain ain't audible.


----------



## Digitalchkn

bigshot said:


> Which for playback is handled by DACs that upsample to apply the anti aliasing filter, then back to redbook.


 
  
 Going from 48K to redbook still needs anti-aliasing applied somewhere, upsampled first or direct downsampling. Upsampling should apply an interpolation filter, technically. So it's not a simple e.g. x2 or /2 operation.


----------



## bigshot

antialiasing is transparent. The problem is the brick wall.


----------



## Digitalchkn

It depends on the filter you apply. You sort of want a sharp filter (brickwall) with narrowest possible stopband bandwidth and highest stopband rejection. All this while having the minimum passband ripple and closest to linear phase response.  The more processing horsepower you have available the higher the filter order can start getting you closer to ideal. But no filter is perfect of course. That's one reason for having headroom in your source's sampling rate when converting and converting to 48K rather than redbook.  It's just you can get slightly sloppier converting to 48K since your <20KHz would be less affected by a less ideal filter.


----------



## castleofargh

digitalchkn said:


> castleofargh said:
> 
> 
> > wouldn't 48khz be better than 44.1 when the dac does the upsample/oversample into 96khz or 192khz as it would be a simple division by 2 or 4 operation?
> ...


 

 again I realize I write stuff as they come in my head without a care in the world for making sens. it looks like I wrote "take (sampleA+sampleB)/2 to generate the middle sample" as if it was a straight line..
 but you answered me anyway (you guys are really strong at getting into my brain even when I'm writing nonsense).
 yeah I was thinking about the all process, if it comes from a master in 24/96, going 48khz for our albums would be best so that we would just have to make an interpolation to get the exact downsampled values back. but yeah I didn't think aliasing, so it can't be that simple and precise. and thus I don't know if 44.1 or 48 change anything for that particular matter.
 I'm still a 48khz fan for everything else, but we would need cds directly done in 48khz, no point turning a perfectly ok redbook to 48 on our own.


----------



## Notus

So people think its okay to pay more for the sake of "just in case". To me that makes no sense at all. True storage is quite cheap these day's but i think people underestimate the need for space with 24bit files. Storage is not the only problem, battery life is also a problem. Everybody knows that files in FLAC eat much more battery life than mp3 files.
 Until they invent new form of storing energy, new battery. I don't think its viable to even consider 24bit at least not for portable use. Currently the only option would be to increase battery size.
 That would affect the price of portable devices.
  
 On a side note, it has always amused me how people have to constantly load smart phones because the battery wont last long.
 Personally i consider that unacceptable. My current phone i load only once a month and it lasts even when i use it every day.


----------



## groovyd

in theory 48khz allows for a 24khz bandwidth of which they say humans only have about 20khz range of hearing.  so the extra 4k would be for good measure but the reality is all of this makes a number of assumptions that are not true in a real system.
  
 recognizing nyquist theory is one thing but if you think for a second about how that sampled waveform will look you see that 2 samples per cycle is really not enough even at a minimum.  take a 1hz sine wave and sample it 2 times per cycle.  it is entirely possible that both of your samples fall at EXACTLY the two zero crossings that occur in every cycle once on the way up and again on the way down and your reconstructed output becomes a flat line. by this it becomes pretty clear to see that 2 samples per cycle really is not enough at all to reconstruct your sine wave.  nyquist theory assumes a number of ideal conditions, none of which hold in a real sampling system.  it merely represents an absolute bottom line minimum spec requirement if you want to represent ANY of the information at a certain frequency but not necessarily all of it in all circumstances.  infact given 2 samples per wave taken at any point in the cycle i can draw for you an infinite number of possible near sine waves of different amplitudes that pass through those points.  lets assume the amplitude of the signal matters.  by raising the sample rate beyond the nyquist rate you quickly begin to remove the singularities and the multitude of waveforms that fit the samples for a given frequency and you allow breathing room for the realities of the limitations of a real sampling system, such as non-ideal pre-ADC bandpass and post-DAC reconstruction filtering.
  
if we assume 20khz is a reasonable limitation of human hearing without regard for the fact that our amp, speakers, etc further attenuate the response at these edge of spec frequencies
 then 48khz is really still an absolute minimum. in my opinion 96khz gives plenty of breathing room to counter all of the additive realities of a true system from end to end.


----------



## Digitalchkn

groovyd said:


> in theory 48khz allows for a 24khz bandwidth of which they say humans only have about 20khz range of hearing.  so the extra 4k would be for good measure but the reality is all of this makes a number of assumptions that are not true in a real system.
> 
> recognizing nyquist theory is one thing but if you think for a second about how that sampled waveform will look you see that 2 samples per cycle is really not enough even at a minimum.  take a 1hz sine wave and sample it 2 times per cycle.  it is entirely possible that both of your samples fall at EXACTLY the two zero crossings that occur in every cycle once on the way up and again on the way down and your reconstructed output becomes a flat line. by this it becomes pretty clear to see that 2 samples per cycle really is not enough at all to reconstruct your sine wave.  nyquist theory assumes a number of ideal conditions, none of which hold in a real sampling system.  it merely represents an absolute bottom line minimum spec requirement if you want to represent ANY of the information at a certain frequency but not necessarily all of it in all circumstances.  infact given 2 samples per wave taken at any point in the cycle i can draw for you an infinite number of possible near sine waves of different amplitudes that pass through those points.  lets assume the amplitude of the signal matters.  by raising the sample rate beyond the nyquist rate you quickly begin to remove the singularities and the multitude of waveforms that fit the samples for a given frequency and you allow breathing room for the realities of the limitations of a real sampling system, such as non-ideal pre-ADC bandpass and post-DAC reconstruction filtering.
> 
> ...


 
  
 As you point out Nyquist sampling is valid from DC to *less* than 1/2 sampling frequency. Perfect reconstruction of the original waveform requires a sinc interpolation filter, a mathematically perfect filter that can't be implemented in real life using physical components.  That's one reason 44.1K was chosen for redbook (and not something close to 40K sampling rate) as a tradeoff.
  
 There is plenty of signal energy beyond 20KHz in majority of recordings. Regardless of whether majority of humans can directly hear ultrasonics or not,  brickwall filtering has immediate impact on time domain waveform response. With modern processing and storage capability limiting to 44.1K makes less sense than it once did.


----------



## esldude

groovyd said:


> in theory 48khz allows for a 24khz bandwidth of which they say humans only have about 20khz range of hearing.  so the extra 4k would be for good measure but the reality is all of this makes a number of assumptions that are not true in a real system.
> 
> recognizing nyquist theory is one thing but if you think for a second about how that sampled waveform will look you see that 2 samples per cycle is really not enough even at a minimum.  take a 1hz sine wave and sample it 2 times per cycle.  it is entirely possible that both of your samples fall at EXACTLY the two zero crossings that occur in every cycle once on the way up and again on the way down and your reconstructed output becomes a flat line. by this it becomes pretty clear to see that 2 samples per cycle really is not enough at all to reconstruct your sine wave.  nyquist theory assumes a number of ideal conditions, none of which hold in a real sampling system.  it merely represents an absolute bottom line minimum spec requirement if you want to represent ANY of the information at a certain frequency but not necessarily all of it in all circumstances.  infact given 2 samples per wave taken at any point in the cycle i can draw for you an infinite number of possible near sine waves of different amplitudes that pass through those points.  lets assume the amplitude of the signal matters.  by raising the sample rate beyond the nyquist rate you quickly begin to remove the singularities and the multitude of waveforms that fit the samples for a given frequency and you allow breathing room for the realities of the limitations of a real sampling system, such as non-ideal pre-ADC bandpass and post-DAC reconstruction filtering.
> 
> ...


 
 Watch this video.  I am beginning to think it should be required viewing before one can post about digital audio.
  
 http://www.xiph.org/video/vid2.shtml
  
 Not singling you out personally, but your post contains several fallacies which are repeated millions of times.  Watch and understand this 23 minute video, and you will get why they are false.  Not false just in theory, but in actuality.  Two or more samples is enough to fully reconstruct the signal, and one and only one waveform fits any possible combination of samples as long as no frequencies exceed half the sample rate.  Yes, filters are imperfect, and reconstruction falls just a bit short of perfect theory.  But most the of the important factors have been dealt with.  We can get something like 95% or more of what is predicted by theory and put the last few percent of inaccuracies in a place where humans do not hear them.  Effectively, audibly very, very close to 100% fidelity to humans in actual use. 
  
 You can go to 96 khz if you just really, really, really want to be sure.  There certainly seems close to zero point, and beyond 96khz there is zero point.  It has never been shown in a credible repeatable test that people hear 96 khz vs 48 or even 44 khz.  Null results so far.


----------



## bigshot

Super audible frequencies are completely superfluous. They add absolutely nothing to the sound quality of music. You might as well stuff your file full of excelsior. It serves no purpose to human hearing. In fact, even the upper audible frequencies are pretty unimportant. A lot of music doesn't contain those frequencies. You can't perceive them as musical notes. A good chunk of the population can't hear all the way up to 20kHz anyway.
  
 It's just more kicking the line along a bit further just to satisfy OCD. Redbook is fully capable of reproducing sound with everything humans can hear.


----------



## Digitalchkn

esldude said:


> Two or more samples is enough to fully reconstruct the signal, and one and only one waveform fits any possible combination of samples as long as no frequencies exceed half the sample rate.


 
  
 This is a technicality, from theoretical point of view it is impossible to recover a sinosoidal with frequency of precisely half the sampling frequency. It gets folded into DC.
  
 On a different note, you shouldn't believe everything you hear on the internet. I personally have tendencies in trusting my university professor who's published numerous papers on IEEE over some confident kid with a video camera and a computer.


----------



## bigshot

I prefer well conducted listening tests over either of those!


----------



## Digitalchkn

bigshot said:


> Super audible frequencies are completely superfluous. They add absolutely nothing to the sound quality of music. You might as well stuff your file full of excelsior. It serves no purpose to human hearing. In fact, even the upper audible frequencies are pretty unimportant. A lot of music doesn't contain those frequencies. You can't perceive them as musical notes. A good chunk of the population can't hear all the way up to 20kHz anyway.
> 
> It's just more kicking the line along a bit further just to satisfy OCD. Redbook is fully capable of reproducing sound with everything humans can hear.


 
  
 I'm sure this has been discussed countless times already so this is bound to lead nowhere. As we all know truncated frequency spectrum leads to slight amounts distortion to the time-domain waveform (at least if you consider things in context of Nyquist sampling). I am sure we all can agree that there is plenty of energy beyond 20KHz.
  
 Regardless of whether it can or can't be heard directly by vast majority of the population in nominally noisy environments, I argue there is no longer an overwhelming reason not to minimize the amount of distortion introduced by modern reproduction systems.


----------



## Digitalchkn

bigshot said:


> I prefer well conducted listening tests over either of those!


 
  
 Theory provides us with a foundation upon which we can build. It's a means to close the loop - to understand the problem top to bottom.
 Pure theory has no value if it can't be translated into a tangible thing. A tangible thing has no weight if it is unexplainable through a common foundation. Without a satisfactory basis, all discussion becomes merely crossfire of opinions.


----------



## bigshot

I'm putting together a stereo system to listen to music. The ultimate goal is to listen to music. Theory is great for helping to solve problems, when problems exist. But for the purposes of listening to music, if I can't hear it, it isn't a problem. Armchair Einsteins can spend all their time thinking about sound they can't hear. I would rather spend that time listening to music.


----------



## bigshot

digitalchkn said:


> I'm sure this has been discussed countless times already so this is bound to lead nowhere. As we all know truncated frequency spectrum leads to slight amounts distortion to the time-domain waveform (at least if you consider things in context of Nyquist sampling). I am sure we all can agree that there is plenty of energy beyond 20KHz.


 
  
 Listening tests have shown that super audible frequencies add absolutely nothing to the sound quality of recorded music, even when a person can perceive the frequencies as sound pressure in test tones. In fact, there was one test where people didn't even think that sound quality of music was greatly affected by rolling the entire top octave of audible sound off. There is not plenty of energy beyond 20kHz with acoustic instruments. That's beyond any fundamental and beyond the important harmonics. Even if we could hear beyond 20kHz, those upper frequencies would probably be inaudible anyway due to masking by much louder lower fundamentals and harmonics.


----------



## Digitalchkn

bigshot said:


> There is not plenty of energy beyond 20kHz with acoustic instruments.


 
  
 I have spectrum analyzed lots of commercial "hi-rez" classical and jazz and saw that modern systems can pick up spectral content to no less than 30KHz. Even older recordings made during the infancy of magnetic medium have a positive SNR beyond 20KHz.
  
 Basically if you take your redbook audio  recording of your acoustic instrument and your 22.05KHz frequency point is wiggling in your spectrum analyzer then there bound to be energy beyond 22.05KHz. It's a different debate whether you or I are able to hear a test tone up there or not.
  
 As an engineer, I ask myself how much deterministic energy is up there and if it's sufficiently large then I would want to capture it.  Oddly enough, I have yet to see some  analysis tool that plots integrated energy(power) over frequency as it applies to audio applications. That would allow one to determine 90 or 95% bandwidth requirements.


----------



## bigshot

digitalchkn said:


> Basically if you take your redbook audio  recording of your acoustic instrument and your 22.05KHz frequency point is wiggling in your spectrum analyzer then there bound to be energy beyond 22.05KHz. It's a different debate whether you or I are able to hear a test tone up there or not.


 
  
 The stuff I can hear is what matters to me. I'm not putting together a stereo system for my dogs and the bats in the attic to enjoy. Besides, the dogs don't dig Brubeck. (I haven't asked the bats.)


----------



## Digitalchkn

bigshot said:


> The stuff I can hear is what matters to me. I'm not putting together a stereo system for my dogs and the bats in the attic to enjoy.


 
  
 Sure. That's why, as a consumer, you make a choice. Your choice may differ from choice of others who may find benefit from a, say, 24/48 setup. However, your choice need not be universally accepted by all. I feel it is important for others to do their homework prior to making this choice.  I think this forum should be a place for all to learn and understand the basics of what's involved and then make an educated choice based on this information. Assertive messages to the effect of "I can't tell a difference therefore you shouldn't either" is of not much help to anyone.


----------



## Tuco1965

bigshot said:


> The stuff I can hear is what matters to me. I'm not putting together a stereo system for my dogs and the bats in the attic to enjoy. Besides, the dogs don't dig Brubeck. (*I haven't asked the bats.*)


 
  
 You'd need to get above 20khz.


----------



## nick_charles

digitalchkn said:


> I have spectrum analyzed lots of commercial "hi-rez" classical and jazz and saw that modern systems can pick up spectral content to no less than 30KHz. Even older recordings made during the infancy of magnetic medium have a positive SNR beyond 20KHz.
> 
> Basically if you take your redbook audio  recording of your acoustic instrument and your 22.05KHz frequency point is wiggling in your spectrum analyzer then there bound to be energy beyond 22.05KHz. It's a different debate whether you or I are able to hear a test tone up there or not.
> 
> As an engineer, I ask myself how much deterministic energy is up there and if it's sufficiently large then I would want to capture it.  Oddly enough, I have yet to see some  analysis tool that plots integrated energy(power) over frequency as it applies to audio applications. That would allow one to determine 90 or 95% bandwidth requirements.


 
  
 The Balinese Gamelan has "musical" energy up to about 50 KHz and a cymbal crash can have _energy above noise_ up to 102 KHz with 40% of it above 20 KHz (according to James Boyk - http://www.cco.caltech.edu/~boyk/spectra/spectra.htm)  _if played absurdly loudly,_ to make sure you do not lose this would necessitate sampling at a rate in excess of 200 KHz - a bit like making sure your digicam captures gamma rays


----------



## ab initio

nick_charles said:


> The Balinese Gamelan has "musical" energy up to about 50 KHz and a cymbal crash can have _energy above noise_ up to 102 KHz with 40% of it above 20 KHz (according to James Boyk - http://www.cco.caltech.edu/~boyk/spectra/spectra.htm)  _if played absurdly loudly,_ to make sure you do not lose this would necessitate sampling at a rate in excess of 200 KHz - a bit like making sure your digicam captures gamma rays


 

 Hey, If there is a supernova when i'm taking a scenic photograph, I sure as heck want to capture all those xrays, gamma rays, neutrinos, and high energy particles that come screaming in from space. I hear they really add to the sparkle and shine in some of the landscapes panoramas.
  
 Cheers


----------



## esldude

digitalchkn said:


> This is a technicality, from theoretical point of view it is impossible to recover a sinosoidal with frequency of precisely half the sampling frequency. It gets folded into DC.
> 
> On a different note, you shouldn't believe everything you hear on the internet. I personally have tendencies in trusting my university professor who's published numerous papers on IEEE over some confident kid with a video camera and a computer.


 

 Okay, so with perfect filtering you can fully reconstruct 22,049 hz, but not 22,050.  Glad you pointed it out as that hertz would have been terrible to lose.  Lack of theoretically perfect filtering is why we have a transition band.  Full response to 20 khz and close enough to full rejection not to matter at 22.05 khz.  The transition band can have some things going on, but that is where putting the artifacts of less than theoretical perfection is a place we aren't going to hear. 
  
 As for calling Monty Montgomery some confident kid with a vidcam and computer, one always must tar the messenger when one has no reasoning to refute.   Of course what do you do when the widely published in IEEE prof also agrees with the fellow on the video?
  
 One reason the video is instructive even to those who don't know, understand or care to be bothered by the theory is how it is all checked with quality equipment in the analog realm.  Signals are generated with very high quality, low distortion analog sources, undergo AD then DA conversion and checked again with all analog low distortion scopes and frequency analyzers.  Most of the most common claims about the inabilities or distortions of digital audio are shown to not be so at all. If you can use analog at both ends and show these myths of digital inadequacy for what they are...fallacious imagination it wouldn't matter if the video was by a kindergartner or a 100 year old Nobel prize winning professor.  Myths are debunked either way.


----------



## Digitalchkn

>


 
  


esldude said:


>





> One reason the video is instructive even to those who don't know, understand or care to be bothered by the theory is how it is all checked with quality equipment in the analog realm.


 
  
 Can we elaborate on the term "quality equipment"? I hope there is an understanding that the dynamic range of waveforms shown on analog scopes is nowhere near that of even 16 bits.
  
 There is no doubt in my mind the video is a great introduction into the subject. There are other details at play that requires more in-depth look. Hence, professors in IEEE.


----------



## groovyd

esldude said:


> Watch this video.  I am beginning to think it should be required viewing before one can post about digital audio.
> 
> http://www.xiph.org/video/vid2.shtml
> 
> ...


 
  
 Having passed my masters in digital signal processing with flying colors I respectfully disagree with you and that link.  I can easily fit an infinite number of near sine waves into a set of samples at twice its frequency without even introducing the concept of sample phase.  Introducing sample phase gives me yet another infinite set of this time absolutely perfect sine waves to choose from.  Of course this assumes nobody cares about phase either in addition to amplitude.  In fact, as sample phase approaches integer multiples of pi the variety of near-perfect sine waves that fit the samples approaches infinity.  What this means is your filter must be dead nuts perfect to even have a chance of reconstructing one of an infinite number of phase shifted sine waves that 'might' be correct but absolutely is not with inverse probability to sample phase.  
  
 Sure, mathematically only one wave fits if: 1) sample phase is not a multiple of pi and 2) it is an absolute perfect sine wave and 3) you are assuming perfect brick wall filters with zero phase shift.
  Unfortunately none of these conditions even remotely represent a real system in any way.  Yours is a straw man argument that I won't argue with anymore... you can make believe whatever you want.


----------



## castleofargh

groovyd said:


> in theory 48khz allows for a 24khz bandwidth of which they say humans only have about 20khz range of hearing.  so the extra 4k would be for good measure but the reality is all of this makes a number of assumptions that are not true in a real system.
> 
> recognizing nyquist theory is one thing but if you think for a second about how that sampled waveform will look you see that 2 samples per cycle is really not enough even at a minimum.  take a 1hz sine wave and sample it 2 times per cycle.  it is entirely possible that both of your samples fall at EXACTLY the two zero crossings that occur in every cycle once on the way up and again on the way down and your reconstructed output becomes a flat line. by this it becomes pretty clear to see that 2 samples per cycle really is not enough at all to reconstruct your sine wave.  nyquist theory assumes a number of ideal conditions, none of which hold in a real sampling system.  it merely represents an absolute bottom line minimum spec requirement if you want to represent ANY of the information at a certain frequency but not necessarily all of it in all circumstances.  infact given 2 samples per wave taken at any point in the cycle i can draw for you an infinite number of possible near sine waves of different amplitudes that pass through those points.  lets assume the amplitude of the signal matters.  by raising the sample rate beyond the nyquist rate you quickly begin to remove the singularities and the multitude of waveforms that fit the samples for a given frequency and you allow breathing room for the realities of the limitations of a real sampling system, such as non-ideal pre-ADC bandpass and post-DAC reconstruction filtering.
> 
> ...


 
 I tend to disagree, but let's say you're right, bad luck we get only one or even no usable sample, then your example would make some crap for 1 and only 1 period of a 22.05khz sine. a sine that will be cut out by the low pass filter anyway.
 and for everything lower, your example becomes false as every other frequencies will get more than 2 points of reference with 44.1khz. that 1khz sine will get 44 samples, it's gonna be hard to misinterpret that one when 2 could be enough.
 I am still in favor for 48khz to be sure that no one will mess up the low pass filter. but it's more as a security against pieces of gear done badly than a real need. doubling everything for "just in case" peace of mind at frequencies most of us don't hear, on audio systems that roll them off or cut them out, I fail to see the point.
  


digitalchkn said:


> bigshot said:
> 
> 
> > There is not plenty of energy beyond 20kHz with acoustic instruments.
> ...


 
 I understand the idea, but I'm completely against it for the exact reason you want it. I see that as an added chance to damage my hearing, and even though the sound pressure and duration are what count most, high freqs have more energy for the same amplitude right? I remember reading something about violin players and hearing loss where instruments making high frequencies were seen as more dangerous overall. and some hearing protection dealer saying that what was most important was to stop high freqs when they made some for me.
 I really do not want my ears to receive too much of something I don't hear, but could still had damages on the long run.
  
 well in my case it's not a real problem as pretty much none of my headphones/IEMs are able to deal with 20khz anyway, and most IEMs I've owned were almost dead silent above 14 or 15khz ^_^.


----------



## bigshot

digitalchkn said:


> Sure. That's why, as a consumer, you make a choice. Your choice may differ from choice of others who may find benefit from a, say, 24/48 setup.


 
  
 My dog can't afford a stereo.


----------



## Digitalchkn

bigshot said:


> My dog can't afford a stereo.


 
  
 Didn't know your dog has the final say on your purchasing decisions


----------



## bigshot

castleofargh said:


> I am still in favor for 48khz to be sure that no one will mess up the low pass filter. but it's more as a security against pieces of gear done badly than a real need. doubling everything for "just in case" peace of mind at frequencies most of us don't hear, on audio systems that roll them off or cut them out, I fail to see the point.


 
  
 Isn't it nice that we all make different compromises... and all those compromises are beyond the range of our ability to hear! It makes it easy. Any decision on where to compromise is the right decision. (Dogs and bats notwithstanding, of course.)


----------



## groovyd

castleofargh said:


> ... I fail to see the point.
> 
> pretty much none of my headphones/IEMs are able to deal with 20khz anyway, and most IEMs I've owned were almost dead silent above 14 or 15khz ^_^.


 
  
 agreed.
  
 and also in failing to see the point of manufacturers saving a few cents to not produce equipment that outperforms the music it will be used with.  ofcourse this is head-fi and some people do manage to find crystalized gold connectors that are dubious at best to improve the sound in any meaningful way when adding a few more bits to the source file would have a much more dramatic impact for let's just round down and call it free.  in this day and age there is no good argument to not increase both the sample rate and bit depth of recorded music as a common standard.


----------



## bigshot

digitalchkn said:


> Didn't know your dog has the final say on your purchasing decisions


 
  
 I buy my own stereo and he mooches music off of me. Haven't heard him complain about the lack of sound that is completely inaudible to human ears yet.


----------



## Digitalchkn

castleofargh said:


> I understand the idea, but I'm completely against it for the exact reason you want it. I see that as an added chance to damage my hearing, and even though the sound pressure and duration are what count most, high freqs have more energy for the same amplitude right? I remember reading something about violin players and hearing loss where instruments making high frequencies were seen as more dangerous overall. and some hearing protection dealer saying that what was most important was to stop high freqs when they made some for me.
> I really do not want my ears to receive too much of something I don't hear, but could still had damages on the long run.


 
  
 Luckily high frequency content tends to falls off. And I am not talking about your sound system reproducing gamma rays. But yeah, point taken.


----------



## bigshot

groovyd said:


> in this day and age there is no good argument to not increase both the sample rate and bit depth of recorded music as a common standard.


 
  
 Isn't "you can't hear the difference anyway" a good enough argument?
  


digitalchkn said:


> Luckily high frequency content tends to falls off.


 
  
 Wait a minute! A few posts back didn't you argue that there was plenty of content up there?


----------



## Digitalchkn

groovyd said:


> in this day and age there is no good argument to not increase both the sample rate and bit depth of recorded music as a common standard.


 
  
 ++


----------



## Digitalchkn

bigshot said:


> Haven't heard him complain about the lack of sound that is completely inaudible to human ears yet.


 
  
 Well I guess he's not going to post on this forum anytime soon then.


----------



## Digitalchkn

bigshot said:


> Wait a minute! A few posts back didn't you argue that there was plenty of content up there?


 
  
 There is plenty of content beyond 20KHz. It just falls off ... eventually.


----------



## bigshot

At some point it becomes radio waves and your stereo can receive it again, right?!


----------



## bigshot

digitalchkn said:


> Well I guess he's not going to post on this forum anytime soon then.


 
  
 Here is my woofer.


----------



## Digitalchkn

bigshot said:


> Here is my woofer.


 
  
 Hmmm. Not up to audiophile standard just yet. Maybe after some burn-in frequency response will change to handle >20KHz.


----------



## Digitalchkn

bigshot said:


> At some point it becomes radio waves and your stereo can receive it again, right?!


 
  
 My stereo now only receives hi-rez.


----------



## SilverEars

bigshot said:


> Here is my woofer.


 
 What happened to it's eyes?  What is this thing called?  Is it yours bigshot?  Have you burn it in yet?


----------



## esldude

groovyd said:


> Having passed my masters in digital signal processing with flying colors I respectfully disagree with you and that link.  I can easily fit an infinite number of near sine waves into a set of samples at twice its frequency without even introducing the concept of sample phase.  Introducing sample phase gives me yet another infinite set of this time absolutely perfect sine waves to choose from.  Of course this assumes nobody cares about phase either in addition to amplitude.  In fact, as sample phase approaches integer multiples of pi the variety of near-perfect sine waves that fit the samples approaches infinity.  What this means is your filter must be dead nuts perfect to even have a chance of reconstructing one of an infinite number of phase shifted sine waves that 'might' be correct but absolutely is not with inverse probability to sample phase.
> 
> Sure, mathematically only one wave fits if: 1) sample phase is not a multiple of pi and 2) it is an absolute perfect sine wave and 3) you are assuming perfect brick wall filters with zero phase shift.
> Unfortunately none of these conditions even remotely represent a real system in any way.  Yours is a straw man argument that I won't argue with anymore... you can make believe whatever you want.


 

 Funny, how you can put all these signals through AD/DA conversion and recover them without problems.  If you have an exact 22.050 khz sine wave, and you put another one in there, you could say you have two, of course you would have one twice as big.  Unless you put in one out of phase in which case the samples of the new composite waveform would be different than before the addition.   So no without a masters in DSP I don't get what you are positing here.  I do get that the equipment works as it is said to work.


----------



## esldude

digitalchkn said:


> Can we elaborate on the term "quality equipment"? I hope there is an understanding that the dynamic range of waveforms shown on analog scopes is nowhere near that of even 16 bits.
> 
> There is no doubt in my mind the video is a great introduction into the subject. There are other details at play that requires more in-depth look. Hence, professors in IEEE.


 

 We could.  I have pointed to the video, they describe exactly what they are using.  So do I need to explain, describe, digest the details and spell them out for you or can you just watch it?  I take it by your comments you have not watched it.  Comments about the video make more sense when you have seen it.


----------



## groovyd

esldude said:


> Funny, how you can put all these signals through AD/DA conversion and recover them without problems.  If you have an exact 22.050 khz sine wave, and you put another one in there, you could say you have two, of course you would have one twice as big.  Unless you put in one out of phase in which case the samples of the new composite waveform would be different than before the addition.   So no without a masters in DSP I don't get what you are positing here.  I do get that the equipment works as it is said to work.


 
  
 Yeah it is funny how it works... if you put two signals of the exact same frequency and amplitude in there out of phase you would indeed hear nothing.  all three things, phase, frequency, and amplitude matter.  simple math really.


----------



## esldude

groovyd said:


> Yeah it is funny how it works... if you put two signals of the exact same frequency and amplitude in there out of phase you would indeed hear nothing.  all three things, phase, frequency, and amplitude matter.  simple math really.


 

 By out of phase I didn't mean necessarily inverse phase, just not lined up in phase with each other. Could have been 1 degree out or 5 degrees or whatever including 180 out. But yes that part of the math is quite simple.


----------



## bigshot

digitalchkn said:


> Hmmm. Not up to audiophile standard just yet. Maybe after some burn-in frequency response will change to handle >20KHz.


 
  
 It's designed for near field laps


----------



## bigshot

silverears said:


> What happened to it's eyes?  What is this thing called?  Is it yours bigshot?  Have you burn it in yet?


 
  
 Believe it or not, the eyes are open in this picture. They are just teeny and tight together. This is "Pickles" and she burns me plenty!


----------



## bigshot

esldude said:


> By out of phase I didn't mean necessarily inverse phase, just not lined up in phase with each other. Could have been 1 degree out or 5 degrees or whatever including 180 out.


 
  
 Do you know what the just detectable threshold (JDT) for that is? Do you know what kind of phase error you are talking about? These are the two questions that are the most important that theoreticians rarely get around to asking.


----------



## groovyd

esldude said:


> By out of phase I didn't mean necessarily inverse phase, just not lined up in phase with each other. Could have been 1 degree out or 5 degrees or whatever including 180 out. But yes that part of the math is quite simple.


 
  
 180 out of phase will give silence.  using matlab you should see what waveforms you get adding two identical waves only out of phase with each other.  try it for a variety of different phases.  i think what you are imagining is how the ears are able to remove the significance of phase between signals to some extent, but of course not entirely. coming from different locations like speakers and into different ears at different times your brain is able to pretend phase is not as important as it is however the effects of signals being out of phase is certainly heard for example many have witnessed nodes of high and low volume depending on the placement of their subwoofers in a room and relative to each other.  invert one pair of wires going into one ear of your headphones and you will quite clearly hear the difference.  certainly 2 signals 180 degrees out of phase being put out by the same transducer and you will not hear a thing. this is actually the basis for noise canceling circuitry of some headphones.


----------



## ab initio

bigshot said:


> Here is my woofer.


 
  
 Got a picture of a tweeter?
  
 Cheers


----------



## esldude

groovyd said:


> 180 out of phase will give silence.  using matlab you should see what waveforms you get adding two identical waves only out of phase with each other.  try it for a variety of different phases.  i think what you are imagining is how the ears are able to remove the significance of phase between signals to some extent, but of course not entirely. coming from different locations like speakers and into different ears at different times your brain is able to pretend phase is not as important as it is however the effects of signals being out of phase is certainly heard for example many have witnessed nodes of high and low volume depending on the placement of their subwoofers in a room and relative to each other.  invert one pair of wires going into one ear of your headphones and you will quite clearly hear the difference.  certainly 2 signals 180 degrees out of phase being put out by the same transducer and you will not hear a thing. this is actually the basis for noise canceling circuitry of some headphones.


 

 Don't know what you imagine I am imagining, but it appears you are crossed up on it there.  Two sine waves of the same frequency will add to a larger or smaller wave of the same frequency depending on the amount out of phase they are.  At 180 out it goes to nothing, and at 360 or multiples it is doubled up.  If it happens at 22 khz I won't hear it either way.


----------



## SilverEars

bigshot said:


> Believe it or not, the eyes are open in this picture. They are just teeny and tight together. This is "Pickles" and she burns me plenty!


 
 What is Pickle's nationality?  Breed? Is it a mut?


----------



## castleofargh

groovyd said:


> esldude said:
> 
> 
> > By out of phase I didn't mean necessarily inverse phase, just not lined up in phase with each other. Could have been 1 degree out or 5 degrees or whatever including 180 out. But yes that part of the math is quite simple.
> ...


 
  
 1/ you're taking an example for right vs left and stuff heard probably far under 16khz. so not really our problem here, be it 44khz or 96khz.
 2/ let's take our extreme case (the very useless one in music) 22khz. having 4 samples instead of 2 would help in what proportion for phase error?
  
 from what I remember the most important for our human ears, is to avoid intermodulation distortion as much as possible as it is not at all musical. but for that purpose, the dac will add as many samples as needed by itself when needed to deal with aliasing. I'm not sure I see the benefits of 96khz on the original track.


----------



## bigshot

silverears said:


> What is Pickle's nationality?  Breed? Is it a mut?


 
  
 She is a Pomeranian... also known as a German Spitz.


----------



## bigshot

ab initio said:


> Got a picture of a tweeter?


 
  
 this one is my tweeter... Schlitzee. She is a PomChi (half pomeranian half chihuahua)


----------



## RazorJack

groovyd said:


> Having passed my masters in digital signal processing with flying colors I respectfully disagree with you and that link.  I can easily fit an infinite number of near sine waves into a set of samples at twice its frequency without even introducing the concept of sample phase.  Introducing sample phase gives me yet another infinite set of this time absolutely perfect sine waves to choose from.  Of course this assumes nobody cares about phase either in addition to amplitude.  In fact, as sample phase approaches integer multiples of pi the variety of near-perfect sine waves that fit the samples approaches infinity.  What this means is your filter must be dead nuts perfect to even have a chance of reconstructing one of an infinite number of phase shifted sine waves that 'might' be correct but absolutely is not with inverse probability to sample phase.
> 
> Sure, mathematically only one wave fits if: 1) sample phase is not a multiple of pi and 2) it is an absolute perfect sine wave and 3) you are assuming perfect brick wall filters with zero phase shift.
> Unfortunately none of these conditions even remotely represent a real system in any way.  Yours is a straw man argument that I won't argue with anymore... you can make believe whatever you want.


 
  
 What parts in the video do you disagree with?
  
 And how is any of the above relevant to 44.1 kHz as Nyquist frequency being adequate for (human... sorry dogs!) audio applications?


----------



## groovyd

esldude said:


> Don't know what you imagine I am imagining, but it appears you are crossed up on it there.  Two sine waves of the same frequency will add to a larger or smaller wave of the same frequency depending on the amount out of phase they are.  At 180 out it goes to nothing, and at 360 or multiples it is doubled up.  If it happens at 22 khz I won't hear it either way.


 
 Originally Posted by *esldude* /img/forum/go_quote.gif
  

 By out of phase I didn't mean necessarily inverse phase, just not lined up in phase with each other. Could have been 1 degree out or 5 degrees or whatever including 180 out. But yes that part of the math is quite simple.
  
  
  
 Don't know what you are imagining that you are imagining either...


----------



## groovyd

razorjack said:


> What parts in the video do you disagree with?
> 
> And how is any of the above relevant to 44.1 kHz as Nyquist frequency being adequate for (human... sorry dogs!) audio applications?


 
  
 sorry but this argument is not worth my time...


----------



## Digitalchkn

Quote:


digitalchkn said:


> Can we elaborate on the term "quality equipment"? I hope there is an understanding that the dynamic range of waveforms shown on analog scopes is nowhere near that of even 16 bits.
> 
> There is no doubt in my mind the video is a great introduction into the subject. There are other details at play that requires more in-depth look. Hence, professors in IEEE.


 
  
  
 Quote:


esldude said:


> We could.  I have pointed to the video, they describe exactly what they are using.  So do I need to explain, describe, digest the details and spell them out for you or can you just watch it?  I take it by your comments you have not watched it.  Comments about the video make more sense when you have seen it.


 
  
 I used to work for a well known scope manufacturer. I use higher end versions of this equipment on a regular basis for a living.  Are you still certain you'd like to get into the details of it?
  
 As I said before, this is a good introductory material intended for noobs on the topic. For those who have studies this topic academically and/or work in related fields, this video has nothing to offer and in fact does not paint the full picture. Acceptance of this as gospel simply indicates illiteracy on the subject.  For more thorough introductory material at an academic level I highly recommend "Discrete-Time Signal Processing" by A. Oppenheim and R. Schafer 3rd edition, in particular Ch 1,2,4.  Please review that book prior to posting any arguments about sampling theory and DSP in general on this forum. Until then I have little interest in spending my time reviewing what is already well established knowledge.


----------



## bigshot

groovyd said:


> sorry but this argument is not worth my time...


 
  
 Are we just supposed to assume that just because super audible frequencies exist, we need to make sure our stereo systems are capable of reproducing them? Just because computers and hard drives are cheaper and bigger than they used to be, should we allot massive amounts of bandwidth and processing power to things we can't even hear? That makes no sense whatsoever.
  


digitalchkn said:


> For more thorough introductory material at an academic level I highly recommend "Discrete-Time Signal Processing" by A. Oppenheim and R. Schafer 3rd edition, in particular Ch 1,2,4.  Please review that book prior to posting any arguments about sampling theory and DSP in general on this forum.


 
  
 Ha! In order to use the English language effectively to communicate, I recommend reading all of Shakespere's works. Please read all of the plays and commit the sonnets to memory prior to posting anything on this forum.


----------



## Digitalchkn

bigshot said:


> Ha! In order to use the English language effectively to communicate, I recommend reading all of Shakespere's works. Please read all of the plays and commit the sonnets to memory prior to posting anything on this forum.


 
  
  
 That's not right. I only recommended a few chapters not the whole thing.  Besides, I talk to you in prose and you talk back to me in slang. Some conversation!


----------



## bigshot

I'm reading Hammett right now.


----------



## Digitalchkn

bigshot said:


> I'm reading Hammett right now.


 
  
 Is that Kirk Hammett? Of Metallica?


----------



## ab initio

groovyd said:


> sorry but this argument is not worth my time...


 
  
 Your previous explanations are confusing and unclear what exactly you are talking about, especially when you don't include any graphics to illustrate the point you are trying to make, nor do you link or cite any sources where folks can go for a more in-depth understanding.
  
 If I understand correctly, the issue at hand is regarding discrete-time sampling rates and the reconstruction of band-limited signals (and finite energy---like all real signals). In this case, in the xiph.org video, monty (correctly) explains how a real band limited signal is mathematically perfectly defined by sampling above the signal's nyquist frequency. In other words, given a set of discrete samples, there is exactly one and only one band-limited signal which perfectly intersects all of the sample points. This concept was clearly illustrated in monty's video. He further clarifies the point by demonstrating how the phase of the signal is perfectly captured by discrete time sampling. Again, this is the case because there is a one-to-one relationship between a discrete-time sampled waveform and its corresponding band-limited continuous waveform.
  
 You need to define what you mean by "near sine waves" and relate the difference between a actual sine wave and whatever this other thing is that you are talking about, and then relate what additional Fourier coefficients your near sine waves have in addition to the fundamental. Then you need to relate how those harmonics relate to the concept of a band-limited signal. If you've added Fourier components that exceed one half of the sampling frequency, you have violated the Nyquist criterion and you are now discussing a different topic than covered in the video.
  
 The logical flow of the arguments is:
  
 1) there is a reasonably well defined frequency range of human hearing ( e.g., 20Hz -- 20kHz ). There may be exceptions, but by a few % of the Hz one way or another, and certainly not by a factor of 2 in either direction.
  
 2) neither subsonic nor ultrasonic frequencies contribute to the audibility of sounds (this is by definition, "subsonic" meaning lower than what can be heard, etc.)
 the corollary here is that all sound that can be heard fits neatly into the band-limited frequency range defined by human hearing (e.g., 20Hz--20kHz).
  
 3) discrete-time sampled sound at frequencies of 44.1kHz or more is sufficient to mathematically define all the energy in a signal band-limited in the human audible range. [1]
  
 Therefore, sampling above twice the highest frequency in the band-limited signal is mathematically sufficient to fully define the continuous time waveform.
  
  
 [1] - http://en.wikipedia.org/wiki/Sampling_theorem
 If the wikipedia source is wrong, I encourage you to use your master's degree-level knowledge on the topic to corrected or amend the article. However, in this case, I don't anticipate that you will overturn the fundamental concept of the Nyquist-Shannon sampling theorem. Perhaps others can chime in with sources of the other bullet points on the list as I am quite busy at work now. Maybe I can fill it in later if there is interest, but the sources are already linked in other parts of the forum, so you can use your googlefu to find it.
  
 Cheers


----------



## Digitalchkn

ab initio said:


> If I understand correctly, the issue at hand is regarding discrete-time sampling rates and the reconstruction of band-limited signals (and finite energy---like all real signals).
> 
> Cheers


 
  
 It's been a little while for me so my math may be a bit rusty and I may need some help with this.  If we let x(t)=sin(2π ft), which is a real signal and plug it into the definition for finite total energy of a signal, which is given by:

 That doesn't look like E < inf. Anything I am missing?


----------



## mikeaj

digitalchkn said:


> It's been a little while for me so my math may be a bit rusty and I may need some help with this.  If we let x(t)=sin(2π ft), which is a real signal and plug it into the definition for finite total energy of a signal, which is given by:
> 
> That doesn't look like E < inf. Anything I am missing?


 
  
 It's also been a while for me, but something like a sinusoid over infinite time does have infinite energy. (Think of a voltage signal like that operating for eternity—you'd need an infinite amount of energy from some source to keep applying that voltage across a resistor or something.) That's why these are analyzed in terms of power, so divide by the time of integration. If you're not integrating over infinite time and you're not integrating something that's going to infinity in nasty ways, then the integral will be finite.
  
 Look at power spectral density, etc. and see if there is nonzero content at frequencies above Nyquist. Or just filter those frequencies out, depending on the application.
  
 Actually, I'm not quite sure if I followed the motivation for the question, but I haven't really been following the thread.


----------



## ab initio

digitalchkn said:


> It's been a little while for me so my math may be a bit rusty and I may need some help with this.  If we let x(t)=sin(2π ft), which is a real signal and plug it into the definition for finite total energy of a signal, which is given by:
> 
> That doesn't look like E < inf. Anything I am missing?


 

 Well, either a) the signal must be finite in length, (i.e., at some point for t < t_0, x(t)==0 and for t > t_end, x(t) ==0). Here's a link to a university lecture that discusses energy in signals (http://ocw.usu.edu/Electrical_and_Computer_Engineering/Signals_and_Systems/5_10node10.html). Here they require a finite length signal.
  
  
 Or b) you can talk about the signal's energy per unit time. I believe this is the more typical case. Here, Parseval's theorem is useful for showing that the energy in the signal is the same whether you calculate it in time domain or Fourier domain (http://en.wikipedia.org/wiki/Parseval%27s_theorem). One might do this by considering the energy of some periodic signal. One integrates over the period of the signal.
  
 The energy in the signal is bounded if the energy in the Fourier coefficients decrease sufficiently fast with increasing wavenumber. In the case of a band-limited signal, this is the case, because all Fourier coefficients are zero beyond the Nyquist frequency.
  
 Cheers


----------



## esldude

digitalchkn said:


> Quote:
> 
> 
> Quote:
> ...


 

 Yes, I suppose we can get into if you like. You asked what I meant by "quality equipment".  The equipment is clearly specified in the video.  If you have the idea we are acting as if we can discern 16 bit precision on a scope, well, neither I nor the video made any such claim.  By quality I meant an analog based source for highly precise, very low distortion test signals and analog based, highly precise, very low distortion spectrum analyzers.  Things that were obvious on a scope are things like the idea you don't get good clean sine waves out of AD/DA conversion using only slightly more than 2 samples per wave.  You do.  You couldn't specify it was .05% distortion looking at the scope.  Combine it with a spectrum analyzer and you can see there no large levels of distortion either.  Straightforward stuff indeed.  
  
 The premise of the video is if you don't know or don't believe digital can do what it claims as cleanly as it claims you can investigate with good analog equipment what it can and what it cannot do.  The analog results show that up to 20khz digital is as clean and accurate as it claims to be.  Of course any such introductory type video isn't in depth or detail for any tiny issue one may have with digital audio.  But one could do the same thing if they wished and investigate whichever issue they think they are having.
  
 So with your advanced knowledge and literacy in things digital what are the problems of digital processing in audio?


----------



## Digitalchkn

ab initio said:


> Well, either a) the signal must be finite in length, (i.e., at some point for t < t_0, x(t)==0 and for t > t_end, x(t) ==0). Here's a link to a university lecture that discusses energy in signals (http://ocw.usu.edu/Electrical_and_Computer_Engineering/Signals_and_Systems/5_10node10.html). Here they require a finite length signal.
> 
> 
> Or b) you can talk about the signal's energy per unit time. I believe this is the more typical case. Here, Parseval's theorem is useful for showing that the energy in the signal is the same whether you calculate it in time domain or Fourier domain (http://en.wikipedia.org/wiki/Parseval%27s_theorem). One might do this by considering the energy of some periodic signal. One integrates over the period of the signal.
> ...


 
  
  
  


mikeaj said:


> It's also been a while for me, but something like a sinusoid over infinite time does have infinite energy. (Think of a voltage signal like that operating for eternity—you'd need an infinite amount of energy from some source to keep applying that voltage across a resistor or something.) That's why these are analyzed in terms of power, so divide by the time of integration. If you're not integrating over infinite time and you're not integrating something that's going to infinity in nasty ways, then the integral will be finite.
> 
> Look at power spectral density, etc. and see if there is nonzero content at frequencies above Nyquist. Or just filter those frequencies out, depending on the application.


 
  
 Amazing how the fundamentals slip out your head after some time.  You guys more or less covered it for me. Something along the lines of a certain class of energy signals that fall into category of power signals which can still be analyzed using fourier integration without blowing up, if I recall correctly.  Gets you that delta functions as a consequence, as a sort of a limit.
  
 Anyways, this was a bit off topic. The phrase "all real signals are finite energy" triggered the question for me.


----------



## Digitalchkn

esldude said:


> Yes, I suppose we can get into if you like. You asked what I meant by "quality equipment".  The equipment is clearly specified in the video.  If you have the idea we are acting as if we can discern 16 bit precision on a scope, well, neither I nor the video made any such claim.  By quality I meant an analog based source for highly precise, very low distortion test signals and analog based, highly precise, very low distortion spectrum analyzers.  Things that were obvious on a scope are things like the idea you don't get good clean sine waves out of AD/DA conversion using only slightly more than 2 samples per wave.  You do.  You couldn't specify it was .05% distortion looking at the scope.  Combine it with a spectrum analyzer and you can see there no large levels of distortion either.  Straightforward stuff indeed.
> 
> The premise of the video is if you don't know or don't believe digital can do what it claims as cleanly as it claims you can investigate with good analog equipment what it can and what it cannot do.  The analog results show that up to 20khz digital is as clean and accurate as it claims to be.  Of course any such introductory type video isn't in depth or detail for any tiny issue one may have with digital audio.  But one could do the same thing if they wished and investigate whichever issue they think they are having.
> 
> So with your advanced knowledge and literacy in things digital what are the problems of digital processing in audio?


 
  
 In the context of this thread, this video may be interpreted too literally and one may conclude that based on the experiments that bit depths and sample rates shown are more than sufficient. Consequently, those continue to argue that conclusions in this video is gospel.
  
 Or put it another way, if I wanted to very the quality of my ADC/DAC path in my designs I would be give it a lot of thought about measurement equipment used, considering that noise floor and/or resolution of many instruments is at or worse than the levels of what I am trying to measure.


----------



## bigshot

digitalchkn said:


> In the context of this thread, this video may be interpreted too literally and one may conclude that based on the experiments that bit depths and sample rates shown are more than sufficient.


 
  
 Uh... yeah. Is there something there I'm not hearing? Is what I don't hear what we're talking about? Should I care about what I can't hear?


----------



## bigshot

digitalchkn said:


> Is that Kirk Hammett? Of Metallica?


 
  
 Dashiell. Add that to your reading list before you can use the English language!


----------



## esldude

digitalchkn said:


> In the context of this thread, this video may be interpreted too literally and one may conclude that based on the experiments that bit depths and sample rates shown are more than sufficient. Consequently, those continue to argue that conclusions in this video is gospel.
> 
> Or put it another way, if I wanted to very the quality of my ADC/DAC path in my designs I would be give it a lot of thought about measurement equipment used, considering that noise floor and/or resolution of many instruments is at or worse than the levels of what I am trying to measure.


 

 Okay, so are you saying 44.1 khz sample rates or 16 or 24 bit depth is not sufficient?  What part of digital audio's performance envelope fails to be good enough for human audio?
  
 The video is not gospel.  I do not find much to argue with considering what it shows and claims. 
  
 I understand that to verify the complete depths of your AD/DA path you need instruments cleaner than what is tested.  In the case of the video that appears to not be the case.  But it does show the basic AD/DA path is clean enough it appeared not to corrupt the quite clean analog signals in use.  Making you think the AD/DA path was at least equal or better than the analog sources.  It shows the AD/DA path is transparent to quite low levels of degradation.  Since they were using a cheap, obsolete consumer level AD/DA unit, it seems sufficient to show digital largely works as claimed.  Better implementations would most likely reach closer to the theoretical limits of a digitally sampled system.   You seem to be working pretty hard not to agree.  What is your opinion of where it falls flat? Or is that your opinion?  Or do you just dislike video presentations?


----------



## RazorJack

That video made so much sense to me, I'd say it almost put my DSP and other EE-related courses I had to follow in college to shame, when it comes to explaining a relatively complex matter (for the average Joe, not engineers) in straight forward language.
  


esldude said:


> Watch this video.  I am beginning to think it should be required viewing before one can post about digital audio.
> 
> http://www.xiph.org/video/vid2.shtml
> 
> ...


----------



## Digitalchkn

esldude said:


> Okay, so are you saying 44.1 khz sample rates or 16 or 24 bit depth is not sufficient?  What part of digital audio's performance envelope fails to be good enough for human audio?
> 
> The video is not gospel.  I do not find much to argue with considering what it shows and claims.
> 
> I understand that to verify the complete depths of your AD/DA path you need instruments cleaner than what is tested.  In the case of the video that appears to not be the case.  But it does show the basic AD/DA path is clean enough it appeared not to corrupt the quite clean analog signals in use.  Making you think the AD/DA path was at least equal or better than the analog sources.  It shows the AD/DA path is transparent to quite low levels of degradation.  Since they were using a cheap, obsolete consumer level AD/DA unit, it seems sufficient to show digital largely works as claimed.  Better implementations would most likely reach closer to the theoretical limits of a digitally sampled system.   You seem to be working pretty hard not to agree.  What is your opinion of where it falls flat? Or is that your opinion?  Or do you just dislike video presentations?


 
  
 Summarizing point? no longer a reason for 16/44 as the ultimate distribution format for consumers.
  
 To answer your first two questions we would need assign a metric to the digital audio performance captured vs a given format. Personally, I am not aware of a generally accepted metric that we all can agree on.
  
 Side-point: by now everyone has seen this video. Things works largely as claimed (surprise). The devil is in the details, which seems to me to be the crux of the matter


----------



## bigshot

digitalchkn said:


> To answer your first two questions we would need assign a metric to the digital audio performance captured vs a given format. Personally, I am not aware of a generally accepted metric that we all can agree on.


 
  
 I agree on the metric of audibility. How about you? Does that work?
  
 Ignoring everything but what human beings can hear, for the purposes of playing back music in the home is there any reason to use rates above 16/44? That seems like a really simple question to answer to me.


----------



## Digitalchkn

bigshot said:


> I agree on the metric of audibility. How about you? Does that work?
> 
> Ignoring everything but what human beings can hear, for the purposes of playing back music in the home is there any reason to use rates above 16/44? That seems like a really simple question to answer to me.


 
  
 Let's break this down first. We need to understand our bandwidth and dynamic range requirements. What is it that humans can and cannot detect? As in not resolve into music, but simply perceive as a mechanical vibrations of air. Do we have a consensus on what these requirements are?


----------



## esldude

digitalchkn said:


> Let's break this down first. We need to understand our bandwidth and dynamic range requirements. What is it that humans can and cannot detect? As in not resolve into music, but simply perceive as a mechanical vibrations of air. Do we have a consensus on what these requirements are?


 

 Well I believe there have been a couple people who could perceive 25 khz at very high levels (more than 100 db) in otherwise quiet surroundings.  A very small percentage of young adults have been shown to hear 22-24  khz at similar very high threshold levels.  That is why JJ Johnston or Robert Stuart say 60-65 khz sampling rates would have been the most possibly needed if one wishes to leave nothing on the table.  Nyquist for 25 khz with room for filtering easily.  That is needed however for probably less than .1% of the adult population. 
  
 As for dynamic range, it is few and far between finding electronics with more than 120 db SNR.  So 24 bits should be fine, theoretically there may be situations where 16 could fall short.   It will get mentioned that 120 db is the threshold of pain.  I don't know if you could be expected to hear more loudly than that even though it hurts.  I believe the answer is yes.  I also believe it makes no sense to decide to produce audio playback with that extra pain inducing dynamic range. 
  
 So you could say we need response to 25 khz and 20 bits to be sure of full transparency without question.  For such people 88/24 should be more than enough.   Does it make sense to have standard delivery mediums that more than double file size to satisfy the less than 1% who could hear it likely far less than 1% of their time listening?


----------



## groovyd

digitalchkn said:


>





> I highly recommend "Discrete-Time Signal Processing" by A. Oppenheim and R. Schafer 3rd edition


 
  
 +1 excellent graduate level text


----------



## groovyd

digitalchkn said:


> The devil is in the details, which seems to me to be the crux of the matter


 
  
 +1 - actually entertaining that such a group of people who pretend to know signal processing to the point of arguing over equations that have nothing to do with the reality of the topic don't even recognize where their misplaced theories fall apart in applied science.  it is good though to see at least one other true signal processing guy around


----------



## bigshot

digitalchkn said:


> Let's break this down first. We need to understand our bandwidth and dynamic range requirements. What is it that humans can and cannot detect? As in not resolve into music, but simply perceive as a mechanical vibrations of air. Do we have a consensus on what these requirements are?


 
  
 Here's a start. If you have anything to add, I can update it.
  
 http://www.head-fi.org/t/645851/the-most-important-spec-sheet-the-human-ear


----------



## bigshot

esldude said:


> Well I believe there have been a couple people who could perceive 25 khz at very high levels


 
  
 As sound pressure, not music.


----------



## ab initio

digitalchkn said:


> Amazing how the fundamentals slip out your head after some time.  You guys more or less covered it for me. Something along the lines of a certain class of energy signals that fall into category of power signals which can still be analyzed using fourier integration without blowing up, if I recall correctly.  Gets you that delta functions as a consequence, as a sort of a limit.
> 
> Anyways, this was a bit off topic. The phrase "all real signals are finite energy" triggered the question for me.




What on earth are you rambling About?


----------



## esldude

bigshot said:


> As sound pressure, not music.


 

 regarding hearing 25 khz.  They heard 25 khz, 24 khz sine waves etc etc at levels with a threshold of 102-107 db if my memory isn't faulty.  So I am assuming they could hear it as a part of music were music to have something up there and nearly nothing near it.  I would think it takes very little in lower frequency content to mask it altogether.  Of course the same is true of most middle age people in the 12-15 khz region.


----------



## RazorJack

groovyd said:


> digitalchkn said:
> 
> 
> > The devil is in the details, which seems to me to be the crux of the matter
> ...


 
  
 Show me one scientific test where you (or anyone) discerns CD quality of an album from >16 bit, >44.1 kHz sampling frequency.
  
 Let me guess, not worth your time?
  
 The devil is not in the details. The crux of the matter is between your ears.


----------



## groovyd

razorjack said:


> Show me one scientific test where you (or anyone) discerns CD quality of an album from >16 bit, >44.1 kHz sampling frequency.
> 
> Let me guess, not worth your time?
> 
> The devil is not in the details. The crux of the matter is between your ears.


 
  
 lol


----------



## magiccabbage

what a thread


----------



## ab initio

Look, if you guys want to move the discussion forward in a constructive fashion, you can start by contributing scientific principals and linking to sources. The best sources are places like wikipedia, wolfram, and free online webpages from universities, etc. Claiming you have some textbook that somehow invalidates the sampling theorem is not helping. You have to provide the arguments to support your claim---in this case math, since you are arguing against a mathematical theory---and provide third-party sources that establish your arguments as valid. People are here to learn, so they are very receptive to intelligent arguments. Until you provide said intelligent argument, you've got no basis for claiming the sampling theorem is invalid. Instead, you are either a) providing a lot of evidence that you don't understand the concepts yourself, b) arguing a strawman, or c) trolling.
  
 You are nay-saying a well established theorem, but refusing to actually explain anything in a coherent fashion, nor provide any sources that back your claim. If your argument has any merit then it would prove invaluable to support your argument with evidence. "I have a MS in computer engineering" is not evidence. If you would like learn more about how to refute an argument, I encourage you to read the article linked in my sig. Furthermore, there is another thread linking to an explanation of common fallacies, so now you know how to avoid making statements and arguments that fail to further this discussion.
  
 If you want to make a point about the Nyquist-shannon sampling theorem being invalid, you could go a long way toward supporting your claim if you can point us to a proof, example, or explanation on what breaks down. You might (or might not) find a publicly available source here.
  
 Cheers
  
 PS for those who are interested, the MIT course notes on Discrete-Time Signal Processing is freely available online, and is given by Dr. Oppenheim.


----------



## bigshot

The bottom line for them is "There's no reason not to use high bitrates- hard drives are cheap, processors are fast." But it's also undeniably true that there is no reason to use them either. They can't point to anything that says there is an audible difference. That's the stalemate here. Some of us listen with our ears, and others care more about measurements on paper that are beyond our ability to hear.


----------



## a_recording

Thought you guys might be interested, I made a video about 24 bit audio drawing from the Xiph.org resources and others. It was this thread which made me interested in this topic in the first place!
  
 There's lots of little inaccuracies in the video but hopefully will give people a good primer about 24 bit.


----------



## Danz03

All these arguments! Let's settle them with a blind test, download the 4 files from the links below.
  
https://www.dropbox.com/s/4wszs6k2s146c4g/GA%2024A.aif
https://www.dropbox.com/s/v78d6asrlckhwzf/GA%2024B.aif
https://www.dropbox.com/s/16xm59jjgwv6ds2/QR%2024A.aif
https://www.dropbox.com/s/5duud72ih4ftc6u/QR%2024B.aif
  
 There are 4 sound clips from 2 songs, 2 different mixes from each clip, A&B. One of each is a true 24bit 96kHz mix while the other one is a 16bit 48kHz mix but up-sampled to 24bit 96kHz without dithering. Can you guys tell which is which? If most people here can get it right, then it proves that 24bit 96kHz is worth it, if not, then we'll know it's just a waste of space.


----------



## stv014

It could have been 44.1 kHz instead of 48, as the latter is much less common for music. By the way, none of these seem to be real 96 kHz samples, as all of them have a brick wall filter at either ~22 or 24 kHz. Also, there are differences in the audio band that suggest that these might be different masters, rather than just the same "high resolution" sample in its original and 96/24->CD->96/24 converted version ?
  
 For those interested in testing the effects of quantization and band-limiting, I have some older samples here and here. Obviously, the samples used can make quite a difference, and for the lowpass filtering a 96 kHz source file would have been better, but some will probably be surprised how much the sample rate and bit depth can be reduced until the degradation becomes clearly audible.


----------



## Notus

Actually its very difficult to test if there is difference between 24bit / 96khz and 16bit / 44.1khz in practice. In my opinion you would need at least 4 - 10 songs with 3x samples from each song. Recorded up to something around 40khz to make sure there is all of the ultasonic frequency's. Then you remove all of the frequency's above 20khz on one of the sample's. Second file would need to be altered so you remove all of the frequency's above 20khz but introduce random ultasonic noise to the sample. Last would be the unaltered sample. You would have to make sure every hardware in the chain is able to playback the highest ultasonic frequency's in the samples in order to do this test properly. Then you need to get a lot of test subjects with both trained ears and untrained.
 I don't think you are able to test this by simply downloading a file from the internet since you cant be sure if the hardware used was able to capture everything needed for testing.
 Also this would need to be tested with both speakers and headphones.
 Just my opinion. But what do i know, i don't have the deep understanding on the subject that many of you have.


----------



## stv014

notus said:


> Actually its very difficult to test if there is difference between 24bit / 96khz and 16bit / 44.1khz in practice. In my opinion you would need at least 4 - 10 songs


 
  
 Feel free to upload samples you think would be well suited for testing. For practical (file size) and copyright reasons, they should be limited to 30 seconds length, but that is enough if you choose a section that is most likely to have audible differences (if any).
  


> Originally Posted by *Notus* /img/forum/go_quote.gif
> 
> I don't think you are able to test this by simply downloading a file from the internet since you cant be sure if the hardware used was able to capture everything needed for testing.


 
  
 Fortunately, the downloaded files can be analyzed to find out if there is any "useful" (that is, probably still inaudible, but not just noise or other artifacts) ultrasonic content without an early roll-off. Obviously, the transducers used for playback make a difference, as most tend to roll off steeply above 20-30 kHz at most, but then the people who use them should not worry as much about recordings lacking ultrasonics anyway.


----------



## Danz03

What do you mean they are not real 96kHz samples? What made you think I used a brick wall filter at 22 or 24 kHz? You can hear that high? And what do you mean there are differences in the audio band? I personally recorded these tracks and mastered them myself using 32bit floating point and made sure they have no post production afterwards. With one version mastered at 24bit 96kHz, and the other at 16bit 48kHz and directly up sampled to 24bit 96kHz. I used 48kHz instead of 44.1 because since I'm not using dithering, up sampling 48kHz to 96kHz would be much straight forward and would cause less error. And why would someone need 4-10 songs and recorded to 40kHz to test it properly? I thought most people who agreed that 24bit is better can hear a noticeable difference right away.
  
 Quote:


stv014 said:


> It could have been 44.1 kHz instead of 48, as the latter is much less common for music. By the way, none of these seem to be real 96 kHz samples, as all of them have a brick wall filter at either ~22 or 24 kHz. Also, there are differences in the audio band that suggest that these might be different masters, rather than just the same "high resolution" sample in its original and 96/24->CD->96/24 converted version ?
> 
> For those interested in testing the effects of quantization and band-limiting, I have some older samples here and here. Obviously, the samples used can make quite a difference, and for the lowpass filtering a 96 kHz source file would have been better, but some will probably be surprised how much the sample rate and bit depth can be reduced until the degradation becomes clearly audible.


 
  
  


notus said:


> Actually its very difficult to test if there is difference between 24bit / 96khz and 16bit / 44.1khz in practice. In my opinion you would need at least 4 - 10 songs with 3x samples from each song. Recorded up to something around 40khz to make sure there is all of the ultasonic frequency's. Then you remove all of the frequency's above 20khz on one of the sample's. Second file would need to be altered so you remove all of the frequency's above 20khz but introduce random ultasonic noise to the sample. Last would be the unaltered sample. You would have to make sure every hardware in the chain is able to playback the highest ultasonic frequency's in the samples in order to do this test properly. Then you need to get a lot of test subjects with both trained ears and untrained.
> I don't think you are able to test this by simply downloading a file from the internet since you cant be sure if the hardware used was able to capture everything needed for testing.
> Also this would need to be tested with both speakers and headphones.
> Just my opinion. But what do i know, i don't have the deep understanding on the subject that many of you have.


----------



## ab initio

One can use a spectrum analyzer abd other mathematical tools to check the validity of the tracks being tested. In this case, we want to test the difference between sample rates and bit depths, not differences between masters or software filters, etc.

One should start with the high resolution original and compare with downsampled versions of the same track. Special care must be taken so that the downsampling algorithm doesn't introduce aliased signals (e.g., one cannot niavely decimate)

Cheers


----------



## stv014

danz03 said:


> What do you mean they are not real 96kHz samples? What made you think I used a brick wall filter at 22 or 24 kHz? You can hear that high?


 
  
 Well, I am not the one claiming to be able to hear a difference, but these graphs show there are problems with the samples:
  
    
    
 None of the files seem to have any content that looks like audio above 24 kHz, and one of each pair drops off already at the Red Book standard 22.05 kHz. To me, even the higher bandwidth versions look more like low level artifacts between 22 and 24 kHz from bad quality sample rate conversion than real content. The lower bandwidth files include what seems to be noise shaping in the ultrasonic range. I would guess the low bandwidth samples are resampled Red Book (which is indeed not uncommonly sold as "high resolution"), and the others have been converted from that to 48 kHz with not quite perfect resampling. However, the low frequency difference in one of the tracks is odd, at first I thought there is a mastering difference, but I am not sure.
  


danz03 said:


> I personally recorded these tracks and mastered them myself using 32bit floating point and made sure they have no post production afterwards.


 
  
 If you recorded them yourself, maybe the sample rate was converted due to some software problem. Unwanted sample rate conversion by the drivers or the operating system is a common issue on Windows. But why record twice, when you can just convert the higher quality version in software, which avoids unexpected differences like what is shown above ?
  


danz03 said:


> I used 48kHz instead of 44.1 because since I'm not using dithering, up sampling 48kHz to 96kHz would be much straight forward and would cause less error.


 
  
 For a good converter, non-integer ratios should not be a problem. These tests show 200 dB stopband (imaging) rejection achieved with resampling from 44.1 to either 88.2 or 96 kHz,  and the only practical difference is that the latter takes longer to process (but by a factor of less than 2).


----------



## bigshot

Whenever you put up listening tests on the internet, it's inevitable that people are going to cheat and use measurements instead of their ears.


----------



## esldude

bigshot said:


> Whenever you put up listening tests on the internet, it's inevitable that people are going to cheat and use measurements instead of their ears.


 

 I have had that experience certainly. 
  
 I have cheated on myself.  Purchased hirez recordings done with modern digital gear.  Checked to see it has something above 20 khz.  Then digitally filtered out everything below 20 khz.  Play the result and heard complete silence.  My speakers go to 28 khz, and a test mic showed at least some response actually making it into the room.  To my hearing through my ear....nope, nothing.  Even more clearly when I generated high frequency test tones so I can be more certain something steady is there to be heard.  And I hear....nothing.  These wonderful qualities of hirez formats sure are hard to actually find.


----------



## Notus

I did say that i don't have deep understanding on the subject. 


> I thought most people who agreed that 24bit is better can hear a noticeable difference right away.


 
 I am not one of those people.
 They claim that the ultasonic frequency's bring something to the music and so on. You would have to test and prove it in some form of blind test and the test should contain several songs to make sure the testing material has enough variance and it cant be blamed on just being one song. I personally don't think 16bit / 44.1khz sound any different when compared to 24bit / 96khz.
  
 Even if you can analyze a file and it shows something above 20khz can you actually test and tell what that something is? Since you cant hear it........ 
	

	
	
		
		

		
			




 That's why random file from internet does not really work for scientific testing.


----------



## kraken2109

It's important to remember that, like headphones and speakers, microphones are transducers and most do not respond well at all above 20kHz.


----------



## ab initio

kraken2109 said:


> It's important to remember that, like headphones and speakers, microphones are transducers and *most do not respond well at all above 20kHz*.


 
 They aren't designed to because they don't need to. Not for audio applications, anyway.
  
 In the bat flight lab I used to work in, some of the biologists were interested in echolocation. They had microphones and speakers for the ultrasound. Those measure and produce ultrasound because it's relevant to the application.
  
  
 Cheers


----------



## castleofargh

ab initio said:


> kraken2109 said:
> 
> 
> > It's important to remember that, like headphones and speakers, microphones are transducers and *most do not respond well at all above 20kHz*.
> ...


 


 someone is making some nice 24/96 recordings!


----------



## ab initio

castleofargh said:


> someone is making some nice 24/96 recordings!


 

 For bats, even 24/96 doesn't cut it! See (Fenton and Bell, 1981)
  
 Cheers


----------



## bbmiller

Does anybody here think that music video producers give you a better production on 24 bit depth Blu-ray rather than 16-bit DVD not because the music produces cannot give you just as good sonic experience on 16 bits DVD, but they simply don't?


----------



## kraken2109

ab initio said:


> They aren't designed to because they don't need to. Not for audio applications, anyway.
> 
> In the bat flight lab I used to work in, some of the biologists were interested in echolocation. They had microphones and speakers for the ultrasound. Those measure and produce ultrasound because it's relevant to the application.
> 
> ...


 

 That's kind of what I mean, what's the point in recording to a format that can store these frequencies when they won't actually be in the recording.


----------



## bigshot

bbmiller said:


> Does anybody here think that music video producers give you a better production on 24 bit depth Blu-ray rather than 16-bit DVD not because the music produces cannot give you just as good sonic experience on 16 bits DVD, but they simply don't?


 
  
 I think that many times, lower bit rate audio is deliberately hobbled, yes. Every legacy rock SACD I ever owned has different mastering on the redbook layer than on the SACD layer. The inferior sound quality of the redbook was always because of the inferior mastering, not the format itself.


----------



## Notus

It all comes down to profit as always. Its the music industry that is spreading the myth of 24bit / 96khz superiority. Or you could say force feeding it to people. I have noticed that a lot of recordings or masters seem to be worse these day's. In some it is so bad there is even audible distortion. I am not a big fan of conspiracy theory's but Music industry has done some very dodgy decisions in the past and i am not surprised if they are actually using different masters for CD and DVD / SACD layer / Blue-ray releases. Anybody remember the mess with copyright protection on CD's?
 They are trying to sell new products but people wont buy them if the product is identical in quality, only difference is the price.


----------



## RazorJack

Agreed, and as this thread has shown, they sure have done a great job into making people (experts on the matter even, self proclaimed or not) believe there is any audible benefit to anything beyond Redbook. I say keep it simple, enjoy your CD's and compressed online streams, or show me the DBT results that proves otherwise.


----------



## Rooster81

I'm going to play the devil's advocate here (please note this is honest curiosity and I'm trying to further my somewhat limited knowledge).
  
 What about the arguments that one can _"feel"_ a difference in the music at 24 bits?  One example would be greater listening fatigue when comparing 16 to 24.  This type of difference wouldn't be apparent in a quick ABX test, but only after extended listening periods.
  
 Would it be possible that the lesser noise of 24 bits over 16 bits (even though slight) could be _perceived,_ maybe even on a subconscious level, and would lead to a better listening experience?


----------



## bigshot

Listening fatigue is due to response spikes. The lack of super high frequencies won't cause it, but having too much up there will.


----------



## esldude

rooster81 said:


> I'm going to play the devil's advocate here (please note this is honest curiosity and I'm trying to further my somewhat limited knowledge).
> 
> What about the arguments that one can _"feel"_ a difference in the music at 24 bits?  One example would be greater listening fatigue when comparing 16 to 24.  This type of difference wouldn't be apparent in a quick ABX test, but only after extended listening periods.
> 
> Would it be possible that the lesser noise of 24 bits over 16 bits (even though slight) could be _perceived,_ maybe even on a subconscious level, and would lead to a better listening experience?


 

 I don't know if such a test of just this for only 16 vs 24 bits has been done.  It is very similar to an oft heard claim that longer auditioning is more perceptive, that quick switches interfere with more subtle perception.  That has been tested and always quick switches are more perceptive.  Fairly large distortion differences are not perceived when auditioning is over several minutes (or in one case several days to weeks).  Yet the same level of distortion is unambiguously perceived with segments of a a couple dozen seconds and rapid switching. 
  
 Your ear has to perceive an actual difference for it to possibly effect you even over longer periods of time.  If some level of difference, 16 vs 24 bits, is below what human ear drums  can respond to, then how does more exposure over longer periods somehow uncover that?  If the ear cannot perceive something it cannot have an avenue of cumulative effect.


----------



## ferday

rooster81 said:


> Would it be possible that the lesser noise of 24 bits over 16 bits (even though slight) could be _perceived,_ maybe even on a subconscious level, and would lead to a better listening experience?




Where is this lesser noise?

I've never seen evidence showing there is more noise in 16 bit....there is evidence that higher res is actually noisier than Redbook 

If your audio chain isn't designed to deal properly with ultrasonics then your subconscious is far more likely to perceive the distortions created in the chain


----------



## SilverEars

bigshot said:


> Listening fatigue is due to response spikes. The lack of super high frequencies won't cause it, but having too much up there will.


 
 Would fatigue also degrade the quality of the sound perceived also?  I notice this when I listen to fatiguing phones, as over time it sounds more distorted.


----------



## tdog

rooster81 said:


> I'm going to play the devil's advocate here (please note this is honest curiosity and I'm trying to further my somewhat limited knowledge).
> 
> What about the arguments that one can _"feel"_ a difference in the music at 24 bits?  One example would be greater listening fatigue when comparing 16 to 24.  This type of difference wouldn't be apparent in a quick ABX test, but only after extended listening periods.


 
  
 I have no problem with the theoretical possibility that 24-bit audio might "feel" better than 16-bit, i.e. the perception is fairly subconscious. However, of the many listening test that have been conducted (thousands or only hundreds? I've lost count) there would be a result of preference for 24-bit, and _every_ test I have seen or been a part of has shown that people are guessing (as long as the material is properly dithered to 16-bit etc).
  
  


> Would it be possible that the lesser noise of 24 bits over 16 bits (even though slight) could be _perceived,_ maybe even on a subconscious level, and would lead to a better listening experience?


 
  
 It is possible that, given the right program material and a pristine listening environment, quantisation distortion or noise is just audible on undithered 16-bit audio. However it is highly dependent on the program material, and only if it's undithered.
  
 None of this stops me buying and enjoying hi-res music files, and as an audio professional I always record in hi-res for the obvious reasons (headroom on production, easier anti-aliasing filtering etc), but as a _delivery_ format, 16-bit is just fine and completely transparent if it is done properly... (of course, it's not always done properly, but that is hardly the fault of the format).


----------



## bigshot

silverears said:


> Would fatigue also degrade the quality of the sound perceived also?


 
  
 Not if the spike is narrow enough, or is out of the range of human hearing.


----------



## bigshot

tdog said:


> It is possible that, given the right program material and a pristine listening environment, quantisation distortion or noise is just audible on undithered 16-bit audio. However it is highly dependent on the program material, and only if it's undithered.


 
  
 What sort of audio is undithered when it's bounced down? I've never heard of anyone doing that.


----------



## SilverEars

bigshot said:


> Not if the spike is narrow enough, or is out of the range of human hearing.


 
 So you're talkin about sibilance.  Tsss, Sssss.  That thick spike?


----------



## bigshot

Not necessarily. A spike anywhere in the core frequencies can be irritating, but not catastrophic if the band is narrow enough. Sibilance is just one of these sorts of things.


----------



## castleofargh

I cannot stand spikes on the frequencies. a bad crossover with complete silence somwhere, I might not even notice, and when I do, I'm just sad that I'm missing something. that's it, no aggressive sounding from it.
 but 1 spiky bump of more than 5db somewhere and you can be sure I will dispise the headphone and find it fatiguing even if I don't know why beforehand(the 1khz/5khz being my nemesis).
 now I'm even more focused on how they smoothed out the frequency response graphs than on how it is compensated. I've been tricked by golden ears measurements that way they make everything so smooth that you know nothing about how it really is. Tyll is my favorite for that very reason.


----------



## cjl

ferday said:


> Where is this lesser noise?
> 
> I've never seen evidence showing there is more noise in 16 bit....there is evidence that higher res is actually noisier than Redbook
> 
> If your audio chain isn't designed to deal properly with ultrasonics then your subconscious is far more likely to perceive the distortions created in the chain


 
 There's absolutely no question that a properly implemented 24 bit playback chain has a better SNR and lower noise than 16 bit. The ultrasonics you are talking about have nothing to do with bit depth - high sample rate is what allows for ultrasonic content (and, as you said, potential distortion). Now, you could make a very good argument that the SNR and noise levels in a modern 16 bit setup (usually around -100dBFS or better) are good enough that it doesn't really make any difference anyways. Certainly every blind test implemented using reasonable audio samples has indicated that this is the case. However, the basic premise of the original statement (that 24 bit can have lower noise and better SNR) is definitely true.


----------



## bigshot

For the purposes of listening to music in a home audio situation, there is absolutely no difference in noise levels between 16 and 24 bit.


----------



## headdict

bigshot said:


> For the purposes of listening to music in a home audio situation, there is absolutely no difference in noise levels between 16 and 24 bit.



In a noisy on-the-go situation there is even much less difference.


----------



## bigshot

Much less than none at all!


----------



## RazorJack

rooster81 said:


> I'm going to play the devil's advocate here (please note this is honest curiosity and I'm trying to further my somewhat limited knowledge).
> 
> What about the arguments that one can _"feel"_ a difference in the music at 24 bits?  One example would be greater listening fatigue when comparing 16 to 24.  This type of difference wouldn't be apparent in a quick ABX test, but only after extended listening periods.
> 
> Would it be possible that the lesser noise of 24 bits over 16 bits (even though slight) could be _perceived,_ maybe even on a subconscious level, and would lead to a better listening experience?


 
  
 Feeling, or perceiving, how exactly, with which of the five senses?
  
 This has nothing to do with science. 
  
 That said, I think people _feeling _things is what keeps 99,9% of Head-Fi and the audiophile community thriving, and there's nothing wrong with that.


----------



## headdict

razorjack said:


> Feeling, or perceiving, how exactly, with which of the five senses?
> 
> This has nothing to do with science.
> 
> That said, I think people _feeling_ things is what keeps 99,9% of Head-Fi and the audiophile community thriving, and there's nothing wrong with that.




Feeling can be subject to scientific studies. Does anybody know whether extended double blind testing has ever been done to capture potential long-term effects of higher bit rates?
Fully agree on the importance of feelings (and money!) for the audiophile community. Beats science anytime.


----------



## cjl

bigshot said:


> For the purposes of listening to music in a home audio situation, there is absolutely no difference in noise levels between 16 and 24 bit.


 

 Absolutely. That's a very different claim than the claim that 24 bit is noisier than 16 bit though.


----------



## Rooster81

cjl said:


> There's absolutely no question that a properly implemented 24 bit playback chain has a better SNR and lower noise than 16 bit. The ultrasonics you are talking about have nothing to do with bit depth - high sample rate is what allows for ultrasonic content (and, as you said, potential distortion). Now, you could make a very good argument that the SNR and noise levels in a modern 16 bit setup (usually around -100dBFS or better) are good enough that it doesn't really make any difference anyways. Certainly every blind test implemented using reasonable audio samples has indicated that this is the case. However, the basic premise of the original statement (that 24 bit can have lower noise and better SNR) is definitely true.


 

 Thanks for answering the question.


----------



## Rooster81

razorjack said:


> Feeling, or perceiving, how exactly, with which of the five senses?
> 
> This has nothing to do with science.
> 
> That said, I think people _feeling _things is what keeps 99,9% of Head-Fi and the audiophile community thriving, and there's nothing wrong with that.


 
  
 "Feeling" a difference (at least in this context), would be picking up on minute differences on a subconscious level.  These minute differences would accumulate over a listening period and would make the listening experience overall less satisfactory when compared to the alternative.
  
 I *believe* there are documented studies that show small volume differences, at a1 db or maybe less, will be enough to persuade a listener they "feel" A is better than B, even though they are the same recording.  That might be one example.


----------



## bigshot

rooster81 said:


> "Feeling" a difference (at least in this context), would be picking up on minute differences on a subconscious level.  These minute differences would accumulate over a listening period and would make the listening experience overall less satisfactory when compared to the alternative.


 
  
 When it comes to human perception, differences in sound don't "reveal themselves over time". Our auditory memory for comparing two similar sounds is around 3-4 seconds at most. Our minds do *adapt* to sounds though. What sounds muffled initially might sound more normal over an extended period of listening. So the exact opposite to your theory is the truth of the matter. You don't become more perceptive over time, you become less perceptive and more forgiving.
  
 All of the tests of frequencies above the range of human hearing that I've ever seen come to the same conclusion... they add absolutely nothing to sound quality. They can only detract if they create distortion at lower frequencies within the audible range, or cause irritation if the volume is too high. They use high volume super-audible frequency blasts on rioters to induce headaches and nausea.
  
 You're better off with plain old redbook.


----------



## Rooster81

I've another question regarding equipment, again playing devil's advocate.  And again, I'm trying to increase my knowledge.
  
  
 If a DAP has a frequency response of, say 10Hz ~ 20kHz, wouldn't that mean a recording in 192 kHz is meaningless?
  
 Thanks.


----------



## castleofargh

rooster81 said:


> I've another question regarding equipment, again playing devil's advocate.  And again, I'm trying to increase my knowledge.
> 
> 
> If a DAP has a frequency response of, say 10Hz ~ 20kHz, wouldn't that mean a recording in 192 kHz is meaningless?
> ...


 
 high sample rate on the track has no real purpose IMO. but high sample capabilities from the dac can be good( again for filters more than anything).
  if we're talking only frequency response, then I would answer your question by another one, if a human can hear between 10hz ~ 20khz, wouldn't it be meaningless to record higher frequencies? ^_^


----------



## a_recording

It would be very difficult to design a study that relied on people reporting their 'feelings' without specifically asking them leading questions that might lead to confirmation bias. I do know of studies like testing of Tsutomu Oohashi's idea of the 'hypersonic effect' where they did things like monitor brain waves etc. The results have not been replicated in similar experiments though.
  
http://en.wikipedia.org/wiki/Hypersonic_effect
  
 As far as testing of 24 bits goes Tomscy2000 suggested that maybe the ear can selectively tune out the noise floor in an environment (could be possible, the auditory system is pretty complex) and so perhaps would be able to make use of the extra dynamic range 24 bit audio provides.
  
 This still doesn't solve the problem that most content does not even use the 96dB dynamic range of 16 bit audio, and end users have a tendency to turn down the volume rather than have 96dB peaks.


----------



## bigshot

rooster81 said:


> If a DAP has a frequency response of, say 10Hz ~ 20kHz, wouldn't that mean a recording in 192 kHz is meaningless?


 
  
 Yes. And even if the DAP is capable of reproducing super audible frequencies the answer is the same.


----------



## thievesarmy

bigshot said:


> Now that you've had the scales pulled off your eyes when it comes to 24 bit, would you like us in Sound Science to turn you on to just as surprising truths about high bit rate lossy files, headphone amps and external DACs?


 
  
 I would. I'm slowly making my way through this ENTIRE thread. Been very enlightening, to say the least. Seriously though, from here I'd love recommendations for reading up on the 'surprising truths' you mentioned.


----------



## ferday

thievesarmy said:


> I would. I'm slowly making my way through this ENTIRE thread. Been very enlightening, to say the least. Seriously though, from here I'd love recommendations for reading up on the 'surprising truths' you mentioned.


 
 the truths really aren't that surprising...unless you're a salesman for cables or something.
  
 a good place to start reading is here, some nice links to listening tests regarding flac/mp3 and stuff like that   http://www.hydrogenaud.io/forums/index.php?showtopic=82777
  
 there are numerous threads here on headfi, just scroll around sound science and pick a topic of interest.  there's a little bit of good info in most threads.
  
 ethan winer (just google) is an excellent source of info, he has some good videos which are easy to watch and understand and a lot of info on his site.  archimago is one of my favorites, he's kind of an everyman tester, doing testing on normal (as in not the highest end) equipment, using normal equipment.  and it's a fun read   http://archimago.blogspot.ca/


----------



## bigshot

thievesarmy said:


> I would. I'm slowly making my way through this ENTIRE thread. Been very enlightening, to say the least. Seriously though, from here I'd love recommendations for reading up on the 'surprising truths' you mentioned.


 
  
 Ah! That is easy. At this point in digital audio, the solid state electronics part (CD players, DACs, amps) are all audibly transparent if they are operating to spec and not broken. That means that a $40 Coby CD player sounds just as good as a high end one, and as long as an amp has enough power to do the job and matches the impedance of the transducers, you are set to go. Many headphones work fine right out of an iPod and don't require amping at all. All modern DACs should be audibly transparent, whether they are external or built into a DAP or disk player. None of these things are what you should focus your attention on. The most important consideration in electronic components is features and usability. You shouldn't spend a lot of money on this part of your system.
  
 As long as the bitrate is high enough and the codec is recent enough, even lossy files will sound as good as any higher bitrate formats. Don't depend on numbers on a sheet, do your own controlled listening tests and figure out for yourself what works. For me, it's LAME 320 or AAC 256 VBR... and I am VERY picky. Anything beyond that is fine, but not an audible improvement.
  
 Transducers, however, make a BIG difference. They are the wild card of the system. Audition them before you buy them and buy the ones that sound the best to you. In general, good headphones and speakers are quite expensive. Here is where you should put your money.
  
 Once you have functional electronics and high quality headphones or speakers, you can fine tune them with equalization- or in the case of speakers, a combination of room treatment and EQ. Balancing the response can make any system sound better. Get a good digital equalizer and experiment. Learn what different frequency bands sound like.
  
 Once you have set up your system the way you think it should work well, spend time carefully listening to really good recordings. Analyze what you hear and try to identify problems. If you clearly define a problem and understand what is causing it, you are most of the way to solving it. Always refine your settings until you hit perfection. If you have speakers, consider going to 5:1 and implementing DSPs. If you already have a solid two channel system, 5:1 will hit it out of the park. Then sit down for a few decades and enjoy great music.


----------



## thievesarmy

bigshot said:


> Ah! That is easy. At this point in digital audio, the solid state electronics part (CD players, DACs, amps) are all audibly transparent if they are operating to spec and not broken. That means that a $40 Coby CD player sounds just as good as a high end one, and as long as an amp has enough power to do the job and matches the impedance of the transducers, you are set to go. Many headphones work fine right out of an iPod and don't require amping at all. All modern DACs should be audibly transparent, whether they are external or built into a DAP or disk player. None of these things are what you should focus your attention on. The most important consideration in electronic components is features and usability. You shouldn't spend a lot of money on this part of your system.
> 
> As long as the bitrate is high enough and the codec is recent enough, even lossy files will sound as good as any higher bitrate formats. Don't depend on numbers on a sheet, do your own controlled listening tests and figure out for yourself what works. For me, it's LAME 320 or AAC 256 VBR... and I am VERY picky. Anything beyond that is fine, but not an audible improvement.
> 
> ...


 
  
 Heh. Ok, interesting starting point. I have noticed some of the things you're talking about already. For example…
  
 Last week I finally was able to obtain some good quality HD 650's (used, at a great price). I had long been looking forward to hearing some high-end Senn's, as they are very praised on this site. Well, I wasn't disappointed, and I really am loving them, but I was a little surprised by something. First, I tried using them with my computer as the source (retina MBP) to a Dragonfly DAC @ 16, 44.1. The amp was a new one I just got, a S.M.S.L. sAp II that set me back $60. That rig sounded awesome and I was thinking, WOW, I can't wait to hear how it sounds at home on my Bifrost + Vali! This is gonna be AWESOME.
  
 Well, it did sound awesome, but that isn't the important part. After these more "capable" rigs, I decided to try an experiment. I plugged them straight into my iPhone 5, and to my surprise, these 300 Ohm high-end phones sounded pretty great from there, too. I don't know if they sounded QUITE as good as the other rigs, and I didn't listen quite as long, or to the same pieces of music (for a more accurate test), but I was pleasantly surprised at how well they sounded straight out of the iPhone. I almost came here asking about it, but I figured there was a lot I still didn't fully understand about impedance & sensitivity, etc.
  
 I still need to finish this thread, I'm only on page 105…


----------



## Krutsch

bigshot said:


> <snip, snip>
> 
> What sounds muffled initially might sound more normal over an extended period of listening.


 
  
 Yeah... just ask anyone with Senn HD-650s... oh, did I say that out loud?


----------



## SilverEars

krutsch said:


> Yeah... just ask anyone with Senn HD-650s... oh, did I say that out loud?


 
 Yes, if you come from Grados.


----------



## stv014

> Originally Posted by *a_recording* /img/forum/go_quote.gif
> 
> As far as testing of 24 bits goes Tomscy2000 suggested that maybe the ear can selectively tune out the noise floor in an environment (could be possible, the auditory system is pretty complex) and so perhaps would be able to make use of the extra dynamic range 24 bit audio provides.


 
  
 Well, it could do the same to the noise floor of dithered 16-bit PCM audio. After all, a -110 dBFS tone can still be heard in that format "under" the noise, if the volume is turned up.


----------



## Krutsch

I will probably regret posting anything on Sound Science, not being a "Sound Scientist", but I feel compelled to mention why I have been buying 24-bit tracks on HDTracks (I don't work for anyone in the industry).
  
 They cost the same, roughly speaking, as the physical CD for newer releases or remasters.
  
 For example, Lana Del Rey's newest release (deluxe edition) lists at $US 17.99 and the same CD + bonus tracks lists at $US 15.99 on Amazon and, BTW, I can download it now instead of waiting for shipment.  Yes, I know you can buy used CDs for less - I am ignoring that, for new stuff, although I do buy used CDs for older releases.
  
 Anecdotally speaking, as an amateur enthusiast, I have been impressed with the mastering on many of the tracks I've purchased from HDTracks; especially compared with the same music purchased from iTunes (AAC), as mastered for iTunes and/or AAC-encoded.
  
 Comparing lossy AAC with lossless versions may be viewed as an unfair comparison of apples and oranges, but many on this forum don't believe that one can reliably hear the difference - so maybe the comparison isn't unfair, I don't know.
  
 I recently did this comparison with Lorde's Pure Heroine - I purchased the iTunes version and downloaded the HDTracks version (24-bit, 48 kHz).  There's a section at the start of the second track (400 Lux) that sounds like an electronic sweep, if you will, and the differences between the AAC version and the HDTracks version was really apparent.  The lossless version sounds like it's sweeping across from left to right and even with headphones has sort of a 3-D imaging effect.  The AAC version, in my amateur-sighted-evaluation-opionion, was very flat and didn't convey that sense of sound staging or movement of the sound source.  Is that from mastering differences, is that from compression?  I have no idea.
  
 So, for me, my ears and my wallet, whether or not I can hear the difference between 16 and 24-bit isn't totally the point (I probably can't, based on what I've read and my ad-hoc experience of taking 24-bit tracks and then re-sampling and dithering using iZotope 64 in my Mac).
  
 But it's also probably not hurting anything, either, and for the mastering and the convenience, why not?


----------



## ab initio

krutsch said:


> I will probably regret posting anything on Sound Science, not being a "Sound Scientist", but I feel compelled to mention why I have been buying 24-bit tracks on HDTracks (I don't work for anyone in the industry).
> ...
> *There's nothing to regret! If you are curious about the topic, interested in learning, and honest and open minded, you will fit in just fine here! Welcome *
> 
> ...


 
 Cheers


----------



## Notus

What makes me concerned is the fact that they are butchering CD masters or not doing them properly to promote the sells of Higher bit rate albums that have different masters.
 This is wrong, but not that uncommon with any industry, unfortunately.


----------



## Krutsch

notus said:


> What makes me concerned is the fact that they are butchering CD masters or not doing them properly to promote the sells of Higher bit rate albums that have different masters.
> This is wrong, but not that uncommon with any industry, unfortunately.


 
  
 I wonder about that - I guess I was trying to say that in my lengthy post.  Are different masters used, for example, with Lorde's Pure Heroine for CD/HDTracks and iTunes?  The conspiracy theory is that this is intentional to dupe "audiophiles" into paying more for higher-res tracks.
  
 But, as I also pointed out in my lengthy post, the costs for the "bonus tracks" versions are basically the same.  Additionally, this all seems like a lot of work to appeal to a relatively limited audience (i.e. HD Tracks downloads).
  
 Not saying I know... I am saying I am wondering.


----------



## bigshot

notus said:


> What makes me concerned is the fact that they are butchering CD masters or not doing them properly to promote the sells of Higher bit rate albums that have different masters. This is wrong, but not that uncommon with any industry, unfortunately.


 
  
 How many copies of Dark Side of the Moon can you be convinced to buy?


----------



## ferday

bigshot said:


> How many copies of Dark Side of the Moon can you be convinced to buy?




Likely the same amount that you have copies of Star Wars on every format and remaster...


----------



## Krutsch

bigshot said:


> How many copies of Dark Side of the Moon can you be convinced to buy?


 
  
 Exactly... let's see: I have DSoM on:
  
 1) vinyl (from the 70s)  2) Japanese virgin vinyl pressing (from the 80s)  3) early generation CD  4) iTunes remastered  5) SACD
  
 I can repeat much of the above with others, like Led Zeppelin, for example.


----------



## sonitus mirus

You do have to be mindful about the mastering of any music you purchase.
  
 I believe that the following recently remastered Led Zeppelin albums are all audibly transparent between, say, an HD Tracks version and a CD purchase at Amazon.
  
 There is over a 50% savings if you are willing to wait 2 days and rip the CD to your format(s) of choice.  You won't be able to hear a difference.
  
 That said, there is plenty of high resolution music that is remastered and sold as such, and there may not be an equivalent CD available to purchase with the same mastering.


----------



## RazorJack

notus said:


> What makes me concerned is the fact that they are butchering CD masters or not doing them properly to promote the sells of Higher bit rate albums that have different masters.
> This is wrong, but not that uncommon with any industry, unfortunately.


 
  
 I can't say I'm aware of this ever being done for an album that I have bought. And I listen to a lot of stuff.
  
 What I find particularly disturbing is that HD tracks or whatever it's called doesn't even always come with all of the artwork intended for the album. So basically you're paying more but get less in the end: no audible benefit (assuming the CD was created with the same master) and incomplete artwork.


----------



## Notus

Well i admit it is not very common and partly was directed at generally badly mastered albums. I feel the masters have been better on DVD releases compared to the Red book releases. It was a bit too generalising comment. Also about the cost, it vary's a lot where you actually live. Difference can be up to 10€ depending how recent the release is.
 Should have made clear that it bothers me that sometimes the audio professional who is doing the masters are intentionally making the Higher bit rate format sound better compared to Red book.
 But i guess what i was after is, why would you even make 2 masters if not intentionally trying to make the other sound better to promote it?
 Why not just use the same master for both releases?


----------



## thievesarmy

well, I FINALLY made it through the entire thread. *pats self on the back* It took 3 days, but I was genuinely interested in engaged in what is for me a VERY complex subject, that even now I certainly don't fully understand, BUT I do feel more equipped now to make better purchasing decisions wrt "audiophile" gear in the future. There are so many people that deserve thanks for their efforts in this thread, but the two that immediately stand out are @gregorio for starting the whole thing off and @bigshot for picking up the slack now that G has effectively gone AWOL. Not that bigshot is the only one picking up the slack, just one that immediately comes to mind and also cause I'm a fan of the way he has been knocking down the naysayers with a certain witty style & flair. Kudos.
  
 The one person that stands out the most in this thread in the WORST way possible would have to be dbbloke (not tagging intentionally) who thankfully doesn't seem to be active anymore. If anyone wants a TLR, do NOT under any circumstances listen to a word this fool has to say. I mean the guy tried to rationalize his denial of science in this thread by referencing the "debate" around climate change. Seriously… hide your daughters.
  
 Anyways, thanks again to all the great contributions from all the people far smarter than myself. Awesome stuff.


----------



## headdict

You nailed it! I have not made it through the entire thread, but could not live without my daily dose of sound science. It is so edutaining and brain-unwashing. Thanks so much for keeping it alive!


----------



## bigshot

razorjack said:


> I can't say I'm aware of this ever being done for an album that I have bought. And I listen to a lot of stuff.


 
  
 When I was trying to do a fair A/B comparison between redbook and SACD, I couldn't find a single legacy hybrid disk that had the same mastering on both layers. The redbook layer was always more compressed and had a higher noise level. I finally found a classical SACD that had the same mastering on both layers, and they both sounded the same. I believe that hybrid SACDs are deliberately hobbled to hold back the redbook layer.


----------



## MrMateoHead

krutsch said:


> I will probably regret posting anything on Sound Science, not being a "Sound Scientist", but I feel compelled to mention why I have been buying 24-bit tracks on HDTracks (I don't work for anyone in the industry).
> 
> They cost the same, roughly speaking, as the physical CD for newer releases or remasters.
> 
> For example, *Lana Del Rey's newest release (deluxe edition) lists at $US 17.99 and the same CD + bonus tracks lists at $US 15.99 on Amazon and, BTW, I can download it now instead of waiting for shipment*.  Yes, I know you can buy used CDs for less - I am ignoring that, for new stuff, although I do buy used CDs for older releases.


 
 I am tired of waiting a week to get a new release. I picked this CD up at Target for $16. The bonus tracks WEREN'T really that great (knew it!). But this has to be one of my favorite albums of the year so far. Really good stuff.
  
 As far as the mastering quality I think it is a bit thick and rough. The heavy use of Reverb is a bit fatiguing on my headphones, and "messy" in my car (with pretty cheap components on board Lana's pretty voice sounds a little pulled apart). Also, many of the tracks have some clipping going on, which isn't as bad as other releases I've looked at. Unlike other female albums I've got recently, the vocals are actually pretty well balanced - they are usually too damn hot.
  
 On the other hand, the new Roots CD is quite well mastered - not too hot, some dynamic range, and some great basslines. It doesn't clip but still has some bright / harsh samples that sound pretty bad on some speakers I have.
  
 It has gotten to the point where recording / mastering quality is more typically my personal source of fatigue and irritation than my speakers. There is no cure for hot, clipped recordings sold by the millions. With unforgiving phones like the HE-400s, I basically just have to roll with the punches sometimes. On the other hand, I don't feel like paying for "HD" tracks which are mostly just how these CDs should have been mastered in the first place. But I should, on that note, add that I think it makes some sense that stuff mastered for "radio play" and "car audio" should be different than home audio. The typical listening environments these days range from vault-quiet to jackhammers, traffic, and trains. Maybe someday we'll buy "raw" audio that can be "mastered" on the fly for our listening environment!


----------



## Mambosenior

This is my daily treadmill read and I am on page 23. As you can see, I cheated and went to this latest chapter (did Gregorio marry Daisy after he returned from the long recording session?). So damn informative! Thanks.


----------



## bigshot

mrmateohead said:


> It has gotten to the point where recording / mastering quality is more typically my personal source of fatigue and irritation than my speakers. There is no cure for hot, clipped recordings sold by the millions.


 
  
 Expand your musical tastes into classical and jazz. All those problems will go away, because those genres are generally well recorded, mixed and mastered.


----------



## headwhacker

bigshot said:


> Expand your musical tastes into classical and jazz. All those problems will go away, because those genres are generally well recorded, mixed and mastered.



 


The problem is I can't just switch to a different genre just to avoid badly mastered recordings. I won't stop listening to rock songs just because of high chance of getting a bad master. 

I listen to music because I like rock the most.


----------



## headdict

headwhacker said:


> bigshot said:
> 
> 
> > Expand your musical tastes into classical and jazz. All those problems will go away, because those genres are generally well recorded, mixed and mastered.
> ...



Isn't it strange that the people who generate the most revenue seem to be taken the least seriously by the music industry whereas lovers of niche genres are treated with the greatest respect?


----------



## castleofargh

headdict said:


> headwhacker said:
> 
> 
> > bigshot said:
> ...


 

 to me the niche genres are sometime saved because the big shots of the industry don't have time to make bad decisions on those. the general will of those guys is to press cds as if they were only adds for radio. what counts is not that the album is good, it is that it impresses enough on radio for people to go buy it. same as trailers for movies where the trailer is pretty much the only good thing in the movie.
 most sound engineers think badly of what they're told to do, but hey they're paid to do it, what can they do? I guess for those niche genres, sometimes nobody comes telling what to do to the engineer so he decides to do a proper job ^_^.


----------



## Krutsch

headwhacker said:


> bigshot said:
> 
> 
> > Expand your musical tastes into classical and jazz. All those problems will go away, because those genres are generally well recorded, mixed and mastered.
> ...


 
  
 +1
  
 Or worse, restricting your catalog of favorite albums based on their "audiophile attributes", versus your love for the music itself.


----------



## cjl

bigshot said:


> Expand your musical tastes into classical and jazz. All those problems will go away, because those genres are generally well recorded, mixed and mastered.


 
 I do enjoy classical and jazz, but I also enjoy rock, electronic, and several other genres. I agree that classical and jazz are usually better recorded and mastered than the other genres, but I don't listen to music for the mastering, I listen to it for the music. Giving up music I enjoy because it isn't mastered as well as it could be would defeat the entire point of listening in the first place, in my opinion.


----------



## SilverEars

You know how people say, you are listening to you gear instead of music?  I feel like I'm listening to recordings, than music I used to enjoy.  I feel it made my ears snobby to better recordings.


----------



## sonitus mirus

No restrictions, it's been the opposite for me.  I listen to every genre of music that is available on Google Music All Access, and I do have my favorites, but it is the quality of the sound in both the recording quality and the prowess of the artist that I enjoy.  I have discovered many gems across many drastically different styles of music.  
  
 Unless you have heard a song in a good audio quality, it probably would never have become a favorite at any point.  Not sure if I would appreciate AC/DC based on the highly compressed CDs I have now.  It is my memory of their music actually sounding great in the past that makes me a fan of this music.


----------



## castleofargh

with headphones it's easy, you can pick a different one for different genres. warm lazy rolled off stuff for rock (to avoid being tortured by badly recorded cymbals). detailed, neutral to bright headphones for classical etc, depending on your own tastes. and personally when I enjoy a song but can't stand how it's recorded, I listen to it on my laptop's tweeters ^_^. they're so bad that it becomes the ultimate forgiving tool.
  
 but listening to some really bad recorded stuff on my not so bad rigs with what I call neutral EQ, no can do. try listening to radioactive(imagine dragons) on your most resolving system and tell me it's not a torture. but in a noisy car with bad audio system that's a pretty cool song.  "every moment has its music" taken to the next level.


----------



## Krutsch

castleofargh said:


> with headphones it's easy, you can pick a different one for different genres. warm lazy rolled off stuff for rock (to avoid being tortured by badly recorded cymbals). detailed, neutral to bright headphones for classical etc, depending on your own tastes. and personally when I enjoy a song but can't stand how it's recorded, I listen to it on my laptop's tweeters ^_^. they're so bad that it becomes the ultimate forgiving tool.
> 
> but listening to some really bad recorded stuff on my not so bad rigs with what I call neutral EQ, no can do. try listening to radioactive(imagine dragons) on your most resolving system and tell me it's not a torture. but in a noisy car with bad audio system that's a pretty cool song.  "every moment has its music" taken to the next level.


 
  
 You nailed it... that's why I have Senn HD-650 and Grado RS2i and keep them on a stand, right next to my chair, using them almost exactly the way you describe.
  
 I need some better Classical cans - the Grados are too bright for strings (IMO) and the Senns are too veiled (IMO).  What would Goldilocks use to listen to Hilary Hahn?


----------



## bigshot

There's only two kinds of response curves... balanced and wrong. If you buy a different set of headphones for every type of imbalanced response, you'll spend an awful lot of money. An equalizer and a set of balanced headphones will get you where you want to go a LOT more efficiently.


----------



## Mambosenior

Audio equipment getting better and audio recording getting worse. Huh! ("It was the best of times, it was the worst of times,...")


----------



## Krutsch

bigshot said:


> There's only two kinds of response curves... balanced and wrong. If you buy a different set of headphones for every type of imbalanced response, you'll spend an awful lot of money. An equalizer and a set of balanced headphones will get you where you want to go a LOT more efficiently.


 
  
 Technically speaking, you are correct.  But where's the fun in that solution?  Just kidding...
  
 So, which headphones are not the wrong headphones?


----------



## bigshot

krutsch said:


> So, which headphones are not the wrong headphones?


 
  
 The right headphones are one with as balanced a response as possible, and the ability to be adjusted with EQ without distorting at loud volumes.


----------



## esldude

mambosenior said:


> Audio equipment getting better and audio recording getting worse. Huh! ("It was the best of times, it was the worst of times,...")


 

 Right you are about that.  Even worse is the audio equipment gets better, and audiophile nonsense to worry about gets worse.  I think that is because it isn't anchored in reality, but rather in imagination.  When the real issues are nearly solved where else to get the little something extra except through imagination.


----------



## headdict

bigshot said:


> The right headphones are one with as balanced a response as possible, and the ability to be adjusted with EQ without distorting at loud volumes.



In my book the right headphones also have to be at least open, closed, comfortable and portable, all at the same time. BTW, based on your previous posts I take it you are not that much into headphones, are you? And yet, you are dedicating so much of your time to this site. How come? Anyway, by all means keep it up! Can't imagine what this place would look like without your relentness pursuit of *sound* (as opposed to pseudo) science.


----------



## Krutsch

bigshot said:


> The right headphones are one with as balanced a response as possible, and the ability to be adjusted with EQ without distorting at loud volumes.


 
  Very PC answer 
	

	
	
		
		

		
		
	


	



  
 It's like what Deep Purple said: can I have everything louder than everything else?


----------



## esldude

headdict said:


> In my book the right headphones also have to be at least open, closed, comfortable and portable, all at the same time. BTW, based on your previous posts I take it you are not that much into headphones, are you? And yet, you are dedicating so much of your time to this site. How come? Anyway, by all means keep it up! Can't imagine what this place would look like without your relentness pursuit of *sound* (as opposed to pseudo) science.


 

 I see what bigshot is getting at, and yet agree with you on the multiple purposes one may have meaning different equipment for different uses.  It is no different whether in headphones or speakers.  Monitoring in the field calls for accurate small monitors.  You give up some things to keep what is most needed.  Mainly no deep bass because they need to be small.  Car speakers vs home speakers have different needs.  There are no perfect for every purpose transducers.  Meaning all involve trade offs.  Those trades offs that make sense vary with location and use in mind. 
  
 So I have one pretty darn nice set of over the ear phones for use walking about driven by mobile phone or tablet.  Another full coverage open phone for non-portable use.  Another good enough, sturdy, closed, not terribly expensive set of phones for music recording on the go. In a pinch any could be used for any purposes, and only one is the most balanced, neutral and good sounding of the three I have.  I have owned a couple phones better than any I have, but for various reasons did not keep them.  I am a bit more uncompromising on my serious home speakers as I can decide the room and other aspects and adapt closer to an optimum level.  Even those are a compromise however. 
  
 I believe in time with more research and expertise thrown at the issue, headphones will be able to provide the most accurate of possible reproductions.  My issue with them over speakers is one of comfort.  I haven't had phones that were put them on and feel comfortable for hours and hours good.  Some were close enough that I could use them a couple hours when needed.  The other thing is really low powerful bass.  One can fix that with a subwoofer and phones for non-mobile use.


----------



## headdict

esldude said:


> I see what bigshot is getting at, and yet agree with you on the multiple purposes one may have meaning different equipment for different uses.  It is no different whether in headphones or speakers.  Monitoring in the field calls for accurate small monitors.  You give up some things to keep what is most needed.  Mainly no deep bass because they need to be small.  Car speakers vs home speakers have different needs.  There are no perfect for every purpose transducers.  Meaning all involve trade offs.  Those trades offs that make sense vary with location and use in mind.
> 
> So I have one pretty darn nice set of over the ear phones for use walking about driven by mobile phone or tablet.  Another full coverage open phone for non-portable use.  Another good enough, sturdy, closed, not terribly expensive set of phones for music recording on the go. In a pinch any could be used for any purposes, and only one is the most balanced, neutral and good sounding of the three I have.  I have owned a couple phones better than any I have, but for various reasons did not keep them.  I am a bit more uncompromising on my serious home speakers as I can decide the room and other aspects and adapt closer to an optimum level.  Even those are a compromise however.
> 
> I believe in time with more research and expertise thrown at the issue, headphones will be able to provide the most accurate of possible reproductions.  My issue with them over speakers is one of comfort.  I haven't had phones that were put them on and feel comfortable for hours and hours good.  Some were close enough that I could use them a couple hours when needed.  The other thing is really low powerful bass.  One can fix that with a subwoofer and phones for non-mobile use.



Exactly what I meant, but was too lazy to elaborate. English being my second language might also work as an excuse.


----------



## bigshot

esldude said:


> I see what bigshot is getting at, and yet agree with you on the multiple purposes one may have meaning different equipment for different uses.  It is no different whether in headphones or speakers.


 
  
 You're talking about different functionality. He was talking about different frequency responses for different genres of music.


----------



## headdict

bigshot said:


> You're talking about different functionality. He was talking about different frequency responses for different genres of music.



Me? No, I wasn't talking about frequency responses at all. Different FR for different genres? What a strange concept!


----------



## castleofargh

I was, but not only FR. in my case my headphones don't all respond to EQ as well as I would like. and for example my hd650 has some funky low freq distortions that can nicely mask some slight clipping, pretty convenient for rap. a neutral clean headphone wouldn't do the trick.
 but overall I would mostly be happy with one real neutral headphone that responds well to EQ. I didn't go that way because what I found almost neutral usually was heavy or not comfy ^_^. it has nothing to do with sound.
  
 and about playing with FR for different genres, I was thinking about some poorly recorded rock or punk with some harsh frequencies. EQing out those frequencies is one way to make it bearable. but well it's a special case, we're talking about ways to listen to bad records. it doesn't represent a lot of what I really enjoy.


----------



## MrMateoHead

bigshot said:


> Expand your musical tastes into classical and jazz. All those problems will go away, because those genres are generally well recorded, mixed and mastered.


 

 I have some of that too 
	

	
	
		
		

		
			




  
 If it wasn't for classical and jazz, I wouldn't have as much basis to "bash" my beautiful-horrible pop recordings.


----------



## MrMateoHead

Also I'd like to know how many albums coming out are actually mastered by Professionals using high-buck equipment.
  
 I've heard some claim that a lot of CDs sound bad because bands basically don't spend big bucks getting top-notch production done. Or, even cheaper, they do it themselves on a Macbook.
  
 Anyone know anything about this?
  
 I know from some experience publishing that color, long-lasting paper and bindings add up. Its a better product, but sure as heck not needed for most "throw-away" texts that we get. Anyway.


----------



## castleofargh

the few bands I've talked to about that(nobody worldwide famous) were all pretty much saying something like this: "if you want to have our best performance come see us on tour". most where whining that it was stupid to force them into making a record as soon as they have the songs, often they actually make the songs while recording, so of course they think they will do better later, and don't put too much love into the albums.
  
 also I guess most are under contract with a record company and might not have the last word when it comes to the mastering.


----------



## esldude

Having made some recordings with a laptop, couple okay condenser mics, and a usb/mic pre, that isn't the bottleneck.  They sound pretty darn good, especially if you don't mess with them.  You can mess with them with a little DSP, and some things are better though some worse.  Either way it is better than probably half of the pop recordings.  I am not doing much because I don't know how.  I don't have equipment equal to the pros.  And much of it has pretty nice sound quality.  I also don't imagine I do a better job than the pros.  My guess is pro guys long in the biz can give you good to great sound.  They can give you what you or maybe the record company manager says to give.  So I don't think in most cases the pro recording guys are to blame.  Either company execs or band members are.  Remember band members don't know what they sound like.  They are all in the middle of playing their own part.


----------



## bbmiller

I am wondering if any of you all think bootlegged recordings can be a very high quality? I think I remember reading that some bands allow fans to plug into this soundboard if I remember that correctly and I am thinking soundboard must mean the sound system for the concert. So do you think you can fine band fan bootleged recordings which are the equal of the best?


----------



## leogodoy

Live recording, bootleg or official, is probably one hell of a tough job. I'm trying to remember which one of my Bob Dylan's boots is a great example of it being possible to achieve good quality...


----------



## bigshot

Most good sounding bootlegs are recordings of live radio broadcasts that weren't meant for release other than as a one time broadcast.


----------



## leogodoy

Ha, those are well above the quality in some punk rock boots I used to collect, those were recorded with a cheap mic positioned in front of a PA.


----------



## cregster

Read your 16 vs 24 article.  I think you misunderstand sound and "real life."
 The whole 16 vs 24 debate is similar to "is 24 bit color better than 16 bit?"  in photos.  It is fairly obvious that 24 bit is better.  It is not as easy to manipulate someone into believing they can't see something as it is to convince them they can't hear something.
  
 Here is real life.  Take two colors of blue, very similar in shade, but not the same.  There are an INFINITE number of gradations in the transition of color from the first blue to the second.  And every one of those transition colors exists.  
  
 It is the same in sound. 24 bit is better, not because it can get something at the two ends of the spectrum, but because it can better get what is in between.  24 bit can capture far more of those real life gradations of the many tones that make up even one instrument sound.  Therefore, 24 bit is more true to actual life in the same way a 24 bit color photo is much more true to life than a 16 bit photo, even though you can see the image very well with 16 bit.  
  
 There are also many emotional nuances put into the music by the players. These can also be more fully described because there are also an infinite number of gradations in pressure, pluck, etc. from one to another.
  
 So, measuring the frequencies and the dynamic range etc.  completely misses the point.
  
 ON the practical side, I have both the 16 bit remastered Beatles CD's and the 24 bit "Apple" usb version.  When I play tunes randomly,  while working, etc.,  and it is a mix of Beatles stuff, other artists,  I can always tell when a 24 bit Beatles tune comes up.  When I occasionally check it on the device, sure enough, it is the 24 bit version.  Every time.
  
 And, even without comparison, I know listening to a 24 bit 192k orchestral recording,  it is not even close to the same 16 bit 44k.  It is obvious.  Not subtle.
  
 Just this very odd thing that gets pushed in sound--24 bit is better than 16 bit in every use of bits (machine running, CPU, photography, robotics, cars sensing the road and on and on)  except just this ONE area,  sound. Very strange.


----------



## bigshot

cregster said:


> 24 bit is better, not because it can get something at the two ends of the spectrum, but because it can better get what is in between.  24 bit can capture far more of those real life gradations of the many tones that make up even one instrument sound.


 
  
 You might want to read up a bit on how digital audio works. There is absolutely no difference between 16 bit audio and 24 bit audio in the audible spectrum. They are identical. No more resolution than in one than the other. No more "gradations". The only difference between redbook and high bitrate / high sampling rate is OUTSIDE the range of human hearing... specifically, sound that is too quiet for you to hear at normal listening volumes and frequencies that are too high for human ears to hear.
  
 The way you think things work isn't necessarily the way they actually work. The differences you hear are likely mastering differences, not improvement in audio quality due to the format of the file.


----------



## Lespectraal

cregster said:


> Read your 16 vs 24 article.  I think you misunderstand sound and "real life."
> The whole 16 vs 24 debate is similar to "is 24 bit color better than 16 bit?"  in photos.  It is fairly obvious that 24 bit is better.  It is not as easy to manipulate someone into believing they can't see something as it is to convince them they can't hear something.
> 
> Here is real life.  Take two colors of blue, very similar in shade, but not the same.  There are an INFINITE number of gradations in the transition of color from the first blue to the second.  And every one of those transition colors exists.
> ...


 
 The ears work with a mechanism different to that of an eye. The way the ears handle information is vastly different so it is not as simple as increasing resolution. It can only go up to a point and beyond that it will not be beneficial and may even produce more distortion. The real question here is whether you can perceive a difference in audio quality. The ears can only ever hear such a small range of frequencies, add to that it diminishes through age.

 To make this simple, take a square. You only ever need to know that a square contains four points which make up the corners. To draw a square you can just do so by first drawing four points that will become the corners, and draw a straight line from point to point to create that square. Increasing the number of points is useless, because you will still end up with a perfect square even if you allocate a trillion points between the corner points. Unless of course if you are talking about say a circle. But in this case, the square can easily be represented through those four points and more points are unnecessary. The same can be said for audio. You can only ever have so much to represent the sound. More than that and you will not gain any difference.

 Also mind you that we are humans by nature, we're not robots with objectivity in mind. Apart from our sensory organs, we do have this "DSP chip", our brains, that processes all information we perceive. The "algorithms" that our "DSP chip" are gained through our own individual experiences, therefore how one person hears will differ substantially from person to person. In this case, the increase in the value of a number(Bit and sample rate) may initiate a response in our "DSP chip" to process that sound heard to be "better" than one that is of a lower value. This is nothing more than psychoacoustics at work here.

 Again, we are emotional, psychological beings that work with all of our senses active at a given time, and all these senses will input information that gets sent to the brain. So sound itself will not be objectively scrutinized by our judgement, everything else like the genre of the song, the location of where you are listening to the song and even your mood at the time you are listening may generate differences that does not originate from the source itself.

 Just my two cents.


----------



## stv014

lespectraal said:


> The ears work with a mechanism different to that of an eye. The way the ears handle information is vastly different so it is not as simple as increasing resolution.


 
   
Actually, vision also has a finite resolution (which is why microscopes are useful, for example), and the ability to perceive different shades of colors is also limited. Commonly used monitor resolutions like 1920x1080 are just not enough to reach the limit yet at a typical viewing distance/FOV, but 8 bits per channel is about right when no further processing is needed.

  
 The discrete "steps" of 16-bit quantized audio can be turned into uncorrelated noise (hiss) with dithering. As long as this noise is not audible, the limited "resolution" of the samples is not a problem.


----------



## bigshot

There are thresholds for everything... there is a frame rate for films that exceeds the "flicker threshold". There's resolution for video that exceeds the ability to see from a normal viewing distance. There's a resolution threshold for images that we can't see beyond without using magnifying glasses. And there is a threshold for recorded music. Redbook exceeds it by a little bit. Everything beyond that is overkill.


----------



## castleofargh

cregster said:


> Read your 16 vs 24 article.  I think you misunderstand sound and "real life."
> The whole 16 vs 24 debate is similar to "is 24 bit color better than 16 bit?"  in photos.  It is fairly obvious that 24 bit is better.  It is not as easy to manipulate someone into believing they can't see something as it is to convince them they can't hear something.
> 
> Here is real life.  Take two colors of blue, very similar in shade, but not the same.  There are an INFINITE number of gradations in the transition of color from the first blue to the second.  And every one of those transition colors exists.
> ...


 

 except that your example is wrong. because both happen to use bits, is only telling us about the fact that it's digital, what they are used for is completely different. increased bit depth in photo brings more colors, increased bit depth in sound bring sounds below the already super low -96db. the most simple color profile is RGB, each color needs to be registered separately 24bit is actually 8bits for each channel. so with sound having 16bit, doesn't that counter your statement? we already have more steps than colors.
  
 if you had to make a parallel between photo and sound, then it would obviously be brightness, not colors. and only the contrast ratio has actually any resemblance with dynamic range. so your point is just plain wrong/false/irrelevant(pick the one you like). apples and oranges.
  if you need to know, TVs specs for brightness are crap compared to what audio systems specs are for sound, the contrast ratios are crap(and usually the specs are lies so it's even worst). printers contrasts can be ok, but they're mostly crap too.
 you talk about photo, but photo would be the studio recording of pictures, not the guy looking at the result in his house on a picture, a photographer can have pro needs and 24bit is cool for post processing, just like it is in audio studios. but we don't output our pics in 24bit for the public, same as music. we give nice 16bit and most of the times depending on the use we give light lossy format. exactly the same as music.
 that's the problem with phony arguments, they can usually be turned around against you.
  
 the variation in bit depth only tells about a change of voltage or air pressure depending where you look at it. all the sound of an album can fit on 1 axis(and it does for 1 ear), we can use a second axis for time, but there are ways to go around that and that's more about sample rates than bit, so let's ignore that for a time^_^.
  
 recorded sound comes down to how many discrete values we need on that one axis to express all the sound we can discern. so let's see what that is for real, and not just according to wishful people:
 -people don't seem to be able to really notice less than 0.1db variations, and they also don't seem to hear something that output less than 1db. (so what would be the point of having 200000values between 0.02db and 0.03db if we hear the same thing with all those 200000? we're humans, our own specs aren't that great.
 -110db being harmful to us after only a few minutes, this is obviously the maximum we should ever use.
 -a calm ambient room has at least 20db of noise that our brain discards voluntarily as it discards so many information all the time to help us behave better than mad people on too much cocaine. so we seem to have a use for 110-20=90db with the best situation possible (listening at 85db instead of 110 would already reduce our needs and usable dynamic range). but let's say we need 90db of dynamic with at least 0.1db increment to listen to music.
 I'm being generous here because as S.E mentioned somewhere, when listening to music, 60DB seems to be the best dynamic we can actually identify. still let's go for 90db of needed dynamic.
 so having 90/0.1=900 discrete values could seemingly cover our daily needs for sound. and 10bit could cover that with 1024 values(K7 tapes did just that or even less, in case you're thinking I'm making stuff up and my numbers are too low to be true).
 the problem with 10bits is noise, we would have quantization noise around 10*6=60db below zero. something that would be very audible on calm passages.
  
 so with 16bit we push that noise down to -96db and in the process we gain 2^16=65536 completely unique values. more than 50 times what we would seem to actually need. and you're here stating as if it was obvious, that we should go for more.
 you can always go for the "more is better" for no actual reason, just like people can buy a 10watt amp to power IEMs. but at some point you need to stop. what is a good number? 100times what we need? 1000times? usually I'm ok with twice as much or 3 for the price of 2, but you might be onto something here. "buy 1 get 16000free!" it sure would sell just fine.
 sorry for being so short sighted and settle for the cheap 50times more than needed redbook and its ugly inaudible noise.
  
  
 anyway back to the actually working analogy of brightness and sound, they have a lot in common:
 we can hear a super quiet sound if there are no other sound at the same time or just before, and we can also hear an explosion at 120db. so 24bit covers really our min and max possible hearing capabilities with 144db of dynamic. and that's why people like you argue in favor of 24bit.
 just like our eyes can see a super feint light from a star at night and the next day see the sun. great dynamic here, we have an effective range usually accepted as 24f-stops(for people unacquainted just imagine 1f-stops as 1bits, that works pretty well for digital media).
 so both have a range clearly superior to what gears are offering us and in fact a dynamic pretty similar. so we can complain like kids who heard only half of the story, or try to understand why our reasoning is false. and why the people who created the stuff we use and knew a lot more than us about it, decided that it was enough.
  
 when the sun is high in the sky, you don't see the stars, just like when you're hearing a sound at 90db, you're not hearing the sounds at 5db or 10db. the stars don't go away at dawn and taking a picture at lunch that doesn't show the stars won't compel you to cry out loud about all the image you're missing and how we should get better cameras.
 so why are you guys doing just that with 24bit audio?
 you do not see the stars in daylight, just like you do not hear the quietest decays of an instrument when another is playing loud at the same time. you can pretend you do, but you do not. in both situations they are present, but the limit is human, not material.
  just like the sun in your face will prevent you from seeing clearly the people under the trees, the loud part of music prevents you from hearing the quietest part of a track. asking for what you could never hear even if the band was in front of you, does seem like the strangest obsession. and it would be laughed at in all areas like


> (machine running, CPU, photography, robotics, cars sensing the road and on and on)  except just this ONE area,  sound. Very strange.


 
 very strange indeed. only in that one area, will people come to believe that they understand a technology and try to convince us, when they really just know how to press play.
  
 you say you can recognize the 24bit tracks of the Beatles, it may well be. it could also be luck, maybe it's because your player doesn't deal with 16 and 24 bit the same way and that leads to a different sound(some IMD because of the higher sample rate?), or the masters are different(did you check for differences in audacity?), or maybe the 24bit file has a little more delay before starting on your dap because it takes 0.08s more to buffer the 24bit one, and your clever brain associated that delay with high quality sound... it could be a lot of things, but it is not because there is more sound or more "precision" on the 24bit track. and that's a statement, not an opinion.
 and having experienced something with one system isn't enough to tell that it's the same for all audio systems. in fact it's not even enough to say if it's about 16 vs 24bit, could be the sample rate or the system itself being picky with something. you're just turning assumptions that what you hear are the 8more bits, into a false conclusion.
  
 and I'm writting all this useless stuff nobody will read when I'm not even against 24bit music(I'm talking bit, not sample rate!!!!!!). that's how much I dislike it when people use false argumentation to mislead others.
 16bit is not enough because sound can't get any better, it's enough because it's already more than what us puny humans can hear. just like we don't go around asking for pictures at more than 300dpi, because 300dpi for something we're holding at reading distance is already more than the resolving power of our eyes. more would bring nothing to us for pictures.
 more than the best we can hear will still just amount to the best we can hear. so no! more is not better for us here, it just makes bigger files.


----------



## ab initio




----------



## bigshot

He won't be coming back.


----------



## Dark_wizzie

I love this thread.


----------



## MicroEuphoneum

We are all hear, because we are not all their.


----------



## headwhacker

I hope he comes back. lol


----------



## xdog

After reading some other forum, where similar discussion was going on,
 with some aggresivness going on both sides, I've decided to make a story...
 Most of this stuff was talked many times before, but whatever...
 Little story of KrzysiekK
 KrzysiekK was an extreme audiophile, paying lots of money into speakers and cables, always checking if something could be improved in his audiochain.
 At some time he found out about new 24bit formats, and thought to give it a try.
 But his more technically advanced friends told his that this is wast of money and space, giving him some technical reasons.
 The first thing which KrzysiekK encounter was the 8bit vs 16bit test of song of his famous artist Psy 'Gangam style',
 he obviously saw that 8bit is no good in capturing complexities of Psy voice (and scored 10/10 in the guessing game!).
 He looked at this recording and found out to be a very compressed stuff, having all the sounds need to the maximum output value.
 So he concluded (with the ease of guessing) that he might need something like 40-48db of dynamic range from the
 average sound level value (and in this case the average roughtly equates to maximum and minimum value due to compresion).
 He gave that matter some thought and gathered, remembering some scientific article, that if THD is audible at 1%, and they claim that 0.3% can also be audible
 than it would be better to have 40-50db of dynamic range below the lowest sound (at least from the lowest sound present continously for some part of the music piece - like brushing of hihats)
 He also remebered some article which claimed that even changes of magnitude -50db from the main sine can be distinguishable of the first harmonic.
 But all that was OK in the Psy song, most of the noises where -6db from top, so he still had 90db of dynamic range for the harmonics and stuff
 So now he went to his favorite music (not counting Psy of course), beeing it classical music.
 (And he did the check without nowing what dither is).
 He went to the opera, sat in the first row, and asked the orchestra to play as loud as they can.
 He mesured (as he now uses scientific method as his friend) 120db, he also asked that they made some quite sound,
 and someone brushed hihats) which resulted in 40db on the SPL-meter.
 He now went home and started doing the math:
 120db at maximum, that means that with the minimum sound will be 24 db, and the next one will be 30db (6db jump!).
 But the quite sound has 40db, and he needs like 40-48 db of dynamic range to grasp the harmonics (at few kHz human sensitivitly goes even to minus few db)
 And then the first harmonic will only have values of 0db,24db or 30 db which does not seem to be right
 (he will be able to change the volumn of the main sine from 40 to 41db, but not of the first harmonic).
 So he went to his friends with his concerns, and they told him that this is no problem, because of dithering.
 So he asked his fellow musicians to use that magical (statistical) trick and bring him back the recordings.
 He sat put on his recording, and while the sound of brushing of hihats was not 100% perfect in the previous recording,
 he now got annyoing 24db noise coming out of his speakers (and he has a very quite room, and very isolating heaphones as it is a must for audio purist)
 So he was disenchanted with the whole scientific stuff, thought that the 16bit is good for Pop music (and they even compresing the dynamic range in those)
 and started buing DSD records.
 (btw. when Psy is going to release his all time hit using DSD, or at least 96kHz/24bit as Lady Gaga)


----------



## bigshot

I'm sure you understand what you're trying to say. That makes one of us.


----------



## castleofargh

xdog said:


> After reading some other forum, where similar discussion was going on,
> with some aggresivness going on both sides, I've decided to make a story...
> Most of this stuff was talked many times before, but whatever...
> Little story of KrzysiekK
> ...


 
  
 you should tell at the beginning that it isn't a fiction story, but a science fiction story.(after all it's fitting for sound science ^_^)
 I mean crazyzic(I believe it is the correct spelling on earth, but is it crazysick instead?) is listening at the end to a 16bit record with all 96db of dynamic range used by music. where does that happen? I own hundreds of albums and couldn't find anything above maybe 75db of used dynamic(usually some room noise recorded on the album). so I'm guessing ... parallel universe?
 he's complaining about 24db randomized noise while having the loudest part of the music reaching 120db. that brings a new question, what's the name of crazyZ's species? because obviously it's not human.
 he has a listening room that has lower noise than your average recording studio(25/30db), so I'm guessing he's listening in a very quiet spaceship while in orbit around some planet.
 the 24db of quantization noise being what is annoying him, brings the question of the speakers. what unknown technology does he use? because with even the very best loudspeakers, at 120db, the level of distortion would be so high that 24db of randomized noise would be the last of his problems. let's dream and pretend that his speakers have 0.1% distortion at 120db(lol this is definitely  an optimistic SF number). then the resulting sounds of harmonics and whatever, would be up to 60db ^_^. so obviously those guys in your story invented a new technology with speakers that have 0.00001% distortion, and that's why the quantization noise is so annoying when listening on the spaceship.


----------



## Notus

Actually i think the noise he is hearing is Tinnitus (ringing of the ears) after listening at 120db. Most likely your extreme audiophile friend is deaf by now.
 Even 85db can cause damage to your hearing if exposed to it for a long time and every +3db up from that halfs the time you can be exposed to it.
  
 Nice science fiction story indeed.


----------



## Dark_wizzie

bigshot said:


> I'm sure you understand what you're trying to say. That makes one of us.


 
 I'm going to steal that line one day and I'm not even going to credit you.


----------



## bigshot

Actually, I think that is an Oliver Hardy line and I didn't credit him either!


----------



## xdog

I thinik you have the wrong midnset/assumptions.
  
 I have similar problem with the need for amplification.
 Like for instance Sennheiser HD600 reaches 94db (according to personalaudio.ru) at ~0.25V, so technically most of the smartphones can provide sufficient power
 for that headphones (or even K701) so that the user will get that 85db of average volumn for modern music
 (I'm totally happy with the volumn which I get with my smartphone on HD600 with music such as pop, alternative rock, trance...);
 then you should be even more happy with the 1V standard (cause more can cause hearing damage).
 However some users heavilly insist that HD600 needs an amplifiler, and this is completely understandable if you consider than they are listening
 for piano/violin duoes, which can be recorded like -20db to  -30db from the top; which might put the requirements for the voltage like 10x compared to my (compressed) music;
 which translates into need for a more or less standard 6V output amplifier.
  
 Other examples are
 - I remeber that one person was bragging that HD800 can reach 120db with low distortion.
 - please note that headphones can provide large isolation (-20db in wide frequency range I would guess are the best results, man like with takstars hd6000, it is difficult to hear what is saying a person next to you)
 - technically in my room I would guess that I have noise on the level like 30db (I think that this is the noise of my quite computer fan, beeing the most noisy think in the room);
 however with my Beats mixr headphones (something like 120db@1V) I can hear (playing 24bit silence @ 1V) the noise on the headphone output STX (I think which is like -100db?), and that something is going on with X-Fi HD (I think the noise is something like 110db); and I had the same feeling about the STX heaphone output with some less efficient headphone; so I would guess that 10db noise is hearable in silent passages [I don't care about it myself, because most of my music if 'full of content'] [and I mean those noise values are very quite, the sound of my breathing seemd quite loud]
 - actually one of my friends has a room with that 3d funny isolating material, so I would guess that he goes below 30db of environment noise
  
 Nobody is saying that they will listen continously at 120db, but [I'm guessing here, probably someone more accustomed to orchestral concerts could provide better values]
 - 120db is the maximum theoretical output (see note below)
 - 100db to 110db is the most likely maxiumum short period value obtainable [but it would be better to have some little space just in case]
 - 90db is the normal orchestral level
 - 70db is the level of small duoes or string quartets, playing soft songs
 - 50db is the level of intro sequences
 - 20-30db is the level of hearable echos, reverberation (that person which I mentioned in the story said that it can hear diffrences just with such kind of hidden content, and not with normal level of music]
  
 You can play a little bit with the numbers, but the thing is that the 96db (which is most likely should be restated as 48db, see the previous post) dynamic range might be not enough.
 The 16bit is ok for almost all contemporary music, but it might be not sufficient for classical (especially if someone would like to record everything with the same bit to real db mapping)
  
 One another analogy [very exagerated ] (please note that this sample had to be logarithmitized, but our hearing also does exponantial->linear conversion):
 Here we sell a car which can do 0-200km/h,
 the only problem is that the speed has discrete values.
 That is not a real problem, cause you have 200km, 199.99km, 199.97km/h
 Oh, wait, but at the low level the first atainable level is 50km/h, then 70km/h, so that might be a problem.
 But, not really, cause we have this super mechanism in which we randomly push brakes at high frequencies,
 so in reality you can get 20,30,36,42 km/h.
 But you are still will drive 150-200km?, well even if you don't want to do it other people will force you to just 180-200km/h
  
 I'm going to restate that, the question if 16bit is sufficient is the question if we with 96db of dynamic range can cover all the spectrum which for example classical music can provide, starting from full output of the orchestra to little brushes of instruments and echoes (those things don't have to occur simultanously) with sufficient quantization levels (technically you could argue that you can record violin playing at constant level of 80db with only 1bit  ); or more likely the dynamic range should handle from highest sound to the lowest hearable harmonic of the most quite instruments/echo (which most likely will turn into the 0db in which humans start to percive sound), and from my calculation (based on the work of KrzysiekK) it seems that 16 bit might not be enought. Technically the cost of 24bit is very low, just 1.5x more space is needed, and you have equipment which can reach, even if not 24bit, than 20bit easily.
  
 PS: Personally I'm really happy with the 16bit


----------



## bigshot

Peaks aren't the issue, the noise floor is. Dynamics in digital audio extends downward, not upward. The peaks are the same level no matter whether you use an MP3 or DSD. The difference in resolution is down in the super quiet stuff.
  
 If your living room is very quiet, it might have a 30dB noise floor. In order to hear the quietest sounds in a 90dB recording, you would have to raise them above the level of your room tone. That means that the 90dB dynamic range is actually 120dB in practice. Your 120dB peaks in orchestral music recordings have a noise floor of the concert hall in the 30dB range too. All 24 bit would add to the sound quality would be a beautifully defined bed of noise that if you raised to an audible level, wouldn't sound all that different from redbook quantization noise.
  
 The fact is, super wide dynamics are unpleasant to listen to. A 45-50dB dynamic range is overkill for comfortably listening to even the most dynamic music at a healthy volume. Most music, even classical music is mixed to keep the dynamics in the range of comfort and does't come close to taxing even the abilities of redbook.


----------



## bigshot

By the way, just because the threshold of pain for hearing is up around 120dB, it doesn't mean that humans can hear down to 0dB at the same time. Human hearing compensates for the average sound level and focuses on that. If there are loud horn blasts alternating with the musician turning the page of his music, you aren't going to hear the page turns the way you would if you had been sitting in a silent room for a few minutes and then heard a page turn.


----------



## stv014

> Originally Posted by *xdog* /img/forum/go_quote.gif
> 
> however with my Beats mixr headphones (something like 120db@1V) I can hear (playing 24bit silence @ 1V) the noise on the headphone output STX (I think which is like -100db?)


 
  
 It is about 20 uV (A-weighted) noise voltage at 44100 Hz sample rate with no load. Using a multiple of 48000 Hz instead improves that by 6-7 dB. So, if your headphones really have 120 dB/V sensitivity, the noise SPL at the most commonly used sample rate would be above 20 dBA.


----------



## Steve Eddy

It bears repeating, the effective dynamic range of human hearing is about 60-70 dB.

se


----------



## xdog

Just one more calculus on the quantization problem:
  
 Info: This process will be done assuming that there is no dithering, where you would trade of something like 1-2bits for some small noise.
  
 So lets assume that we have simple 'music' consiting of 2 sine waves.
 One of them has value of 1 (the smallest posible, the next value is 2 which is +6db more) and is the least hearable/persivable sine wave.
 The second sine has some arbitrary value where quantization should not be a problem, for instance 1024 (the next value is ~0.1% larger).
  
 Now we have to take some physiologic data:
 - the smallest memorable diffrence is sound pressure level is 1db
 - with instant change lets assume that it is ~0.3db (we want to be more audiophile friendly here, you can decrease that value if you feel that your better than that)
 This is cerca (log problem) 1/16th change of 6db which is 2^4.
 So if I would want to change the second sine wave by 0.3db (percivable change); I would have to change the first sine wave also by 0.3db;
 but here I would need at least 4 bits of quantization used for making the volumn transition smooth.
 Otherwise it could happen for example that the second sine went from 50db to 50.3db, but the first one went from 12db to 18db; which would affect sound.
 [those considerations were done without the assumption what the real number of bits per second is]
  
 I still am aware that for, lets say Lady Gaga, where everything is like -10db from the top, and the music is 'full-on'
 for variety of reasons (masking, THD of headhones/speakers, sound leakage) everything which is like -60db is not going to be heard.
 But -60db means still additional -36db till we reach the bottom, which is 2^5, which is more quantization steps than I've calculated.
  
 Thank you, I'll not bother you further, just I love music and math
  
 RE: to those issues above, I understand that, but I said you can use heavilly isolating headphones or have even isolated the whole room (you know, where any kind of noise feels welcome); and the 30db is still some average (meaning here some random more pronounced noises, or defined noises), so might be that 10db 2kHz tone would still reach you. As I have given the example with beats, the noise produce by dac can be lower than the sound of my computer fan, or my breathing, but I'm still able to perceive it due to diffrent place of orgin, and diffrent characteristic; the 30db floor gives too much space for 'buts' to rebuke the 24 bit audio myth


----------



## bigshot

The threshold of perception for volume is between .5 and 1 dB. For music it is higher than with test tones.
 30dB is as quiet as a library.
  
 You aren't thinking in the range of reality. In the real world everything is much less sensitive than your abstract concepts.


----------



## castleofargh

@xdog
  
  the headphone isolating for the external noise, that's actually an interesting part. for the hd600 it doesn't stand, the isolation is pretty much zero until we reach a few khz. but sure some headphones do isolate a good deal, at least enough to dismiss some part of the ambient noises. that's a good point.
  
 but overall you still try to justify 24bit for the sound that can exist, not for the sound we can hear. and that's where we part ways. 
 look at the dynamic range actually used on your albums. I own a few operas and a good deal of classical stuff. I don't have a lot using more than 70db of dynamic. and the few I have, when you listen to the silence you hear a lot of noises. I don't know if it was in the room when recording or if it's something copied from a vinyl, a wax cylinder, or if the tapes got damaged. but all I get past 60/70db is usually noise. and again that must account for less than 0.5% of my albums. and I'm avoiding ultracompressed stuff as much as I can, listening mostly to old stuff as a strange consequence of the loudness war.
 but let's get past this, why restrict ourselves and sacrifice the 10albums that may have one day a use for 24bit dynamic? let's go for it.
  
 if your hears work ok, around 80/90db you will have the stapedius muscle kicking in to moderate the vibrations in your ears. that would turn into a temporary change of sensitivity.
 long story short, it's some kind of recalibration to avoid damages, just like the iris will do for brightness, except it's not an homogeneous response with the ears.
 and just like the stars you stop seeing in daylight because the iris adapts for the brightest part of what you see, past a certain level of sound, the stapedius will try to reduce sensitivity to accommodate for the loudest sound. so the loudest the sounds, the less sensitive you'll be to the quiet parts. if you listen to music with a 100DB dynamic, you stop being able to hear 1 or 2 or 5db sounds as a human reaction to loud sound. 
 that's what Steve Eddy is talking about, he's just mentioning the result of conducted experiments. 
 so that ends the debate about 24bit, nobody with normally functioning hears will hear a 15DB decay when listening to some 100 or 110DB music. and even better, given that you reduced you internal hear sensitivity, you also lost the ability to discern the same levels of variations. you're overal less sensitive to sound, it's a damn waste to actually listen loud for detail retreival.
  
 then there are all the possible noises and distortions that will probably pass 20db with high colors when the music goes past 100db. the HD800 at 100db is still above 0.1% of distortion, that's only -60db below the original signal(and that's the best part of the distortion, there is actually more than that up to almost -40db). do the math and cry. your quiet details will be masked by loudest sounds that are not even part of the music.
  
  
 and last but certainly not least, there is a limit to how much you can register at the same time on a conscious level. when listening to music you move freely from one instrument to another. to the voice, then the drums, something on the left, bam the singer in front. all those conscious or reflex choices make you concentrate mainly on those parts, dismissing most of the rest for sake of concentration. because after all we human can't think about 15things at once. one after another no problem, all at once... no can do.
 do you mean to tell me that you will focus on 25db decays when music is playing at 90db?  all of what makes the music will be in the loudest parts, naturally you would never do that(even if you could), your attention gets captured by one of those loud sound and makse you lose the last decay of another tone by choice.
  
  
 so the only situation where I could agree, ends up being some piece of music where we go from strong loud sounds, to a pretty long passage super quiet. that happens a lot on classical music, but did you look at the actual dynamic. usually the super quiet passage is -30 or -40db below 0. and for us i's already a great deal of difference, I often rise the volume for those passages. again my albums are around 60/70db of maximum dynamic at best. that leaves 26db  on 16bit tracks to define the quietest sound of the entire piece. how important is that part? how far away from anything louder was it? because if it's not isolated then depending on the frequencies, some masking effect might happen and hide that quietest part anyway. soetimes only something 20 or 30db louder are enough to mask the quiet part. making the quietest sound of the album something we cannot actually hear because of other sounds, so something useless again as this sound never existed for us humans in the first place.
  
 all this contribute to most of us in here saying that 16bit is actually enough for us. and people asking for more or pretending to hear more, are liars, aliens from another planet, or simply believe that something they hear is actually the excess bits, when it's something else. and I'm still waiting to see some evidence from the people claiming they do better than what doctors measured for us in experiments. until then I will keep my 16bit and my money.
  
  
  
 and about you first bit that makes for 0 or +6db...  the 1bit=6db is an estimate based on multibit systems, you mentioned it several times, that could work only on a 1bit dac that would happen to have that value of voltage/no voltage. but as soon as there are more than 1bit, that hypothetical +6db step is supported by smaller values and will be activated only when the smaller values are not enough to move to the next amplitude. you got something mixed up here obviously.
 with 5bits and a need for small steps variations at the lowest volume levels on the next sample, it will be the 4th and 5th bit that would change from 0 to 1 or 1 to 0. it's a pretty straightforward system, you want a given value, you use a combination of different discrete options to make the discrete value closest to the one you asked for.
 there will never be a moment where the smallest possible step between 2 discrete values in a 16bit dac will be +6db.


----------



## esldude

I think it is on some of the Wescott audio pages, but one fellow describes blind testing various amps and other components for the pro audio company he once designed for.  He told how it quickly became obvious that the very best, most discriminating blind tests were with volume no higher than 75 db (average level I think).  When they allowed people to set volume for a test he observed how they quickly raised volume to 85 or 90 db and within minutes had very poor ability to discriminate sound quality.   Things easily perceived with statistical validity at the lower volume were not discerned at those higher volume.  Considering peaks with a 75 db level of likely no more than 90 db, and a 30 db noise floor you aren't far from the 60 or so effective real time dynamic range of human hearing.  Which makes sense that would be the most precise range of listening. Low enough you don't often activate the muscle to reduce hearing sensitivity, and high enough to get the effective noise floor out of the noise floor of ambient sound.


----------



## stv014

> Originally Posted by *xdog* /img/forum/go_quote.gif
> 
> So lets assume that we have simple 'music' consiting of 2 sine waves.
> One of them has value of 1 (the smallest posible, the next value is 2 which is +6db more) and is the least hearable/persivable sine wave.
> ...


 
  
 All the above is only really relevant if no dithering is used. With dithering, the volume "steps" and distortion disappear, and there is only a constant noise floor.
  
 You can see this on the graphs below (click to zoom):
    
  
 This 44100 Hz/16-bit format sample is a mix of two sine waves, one has a frequency of 1000 Hz and a constant peak amplitude of 1024 (in 16-bit LSB units), and the other has a frequency of 1250 Hz and the peak amplitude increases exponentially from 0.25 to 2. Since dithering is used, there is a constant noise floor (hiss) without visible distortion products, and the level of the higher frequency tone increases smoothly without "steps". The graph on the right is a zoomed in version of the same sample.
  
 Without dithering, it looks like this:

 Now the noise floor is not as clean and consistent, but the amplitude still appears to increase continuously. This is possible because even the entropy from the higher level tone is enough for some dithering effect. In fact, with complex samples (like music when it is not at a very low level), dithering can be redundant, and just adds a small amount of extra noise. But it can be used to guarantee that quantization distortion is avoided.
  
 With the louder low frequency tone removed, the effect of dithering becomes much more obvious (left: not dithered, right: dithered):
    
 Without dithering, there is now high distortion, the amplitude increases in steps, and at the lowest levels the tone is cut off entirely. Dithering still produces a clean (other than the uncorrelated noise floor) output.


----------



## ab initio

stv014 said:


> Dithering still produces a clean* (other than the uncorrelated noise floor) *output.


 
  
 ... a noise floor at something like -120 dB, which is essentially non-existent! Do we even know of any equipment with a SNR > 120 dB?
  
 Great example of the effect of dithering on sinusoids vs more complex signals (aka, 2 sinusoids!). 
	

	
	
		
		

		
			




  
 What software were you using to create the plots? Was it audio-specific software or general purpose data analysis software?
  
 Cheers


----------



## Krutsch

stv014 said:


> All the above is only really relevant if no dithering is used. With dithering, the volume "steps" and distortion disappear, and there is only a constant noise floor.
> 
> <snip, snip>


 
  
 That was a very helpful explanation of the effects of dithering and I appreciate you taking the time to post the plots with a clear explanation.  
  
 All of the ranting is worth wading through for posts like these...


----------



## bigshot

Even without dithering, the noise is still inaudible under music.


----------



## stv014

ab initio said:


> ... a noise floor at something like -120 dB, which is essentially non-existent! Do we even know of any equipment with a SNR > 120 dB?


 
  
 The overall A-weighted level of the noise floor is actually about -97.3 dBFS, which is normally still more than good enough for music listening. The analysis displays it in relatively narrow bands (the "50 Hz bandwidth" on the graph means 6.02 dB attenuation at +/- 25 Hz from the signal, with a Gaussian window), in which the noise energy is obviously lower than over the entire 22050 Hz bandwidth of the sample. That is also why the tone at about -102 dBFS can still be clearly seen (and heard, with the volume turned up and the much louder 1 kHz tone removed to prevent masking), even though it is "under" the A-weighted noise floor.
  
 On the first graph, it can also be seen that there is more noise in the highest octave. This is intentional, to achieve a lower perceived noise loudness at the same unweighted RMS level.
  


> Originally Posted by *ab initio* /img/forum/go_quote.gif
> Great example of the effect of dithering on sinusoids vs more complex signals (aka, 2 sinusoids!).


 
  
 As shown, higher complexity can actually make dithering less important, as the lowest bits that are to be discarded by the quantization become more like white noise. But with proper dithering, the quantization error always just adds noise at a constant RMS level.
  


> Originally Posted by *ab initio* /img/forum/go_quote.gif
> 
> What software were you using to create the plots? Was it audio-specific software or general purpose data analysis software?


 
  
 I used the utilities from the link in my signature to generate and analyze the samples. In case anyone is interested, I still have the script that runs all the required commands to reproduce the samples and graphs.


----------



## Dark_wizzie

stv014 said:


> The overall A-weighted level of the noise floor is actually about -97.3 dBFS, which is normally still more than good enough for music listening. The analysis displays it in relatively narrow bands (the "50 Hz bandwidth" on the graph means 6.02 dB attenuation at +/- 25 Hz from the signal, with a Gaussian window), in which the noise energy is obviously lower than over the entire 22050 Hz bandwidth of the sample. That is also why the tone at about -102 dBFS can still be clearly seen (and heard, with the volume turned up and the much louder 1 kHz tone removed to prevent masking), even though it is "under" the A-weighted noise floor.
> 
> On the first graph, it can also be seen that there is more noise in the highest octave. This is intentional, to achieve a lower perceived noise loudness at the same unweighted RMS level.
> 
> ...


 
 Even after you broke it down with pictures, it all reads like Martian to me.


----------



## xdog

RE: castleofargh
 Actually I meant Takstars HD6000, not Sennheisers HD600 as the isolating headphones
  
 RE:stv014
 Thank you for changing rather abstract idea into such lovely plots


----------



## castleofargh

xdog said:


> RE: castleofargh
> Actually I meant Takstars HD6000, not Sennheisers HD600 as the isolating headphones
> 
> RE:stv014
> Thank you for changing rather abstract idea into such lovely plots


 

 ah yes, you talked about hd600 at the beguinning then I missread it when it came to hd6000 ^_^.


----------



## Army-Firedawg

Absolutely wonderful explination. Saved me a ton of data storage on my phone, dont have to buy the 128gb sd card now after all my 32 will be plenty.


----------



## jfaaz

I downsampled all my 24bit 96 & 192khz tracks to 24bit 48khz.  They all sound exactly the same as before and I saved about 250gb in disk space.  I'm going to stick with 16/44 and 24/48 from now on.


----------



## bigshot

I got a batch of 24/96 tracks and I am downsampling them to AAC. Sounds the same.


----------



## adisib

bigshot said:


> I got a batch of 24/96 tracks and I am downsampling them to AAC. Sounds the same.


 
  
 Is the time really worth the storage space and inability to convert to future potentially better codecs?


----------



## castleofargh

adisib said:


> bigshot said:
> 
> 
> > I got a batch of 24/96 tracks and I am downsampling them to AAC. Sounds the same.
> ...


 

 I don't think he meant he discarded the lossless copy. but even so, AAC is transparent to him, should he wait for "more transparent than transparent"? ^_^


----------



## bigshot

I am discarding the lossless. Why do I need a better codec than audibly transparent? I can re-encode AAC files ten times and not get any degradation. If I was going to remix them and it was a 24 track master, maybe I might want more latitude. But AAC is just as good as any lossless for the purposes of listening to music.


----------



## adisib

bigshot said:


> Why do I need a better codec than audibly transparent?


 
  
 I'm not aware of any CODECs that can't be audibly transparent. What if some other CODEC like Opus becomes superior at high bitrates in the future? Even AAC will improve. You will be stuck with the more space-consuming AAC. You will probably just say that that amount of space doesn't matter, you have plenty. I bet you have plenty of space for lossless as well. 20 GB can get you about 1000 FLACs.
  
 Sure AAC is good enough. But it seems pointless to spend all that time encoding when there is no _useful_ benefit for it. Its just a waste of time and electricity. Lossy is for portables where space does matter.


----------



## sonitus mirus

Lossy is also for streaming, which is how I most often listen to my music.


----------



## bigshot

adisib said:


> You will probably just say that that amount of space doesn't matter, you have plenty. I bet you have plenty of space for lossless as well. 20 GB can get you about 1000 FLACs.


 
  
 My media server is pushing two years worth of music. It's almost overrunning its 2TB drive as it is. In lossless, it would be very difficult to back up.


----------



## BlindInOneEar

bigshot said:


> My media server is pushing two years worth of music. It's almost overrunning its 2TB drive as it is. In lossless, it would be very difficult to back up.


 

 How long before we start seeing episodes of digital "Hoarders?"


----------



## bigshot

I DEFINITELY am a media hoarder!


----------



## castleofargh

I didn't know that name or even the fact that it was actually a sickness. very interesting.
 I seem to have suffered from the opposite sickness, getting rid of too much stuff with the false pretense to start anew. I think it's called "middle age crisis" ^_^.


----------



## L0SLobos

bigshot said:


> I DEFINITELY am a media hoarder!


 
 I must admit that I am the same, though I have taken steps forward in order to change by deleting any music I have not listened to in the past 6 months, as well as any movies I have not watched within a year.


----------



## bigshot

I have about a year and half's worth of music, so even if I played my music 24/7, I'd still have to cut a third of my collection!


----------



## francopro

jfaaz said:


> I downsampled all my 24bit 96 & 192khz tracks to 24bit 48khz.  They all sound exactly the same as before and I saved about 250gb in disk space.  I'm going to stick with 16/44 and 24/48 from now on.


 

 They will sound depending on your gears and music genres, if your dap/headphone /speaker can't produce the micro details between 48khz and 192khz , you won't hear them !
  
 I always found 192khz to be more airy with a better separation of instruments and clearer vocals. IMO, anything over 88khz would be hard to compare with 192khz , but you still can do it with 48khz


----------



## cjl

francopro said:


> They will sound depending on your gears and music genres, if your dap/headphone /speaker can't produce the micro details between 48khz and 192khz , you won't hear them !
> 
> I always found 192khz to be more airy with a better separation of instruments and clearer vocals. IMO, anything over 88khz would be hard to compare with 192khz , but you still can do it with 48khz


 
 And you've done proper level-matched double blind tests to verify that you can actually hear the difference, right?


----------



## Tuco1965

francopro said:


> They will sound depending on your gears and music genres, if your dap/headphone /speaker can't produce the micro details between 48khz and 192khz , you won't hear them !
> 
> I always found 192khz to be more airy with a better separation of instruments and clearer vocals. IMO, anything over 88khz would be hard to compare with 192khz , but you still can do it with 48khz


----------



## SilverEars

Check to see if better masterings are at high bitrate.  Typically dynamically compressed music is at lower bitrate.


----------



## kraken2109

silverears said:


> Check to see if better masterings are at high bitrate.  Typically dynamically compressed music is at lower bitrate.


 

 bitrate has nothing to do with dynamic range...


----------



## ab initio

francopro said:


> They will sound depending on your gears and music genres, if your dap/headphone /speaker can't produce the micro details between 48khz and 192khz , you won't hear them !
> 
> I always found 192khz to be more airy with a better separation of instruments and clearer vocals. IMO, anything over 88khz would be hard to compare with 192khz , but you still can do it with 48khz


 
  
 Do you use Foobar to play music?
  
 Foobar is an excellent free software that works with windows (and Linux via Wine) and has the ability to add plugins. One such plugin is the SoX resampling tool, which is an excellent resampler. Another tool is the ABX plugin which allows you to use double blind test whether you can detect the difference between two sound clips.
  
 I think it would be very helpful to you and to the rest of the community of you can test one of your 192kHz sampled tracks by comparing it to a down-sampled version. You can use the Sox DSP plugin to transcode your full-res track into a 48kHz version, and then perform an ABX test. If there is any detectable difference the two, then you should be able to differentiate between them in a statistically meaningful way. You can post the results from the ABX tool here to demonstrate the audible difference between a 192kHz track and its 48kHz equivalent. You need multiple trials (say 30) for the results to begin to be statistically significant (unless you can nail 20/20  )
  
 Furthermore, if you use <30 second clips, you can share the two tracks here so others can give it a try as well!
  
 Cheers


----------



## James-uk

http://www.theguardian.com/technology/2014/aug/21/mp3-cd-24-bit-audio-music-hi-res

A very misleading article. Listening set up by an audio store. No blind testing/ probably different masters used.


----------



## sonitus mirus

ab initio said:


> Do you use Foobar to play music?
> 
> Foobar is an excellent free software that works with windows (and Linux via Wine) and has the ability to add plugins. One such plugin is the SoX resampling tool, which is an excellent resampler. Another tool is the ABX plugin which allows you to use double blind test whether you can detect the difference between two sound clips.
> 
> ...


 
  
 I did a similar test using a 96kHz file.
  
 Here are the Cliffs Notes that were provided to me on how to set up the test.
  
 http://www.head-fi.org/t/570621/flac-vs-320-mp3/225#post_8900881


----------



## castleofargh

francopro said:


> ... if your dap/headphone /speaker can't produce the micro details between 48khz and 192khz , you won't hear them !


 
 when imagination becomes reality, walt disney would be so proud.


----------



## SilverEars

kraken2109 said:


> bitrate has nothing to do with dynamic range...


 
 Never said it was.
	

	
	
		
		

		
		
	


	




 Im trying to point out that it could be a coincidence a well mastered is of higher rate.


----------



## bigshot

Micro Details are too Micro.


----------



## Krutsch

I love the Sound Science threads.  At Head-fi.org, along with everywhere else on the Inner-tubes, we have an unsuspecting post like this one:
  
 Quote:


francopro said:


> They will sound depending on your gears and music genres, if your dap/headphone /speaker can't produce the micro details between 48khz and 192khz , you won't hear them !
> 
> I always *found 192khz to be more airy with a better separation of instruments and clearer vocals.* IMO, anything over 88khz would be hard to compare with 192khz , but you still can do it with 48khz


 
  
 ...followed, within seconds, with bot-like predictability, by a post that looks like this one:
  


cjl said:


> And you've done *proper level-matched double blind tests* to verify that you can actually hear the difference, right?


 
  
 ...capped off with something snarky, like this one:
  


castleofargh said:


> when *imagination becomes reality*, walt disney would be so proud.


 
  
 ...rinse and repeat.  Rarely is there any new information added to the discussion - it's like listening to Atheists debate religion with Christians that believe in a strict interpretation of the Bible.  
  
 Always entertaining, though ...


----------



## Lespectraal

Why pull ******** out of your ass


----------



## castleofargh

krutsch said:


> Rarely is there any new information added to the discussion - it's like listening to Atheists debate religion with Christians that believe in a strict interpretation of the Bible.


 
 what new information should we add, all has already been said a hundred times about how PCM really works. about how DACs never ever do anything like the stupid staircases graph shown anytime someone pretends to explain why hirez is better. the differences aren't in *micro details* because of how soundwaves work, the sound is the same and only some noise is added to it when some quantization errors occur. a noise people don't hear even in 16/44.
 about the low pass filters that works better with high sample rate? nobody uses NOS DACs nowadays so it doesn't matter. but people who don't understand PCM itself are unlikely to understand that part.
 about those who pretend to hear more air thanks to the ultrasonics of 192khz tracks, I say go see an audiologist and say that to his face.
 about those who pretend to here the subtle decays that are missing with "only" 96db of dynamic, same thing, go tell this to your audiologist. the guy is bored with old people all day, he needs a good laugh sometimes.
 then people don't believe in placebo effect when all the serious tests ever done show that people will believe anything if they first get the idea that it is better for reasons unrelated to sound.
 we also warn people about checking if the masters are the same, but that doesn't seem to prevent them from mistaking master and resolution and come say how vinyls or sacd or 24/192 are a better technology.
  
 all that is well known, all that has been researched in detail many years ago, yet people chose to believe advertising instead of the guys who actually invented digital audio. at some point we get bored, some here are very very patient, I am not.
  
 we don't add anything new because there is nothing new to add, sound didn't magically evolved in 2014. you want more read something, there are plenty of interesting yet simple enough to understand links across this topic. last stuff I read was this http://www.gearslutz.com/board/attachments/high-end/6491d1114045260-why-didnt-dsd-catch-reshaping_digital_-audio.pdf
 you get all you need to show how right you are about the superiority of hirez(that we never ever refuted as far as measurements go) and at least people can start to use actual reasons instead of pretending to hear ultrasounds and micro details. but then don't miss the end with the pie charts that show how much all this is a giant joke.
 I think my last post was right on the mark.


----------



## Krutsch

castleofargh said:


> <snip, snip>
> 
> *all that is well known, all that has been researched in detail many years ago*, yet people chose to believe advertising instead of the guys who actually invented digital audio. *at some point we get bored, some here are very very patient, I am not*.
> 
> <snip, snip>


 
  
 Exactly, so, why do you even bother to post a reply?
 As I said earlier, it's always entertaining to read a good rant


----------



## bigshot

We're always happy to explain the technical details if you don't understand. Just let us know. But otherwise we cut to the chase and point at what is causing the person's mistaken impression, not the underlying theory behind it.


----------



## Don Hills

lespectraal said:


> Why pull ******** out of your ass


 

 Why indeed. Let nature (and gravity) take its course.


----------



## sonitus mirus

don hills said:


> Why indeed. Let nature (and gravity) take its course.


 
  
 Asstronauts are probably thankful that gravity is not a factor when nature calls.


----------



## francopro

cjl said:


> And you've done proper level-matched double blind tests to verify that you can actually hear the difference, right?


 
  
 All tests have been done through different setups


----------



## francopro

ab initio said:


> Do you use Foobar to play music?
> 
> Foobar is an excellent free software that works with windows (and Linux via Wine) and has the ability to add plugins. One such plugin is the SoX resampling tool, which is an excellent resampler. Another tool is the ABX plugin which allows you to use double blind test whether you can detect the difference between two sound clips.
> 
> ...


 

 I don't trust Foobar anymore coz everytime I reconvert a track , even with hi-res the output file is a real disaster to my ears, as I don't use DSD so Foobar is useless to me
 I prefer VLC and Songbird for playback , DBamp for ripping and music conversion on my PC


----------



## francopro

francopro said:


> I don't trust Foobar anymore coz everytime I reconvert a track , even with hi-res the output file is a real disaster to my ears, as I don't use DSD so Foobar is useless to me
> I prefer VLC and Songbird for playback , DBamp for ripping and music conversion on my PC


 

 Furthermore , I do all tests with my desktop setup (Yamaha amp/ Beyerdynamic T5p/ yamaha SACD/AK100/Audioquest cables) , the PC is just useful to purchase and downloading music, it's very hard to setup an audiophile PC even with a good soundcard


----------



## ab initio

francopro said:


> I don't trust Foobar anymore coz everytime I reconvert a track , even with hi-res the output file is a real disaster to my ears, as I don't use DSD so Foobar is useless to me
> I prefer VLC and Songbird for playback , DBamp for ripping and music conversion on my PC




It sounds like you don't know how to configure foobar. Also, what does DSD have to do with anything? DSD is irrelevant to this issue of PCM formats.

Can you describe your issue? Because Foobar is a free player with realtime upsampling and ABX tool that none of your other options offer, making it the only useful option to you for determining the (in)audibility of hires PCM formats.

If you need help, ask questions. This community is friendly and happy to help.


Cheers


----------



## francopro

ab initio said:


> It sounds like you don't know how to configure foobar. Also, what does DSD have to do with anything? DSD is irrelevant to this issue of PCM formats.
> 
> Can you describe your issue? Because Foobar is a free player with realtime upsampling and ABX tool that none of your other options offer, making it the only useful option to you for determining the (in)audibility of hires PCM formats.
> 
> ...


 

 Yes , I don't know how to configure Foobar the way you mentioned above because i don't use this software !
 What I meant regarding DSD , I know that Foobar is one of the rare softwares that support this format, as I don't use DSD because of compatibility issues with my physical setups , that's why I don't need Foobar.
  
 Thanks anyway , maybe when I'll buy a monster like Alienware systems I may reconsider Foobar but for now my pc is 3 - 4 years old with outdated soundcard


----------



## esldude

francopro said:


> Furthermore , I do all tests with my desktop setup (Yamaha amp/ Beyerdynamic T5p/ yamaha SACD/AK100/Audioquest cables) , the PC is just useful to purchase and downloading music, it's very hard to setup an audiophile PC even with a good soundcard


 

 You could buy a USB to SPDIF converter of good quality which will do up to 192/24 for something like $100.  Connect PC (even an older one) via USB to the device, feed the digital signal to your Yamaha or the AK100 and you have an audiophile PC.  So not so hard.


----------



## SharpEars

OK, I am going to stick my face into the arena while the kicks are flying and say that I have a reason to prefer 24-bit over 16-bit. Here is why:
  
 My workflow consists of taking a 44.1/16 CD and ripping it into FLAC files. I then take the FLAC file for each track and:
  

Upsample to 24-bits
Remove DC Offset, if present
Reduce the volume until there are no clipped samples present
Possibly do heuristic based automated clip repair on those samples that were clipped
(optional) Normalize the volume to a reasonable level inline with the other tracks on the CD
  
 I perform functions 2-5 in the 24-bit domain and I leave the result as a 24-bit (FLAC) audio file that is ready for listening. I do this with all of my music and I stick with 24-bit, because I believe that the steps above performed in the 24-bit domain make more sense. I leave the resultant file as a 24-bit file, so that I can perform additional steps in the future that I deem will improve the tracks sound quality (e.g., equalization, excitement, whatever).
  
 Is my 24-bit workflow nonsense? Could all of this have been done at the 16-bit level with no possible way to tell the difference? Am I stupid in leaving the result at 24-bits and not downsampling it to 16, given that storage costs are negligible (e.g., 4 TB hard drive is $130)?
  
 Let the thread experts speak!


----------



## bigshot

It's unlikely that bumping up to 24 bit makes any difference there, except with the correction of the DC offset. Where is that coming from? Do you have a bad sound card? The clipping correction is all operating up at the top of the volume range, not down by the noise floor where 24 bit would make a difference.


----------



## esldude

Well doing processing on sound at the 24 bit level undoubtedly has lower levels of artifacts from DSP done.  It may or may not be audible, but as you said, given the low cost of storage and your desire to perform these operations it makes pretty good sense to me to do it at 24 bit and have no worries over the results. 
  
 Of course just for kicks, you could do one in 24 bit and one in 16.  When done ABX them to see if you hear a difference.  Myself I wouldn't bother, I would do it 24 bit like you are.


----------



## SharpEars

bigshot said:


> It's unlikely that bumping up to 24 bit makes any difference there, except with the correction of the DC offset. Where is that coming from? Do you have a bad sound card? The clipping correction is all operating up at the top of the volume range, not down by the noise floor where 24 bit would make a difference.


 
  
 The DC offset correction is due to the fact that most (especially pop) recording suffer a DC offset as can be measured over the entire track by any good audio editing program (e.g., Audacity, Audition, etc...). The same is true for determining whether or not there are any clipped samples. I try to make sure that the peak volume levels (not just RMS) do not exceed -0.25 dBFS.
  
 How does 24-bit better 16-bit when it comes to DC offset neutralization as you mentioned in your reply?


----------



## castleofargh

sharpears said:


> OK, I am going to stick my face into the arena while the kicks are flying and say that I have a reason to prefer 24-bit over 16-bit. Here is why:
> 
> My workflow consists of taking a 44.1/16 CD and ripping it into FLAC files. I then take the FLAC file for each track and:
> 
> ...


 

 that's pretty much why studios work on 24bit (or maybe 32now?) you minimize the size of errors while processing stuff, and you get the dynamic margin to move everything without having to worry about crushing the lowest sounds. I don't think anybody here would prefer to mix in 16bit.
  
 about going back to 16bit at the end or not. I have no doubt that there would be no audible difference one way or the other, so the choice is only laziness vs storage space ^_^.


----------



## SharpEars

So, based on the last several responses it seems that:
  

Listening to a 44.1/24 audio file is equivalent to listening to a 44.1/16 audio file, assuming both were properly created from the same master. This is regardless of the quality of DAC, amp and headphones/speakers.
However, if one wants to change the audio waveform (e.g., DC repair, volume correction, equalization, yada yada) then doing this stuff on an up-sampled 24-bit version of a 16-bit file sounds like a good idea in order to minimize
Leaving the file in 24-bit mode or downsampling back down to 16-bit is strictly a function of storage overhead and whether or not any additional editing will need to be performed on the file in the future.
  
 I guess all of that makes sense!


----------



## ab initio

sharpears said:


> OK, I am going to stick my face into the arena while the kicks are flying and say that I have a reason to prefer 24-bit over 16-bit. Here is why:
> 
> My workflow consists of taking a 44.1/16 CD and ripping it into FLAC files. I then take the FLAC file for each track and:
> 
> ...


 
 I have a nit to pick here.
  Quote:


> Upsample to 24-bits


 
 This makes zero sense.
  
 Upsampling involves changing the sampling rate of the audio. All you are doing by converting 16-bit to 24-bit audio is zero padding eight 0's to the end of your 16-bit PCM words. 
  
 My next question is this: why on earth do any of your CD's have a DC offset?  Let's say you have a 60 second sample with approximately 20 Hz content at 0dB full scale. your "DC offset" by chance can never be greater than that from a single half-cycle of this 20Hz info averaged over the length of your sample. In this example here, the DC offset of this sample is 1/(60*20*pi).
  
 If you compare that to 16bit depth, that DC offset works itself out to be 8.3 counts of audio. That's 8 out of the 65536 possible values of 16 bit audio! 
  
When you subtract the DC offest do you ever have audio clips with a bigger DC offest than 8 out of 65536? If so,
what CD's are these and who mastered them, because I would really like to avoid buying anything from them in
the future!
  
I guess my point is this: You aren't worried about the "DC offest" during the first 1/40th of a second at the start of a 
20Hz signal, so why would you care otherwise? Do you _really_ have tracks that you listen to with enough _extreme_ low 
frequency content at _significant_ amplitude such that there is appreciable "DC offset" at > O(1) sec? I guess I don't 
believe that any commerically released recordings contain an actual DC offest, nor to I believe that any competent 
audio chain passes true DC content.
  
 Again, 


> ... leaving the result at 24-bits and not downsampling it to 16


 
 You aren't "downsampling" here. You are downcoverting 24 bit to 16 bit. In this case, you would want to consider dithering between the steps.
  
 Cheers


----------



## bigshot

sharpears said:


> How does 24-bit better 16-bit when it comes to DC offset neutralization as you mentioned in your reply?


 
  
 DC offset is liable to be way down close to the noise floor of the recording. Clipping is up at the top, where 16 and 24 are identical.


----------



## SharpEars

ab initio said:


> I have a nit to pick here.
> This makes zero sense.
> 
> Upsampling involves changing the sampling rate of the audio. All you are doing by converting 16-bit to 24-bit audio is zero padding eight 0's to the end of your 16-bit PCM words.
> ...


 
  
 Before I start, I must apologize for using the term resample, I clearly meant up/down-convert the bit depth. There is no change in sample rate being discussed here.
  
 First of all on the subject of commercially released CDs, you would be surprised at how many releases have a measurable DC offset on their tracks - entire tracks, not portions thereof. I don't know who is mastering this stuff or encoding the content onto CDs, but they are leaving a CD offset on the music. I will not call out such CDs by name, but needless to say that pop and electronic music are ripe with DC offsets. I will admit it is not high, perhaps within 0.25%, sometimes within 0.10%. However, as part of my workflow I remove it. Perhaps an offset this low makes no difference and no audible clicks will be heard at the start/end of the track due to DC offset, but I zero it out in any case.
  
 Next, you should be far less surprised by the fact that many recordings these days have peak amplitudes on samples that result in clipping. I choose to lower the volume on these tracks so that there are no clipped samples. Certainly, no one will argue that (perhaps many) clipped samples may be audible, depending on how a particular DAC handles them as part of conversion to analog. Sometimes I lower the overall track volume further, especially if (perhaps due to compression) its overall loudness is obscene. It should come as no surprise that many popular recordings try to maximize the perceived volume as much as possible to sound loud, even at the expense of allowing clipping to happen. I try to undo some of this damage, within reason and opportunity. Sometimes, as I mentioned previously I try to bring tracks to a common reasonable average perceived volume.
  


bigshot said:


> DC offset is liable to be way down close to the noise floor of the recording. Clipping is up at the top, where 16 and 24 are identical.


 

  
 I understand that clipping is up at the top, but all samples are being shifted by a decimal dB amount. There must be some error in doing this. I have always thought that the error can be minimized by working at a higher bit depth. Perhaps I am wrong.
  
 Now on to the real question. Is there any value to performing my entire process in 24-bits, which involves multiple successive operations on the data, all of which introduce potential errors into the flow due to working with integers and rounding in the process? I don't know, that is primarily my question, does it matter or can multiple changes including possible equalization be done in 16-bits with no perceivable disadvantage to the 24-bit alternative. I am open to reasonable discussion on the topic that shows me the error of my ways.
  
 I am an audiophile with very high quality solid state equipment and headphones, but I am not willing to bury my head in the sand when a reasonable argument is presented that shows me that some of the steps I take are of dubious value. I would just like to understand the logic and thought behind them.


----------



## castleofargh

sharpears said:


> Before I start, I must apologize for using the term resample, I clearly meant up/down-convert the bit depth.


 
 no too late BURNNNNNN!!!! ^_^ we all understood what you meant it wasn't much a problem. still ab initio was right to point it out and was less lazy than I was.
  
  
  


sharpears said:


> ...can multiple changes including possible equalization be done in 16-bits with no perceivable disadvantage to the 24-bit alternative.


 
 only you can tell us. do one track with all your process in 16bit then in 24bit and then ABX the hell out of those.


----------



## SharpEars

castleofargh said:


> no too late BURNNNNNN!!!! ^_^ we all understood what you meant it wasn't much a problem. still ab initio was right to point it out and was less lazy than I was.
> 
> 
> 
> only you can tell us. do one track with all your process in 16bit then in 24bit and then ABX the hell out of those.


 
  
 LOL! In all seriousness, I hate ABX tests and the other problem is that just because I fail at ABXing one track, there's no guarantee that on another track I won't pass. I'd have to ABX every single track I do to see whether this process affected it and I'd have to do it at the right volume level (i.e., 75 dB) to avoid fatigue, which is lower than my listening level. I was hoping for a technical explanation as to why it doesn't matter that convinces me I am wasting my time (up converting).


----------



## kraken2109

Can somebody explain in basic terms what DC offset is? I've seen the option in software but never knew what it was.


----------



## SilverEars

kraken2109 said:


> Can somebody explain in basic terms what DC offset is? I've seen the option in software but never knew what it was.


 
 It means the wave's center is not fixed at the 0 level.  I would think it's a universal term used in electronics or audio samples as it has to do with vertical shift of the wave.  Just like what is taught in trig.  1+sinx will just be the wave shifted up 1 from 0.  1 is the offset.  I found this online.  The waveform on top is offset, and below, it's corrected.


----------



## kraken2109

silverears said:


> It means the wave's center is not fixed at the 0 level.  I would think it's a universal term used in electronics or audio samples as it has to do with vertical shift of the wave.  Just like what is taught in trig.  1+sinx will just be the wave shifted up 1 from 0.  1 is the offset.  I found this online.  The waveform on top is offset, and below, it's corrected.


 

 Ah that makes sense, thanks


----------



## SharpEars

OK to ruffle some feathers, I want to ask a provocative question. Why do we need 16-bit audio, won't 8-bit audio make do? After all, if most recording utilize somewhere around 50 dB of dynamic range and given the ""noisy environments that even the quietest listening rooms are, having 48 dB with 8-bit audio should practically suffice. So, why do we need 16-bit at all - isn't it overkill?
  
 For example, take the 8-bit vs 16-bit test at: http://www.audiocheck.net/blindtests_16vs8bit.php
  
 By the way, even with my high-end equipment I failed the test 
	

	
	
		
		

		
			





 (4/10).
  
 I also learned that I can only hear noise down to -54 dBFS using this test: http://www.audiocheck.net/blindtests_dynamic.php?dyna=54
  
 Equpment used for the test: PC USB (MME) -> OPPO HA-1 (DAC/Headphone Amp) -> Balanced headphone out to Sennheiser HD650 headphones via balanced cable
  
 I set the volume level pretty high for the test.
  
 I think I just ended my long career of being an audiophile...


----------



## James-uk

This statement sums digital audio up for me. 

"Remember, it has been 30 years since the introduction of the CD technology and we have yet to see credible evidence to demonstrate that well-digitized 16/44 isn't transparent beyond anecdotal opinions (just like there's no good evidence to demonstrate superiority of 24-bits or >44kHz sampling rates; assuming we're using a decent DAC playback system)."

Taken from this article. 
http://archimago.blogspot.ca/2014/08/musings-pure-perfect-sound-forever.html?m=1


----------



## bigshot

sharpears said:


> I understand that clipping is up at the top, but all samples are being shifted by a decimal dB amount. There must be some error in doing this. I have always thought that the error can be minimized by working at a higher bit depth.


 
  
 The difference between 16 bit and 24 bit is way down near the depth of the noise floor. Up at the higher volumes, the sound is identical. Working at 24 bit wouldn't give any benefit to fixing clipping, but it would involve more data processing, increasing the chance of error. The reason that studios use 24 bit is because in a mix, they often have to bring the volume of a particular element in the mix up, and when they do that, they drag the noise floor up along with it. It's better for them to keep the sound clean beyond the range of hearing. But for playing back music at normal listening volume, it doesn't make any difference at all.


----------



## castleofargh

sharpears said:


> OK to ruffle some feathers, I want to ask a provocative question. Why do we need 16-bit audio, won't 8-bit audio make do? After all, if most recording utilize somewhere around 50 dB of dynamic range and given the ""noisy environments that even the quietest listening rooms are, having 48 dB with 8-bit audio should practically suffice. So, why do we need 16-bit at all - isn't it overkill?
> 
> For example, take the 8-bit vs 16-bit test at: http://www.audiocheck.net/blindtests_16vs8bit.php
> 
> ...


 

 8bit would mean quantization errors around -48db (8*6) that would be as you yourself have tested in the audible (maybe with dither it can sound better). but else for a lot of modern songs 8bit would be more than enough ^_^.
 you should always test at listening levels, testing louder is often a way to hear less.


----------



## bigshot

8 bit isn't too far from what an LP record would be.


----------



## SharpEars

castleofargh said:


> 8bit would mean quantization errors around -48db (8*6) that would be as you yourself have tested in the audible (maybe with dither it can sound better). but else for a lot of modern songs 8bit would be more than enough ^_^.
> you should always test at listening levels, testing louder is often a way to hear less.


 
  


bigshot said:


> 8 bit isn't too far from what an LP record would be.


 
  
 So, basically 10-bit audio with 60 dB of headroom and proper dither would be indistinguishable from 16-bit in a normal listening environment, no matter how good one's audio gear is. I cannot hear -60 dB noise with the volume cranked up higher than even my loudest listening levels. Even if the noise is shaped/dithered down to -54 dB it is for all practical purposes inaudible in a normal listening environment due to being noise shaped out of the sensitive portions of the spectrum.


----------



## nick_charles

bigshot said:


> 8 bit isn't too far from what an LP record would be.


 
  
 That is a bit unfair. LP can do up to 80db (13 bits)  in extremis,  more routinely you could expect 55 - 70 db ( 9 - 11) - this is fine most of the time but not for stuff like some classical music which can go from deafeningly loud to whisper quiet in the same movement.
  
 My very first experience with CD was back in 1984 when I auditioned an early Marantz CD63 (Philips CD100) a modest 14 bit 2x oversampling box with a humble 90db snr and I got them to put on Solti's CSO Mahler 1 - the lack of noise on the opening bars of the 1st movement compared to my Rega Planar 3 figuratively took my breath away - that pretty much finished LP for me


----------



## bigshot

In practice, LPs, even well mastered classical ones, generally have around 45-55 dB of dynamic range. Peaks may be cut in hot to go higher, but they distort, and stuff down in the quiet range below that are usually buried in surface noise. If your LP of Solti's Mahler had 70dB of dynamic range, you wouldn't have been nearly as impressed with a 90dB CD.


----------



## nick_charles

bigshot said:


> In practice, LPs, even well mastered classical ones, generally have around 45-55 dB of dynamic range. Peaks may be cut in hot to go higher, but they distort, and *stuff down in the quiet range below that are usually buried in surface noise*. If your LP of Solti's Mahler had 70dB of dynamic range, you wouldn't have been nearly as impressed with a 90dB CD.


 
  
 I have no way of knowing what the range of the LP was but I do remember the quiets parts were noisy as were all my classical LPs (in the quiet parts)  and it was not just surface noise, poor tracking, badly pressed centers, rumble,  it drove me bananas especially after plonking down what was for me back then almost a month's salary to upgrade from a Transcriptors Saturn !


----------



## SharpEars

nick_charles said:


> That is a bit unfair. LP can do up to 80db (13 bits)  in extremis,  more routinely you could expect 55 - 70 db ( 9 - 11) - this is fine most of the time but not for stuff like some classical music which can go from deafeningly loud to whisper quiet in the same movement.
> 
> My very first experience with CD was back in 1984 when I auditioned an early Marantz CD63 (Philips CD100) a modest 14 bit 2x oversampling box with a humble 90db snr and I got them to put on Solti's CSO Mahler 1 - the lack of noise on the opening bars of the 1st movement compared to my Rega Planar 3 figuratively took my breath away - that pretty much finished LP for me


 
  
 After it goes below -54 dBFS it's irrelevant, classic music or any other music. To hear it (not to mention hear it loud enough to appreciate it) at that level you would have to increase the volume to such a ridiculous amount that as soon as a loud passage started you would be deafened.
  
 -54 dBFS requires only 9-bits (You can add an extra bit for good measure, more room to dither, yada yada), so a 10-bit ADC / 10-bit DAC converter is transparent for music.
  
 Virtually all recordings have less than 60 dB of dynamic range. It's easily measured in any good DAW on a track by track basis. Try it with the music you listen to that you consider has a higher dynamic range - you will be surprised! I am including audiophile 192/24 recordings in the mix by the way, so do mean virtually all.
  
*Updated with details on how you can perform the test yourself...*
  
 Anybody that thinks that more than a 10-bit sample depth matters needs to watch the following video: https://www.youtube.com/watch?v=BYTlN6wjcvQ starting from 45:48. Let your own ears with your own (high end) equipment be the judge.
  
 Actually, if you want to do this test with complete accuracy, you can download the original .wav file used for this test at: http://ethanwiner.com/aes/bit_reduction.wav
  
 I start hearing noise at around 18 seconds when playing the .wav file which equates to a 7-bit depth, when listening on Sennheiser HD650 headphones connected via a balanced cable to an OPPO HA-1 fed via asynchronous USB (i.e., I don't think anyone can call my system "low res") with the volume set quite loud in a very quiet room. Let me repeat, 7-bits is enough for this song!
  
 Now if you want to do the test yourself, get the .wav file at the link I just posted, see at what second you can hear noise or any objectionable artifacts. Then play the youtube video (also linked above) starting at 46:18 for the number of seconds you played the .wav file. Then you can see from the video at what bit depth you heard the "bad audio." That my friends is the easiest way to convince yourself that 10-bits is plenty.
  
*Updated again:*
  
 If you really want to go all the way, you can download the actual VST plug-in called *+decimate* that was used in the instructional youtube video and try it with your own DAW and your own music. I would love to hear the results. In fact, I've done all of the research for you.
  
 Here is a link to the latest version of the VST collection that includes +decimate: http://www.soundhack.com/freeware/
  
 You want to download the *Delay Trio / Freesound Bundle* from the top left column on that page. The actual plug-in you're looking for from the set is +decimate and can be found under VST/Effect/Sound Hack/+decimate in your DAW after it is correctly located and installed in your DAW software. On windows when I install it, it installs itself in c:\program files\common files\VST2, so I just added that redirectoy to my DAW and refreshed the VST list making it available.
  
 The funny thing is that I have some music high in transients that I thought could use some major (i.e., 24) bit depth and it turned out that 5-bits was enough! I am both flabbergasted and speechless at this point. How can anyone even consider high bit depth audio again after performing this test?
  
 Happy listening! I am beginning to think that this post should be linked to from the top post in this thread for all of the audiophiles that venture into these (murky) waters to get permanently "circumcised" of their (high-end) purchasing habits.


----------



## castleofargh

sharpears said:


> nick_charles said:
> 
> 
> > That is a bit unfair. LP can do up to 80db (13 bits)  in extremis,  more routinely you could expect 55 - 70 db ( 9 - 11) - this is fine most of the time but not for stuff like some classical music which can go from deafeningly loud to whisper quiet in the same movement.
> ...


 

 I wouldn't be so extreme. if you don't get more than -54 for simultaneous music, it doesn't mean that on some classical piece or other you won't have a strong part reaching 0db and a calmer part where the loudest sounds will stay below -10db or -15db. so while listening to that part, you will again get close to your 54db of usable dynamic for yourself (Steve Eddy said 60db max for average humans so your number seems to fit nicely as you said you tried the test loud). so all in all to use most of your hearing potential, the track could have a use for around -64db to -69db in that fake example.
 I think the usually accepted 80db is a nice value with a safe margin for change of rhythm in a song or album.
  
 but hey we were happy with tapes when our average walkman must have been also under 10bits. so to me it is mostly a problem of hiss level. at least that's why I gave up on vinyls and tapes.


----------



## SharpEars

castleofargh said:


> I wouldn't be so extreme. if you don't get more than -54 for simultaneous music, it doesn't mean that on some classical piece or other you won't have a strong part reaching 0db and a calmer part where the loudest sounds will stay below -10db or -15db. so while listening to that part, you will again get close to your 54db of usable dynamic for yourself (Steve Eddy said 60db max for average humans so your number seems to fit nicely as you said you tried the test loud). so all in all to use most of your hearing potential, the track could have a use for around -64db to -69db in that fake example.
> I think the usually accepted 80db is a nice value with a safe margin for change of rhythm in a song or album.
> 
> but hey we were happy with tapes when our average walkman must have been also under 10bits. so to me it is mostly a problem of hiss level. at least that's why I gave up on vinyls and tapes.


 
 I can't hear hiss or a loss of quality even at 8-bits with the test I posted in my updated post.


----------



## Krutsch

esldude said:


> You could buy a *USB to SPDIF converter of good quality which will do up to 192/24 for something like $100*.  Connect PC (even an older one) via USB to the device, feed the digital signal to your Yamaha or the AK100 and you have an audiophile PC.  So not so hard.


 
  
 Really,  $100.00 US?  Please recommend some, I am in the market to replace my NuForce u192s which has problems with drop-outs/artifacts and everything I've looked at is in the $200 - $400 range (e.g. Wyred4Sound, iFi Audio, Bel Canto - I have one of the latter, love it, but it's a little pricey).


----------



## nick_charles

sharpears said:


> After it goes below -54 dBFS it's irrelevant, classic music or any other music. To hear it (not to mention hear it loud enough to appreciate it) at that level you would have to increase the volume to such a ridiculous amount that as soon as a loud passage started you would be deafened.
> 
> -54 dBFS requires only 9-bits (You can add an extra bit for good measure, more room to dither, yada yada), so a 10-bit ADC / 10-bit DAC converter is transparent for music.


 
  
 I re-ripped Mahler 1 1st mvt (1984 CD) and ran it through Audacity - interestingly there was less dynamic range than I had thought
  

  
 Above - the opening bit - ( I trimmed the opening silence) peaks at about -39db RMS at about -50db
  
 below the loudest bit - hitting the end stops - RMS about -12db
  

  
  
 Then i thought I would try the 2007 reissue - which was pretty much the same
  

  
  

  
 so about 39db whichever way you look at it  !
  
 For fun I reripped it as MP3 v0
  

  
 very similar but.... on the loud bits
  

  
 Not clipping according to Audacity but right on the limit more often than with either Cd copy
  

  
  
 Then I though about how quiet i could hear at normal levels - I took the Cd rip and applied a -10db amplification
  

  
 This was absolutely on the limit to be able to hear all the instruments - I lowered it another 10db and could still hear _something_ but I definitely lost some instruments


----------



## bigshot

On vinyl, a 45 to 55dB dynamic range would be a half speed mastered pressing with lots of headroom. I had a Teldec LP of the Bizet's Carmen Suite that had a bass drum wallop that was completely untrackable by even the best turntables. (On CD it didn't sound particularly out of the ordinary, but on LP it was too much bass energy too close to the center groove where tracking and resolution are at their lowest.) A normal record from the late 70s or so would be pretty close to 8 bit. particularly a record with more than 17 minutes on a side. The surfaces during the oil crisis were particularly bad. Most CDs don't exceed 55dB. The only ones I can think of that do that on a regular basis are BIS classical recordings, but those are often *too* dynamic for comfortable listening. It's easy to fall into the "bigger numbers are better" trap. But the truth is that balance in the sweet spot is much more important than the extremes.


----------



## Don Hills

silverears said:


> It means the wave's center is not fixed at the 0 level.  I would think it's a universal term used in electronics or audio samples as it has to do with vertical shift of the wave. ...


 
  
 Also, if you perform spectrum analysis of a signal with DC or near DC content, you'll see the level increase at the 0 Hz end of the display. I see this quite often on commercial productions. Here's a particularly bad example:
  

  

  
 It's a quite sensitive method of checking for offset. It readily shows offset that doesn't show up on a waveform display.


----------



## ferday

don hills said:


> Also, if you perform spectrum analysis of a signal with DC or near DC content, you'll see the level increase at the 0 Hz end of the display. I see this quite often on commercial productions. Here's a particularly bad example:
> 
> 
> It's a quite sensitive method of checking for offset. It readily shows offset that doesn't show up on a waveform display.




Informative post, thanks

I've seen probably hundreds of spectrum analysis in audacity with the 0hz peak and always wondered why but never bothered to look into it, in the waveform it always seems so close to center (and nothing audible to me). 

Makes perfect sense I just never made the (now obvious) connection. I sincerely doubt it means much but it might make a fun ABX to geek out with


----------



## HWTest

krutsch said:


> Really,  $100.00 US?  Please recommend some, I am in the market to replace my NuForce u192s which has problems with drop-outs/artifacts and everything I've looked at is in the $200 - $400 range (e.g. Wyred4Sound, iFi Audio, Bel Canto - I have one of the latter, love it, but it's a little pricey).


 

 Musiland Monitor 01 USD - the only downside is it is Windows only.
  
 http://www.head-fi.org/t/423960/musiland-monitor-01-usd-24-192-usb-to-spdif
  
 http://www.amazon.com/Musiland-01USD-Digital-Stereo-384khz/dp/B00ACG5K3O


----------



## SharpEars

krutsch said:


> Really,  $100.00 US?  Please recommend some, I am in the market to replace my NuForce u192s which has problems with drop-outs/artifacts and everything I've looked at is in the $200 - $400 range (e.g. Wyred4Sound, iFi Audio, Bel Canto - I have one of the latter, love it, but it's a little pricey).


 
  
 How about this littly device for for $55:
  
http://www.amazon.com/192Khz-Coaxial-Optical-Headphone-MUSE-MiniUSBDAC-Silver/dp/B00BN39RSE


----------



## esldude

krutsch said:


> Really,  $100.00 US?  Please recommend some, I am in the market to replace my NuForce u192s which has problems with drop-outs/artifacts and everything I've looked at is in the $200 - $400 range (e.g. Wyred4Sound, iFi Audio, Bel Canto - I have one of the latter, love it, but it's a little pricey).


 

 Well I guess this one is more like $149, but it is an asynchronous converter.  I had the other Peachtree in mind, and it is $99, but it isn't asynchronous.  So a bit of confusion on my part.  My experience has been the asynch models generally have no problems with drop outs or other glitches.  Other approaches used sometimes work fine and sometimes don't with the few I have used or known about.   Of course your NuForce I believe is asynch itself.  Unusual that it would suffer drop outs.
  
 http://www.crutchfield.com/p_731X1/Peachtree-Audio-X1.html?tp=60317
  
 Also if you look a bit you can find 2nd hand Music Fidelity V-links which were asynchronous USB.  I know people with those and they don't have problems with dropouts or other issues.  Work with Mac, Windows or Linux.


----------



## Krutsch

^^^ thanks, everyone, for the helpful suggestions.  Food for thought...


----------



## Kaffeemann

nick_charles said:


> For fun I reripped it as MP3 v0
> 
> 
> 
> ...


 
  
 Keep in mind that there might be intersample clipping. Audacity does not display the real waveform so you can't see it.
 This is how extreme intersample clipping looks like:

  
  
 Back to topic. Benchmark Media Systems published this article on the bit depth stuff:
http://benchmarkmedia.com/blogs/news/15121729-audio-myth-24-bit-audio-has-more-resolution-than-16-bit-audio


----------



## audiosampling

Just a little intrusion here, to tell that the 16-bit v/s 8-bit blind test page that has been often cited in this thread, has been updated. Now using Neil Young's own music. I am such a bad guy indeed...
  
 http://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php


----------



## Hungryhoss

audiosampling said:


> Just a little intrusion here, to tell that the 16-bit v/s 8-bit blind test page that has been often cited in this thread, has been updated. Now using Neil Young's own music. I am such a bad guy indeed...
> 
> 
> 
> ...



 


Genius. I am glad you are going there. I look forward to the result.


----------



## Yourt

I've started reading and advanced well through pages 1 tot 20. then I started skipping some parts of the thread, ending up reading the last set of pages (boy, what a job, let alone the fact that some clicking to referenced pages need sto be done too).
  
 What I already knew got confirmed now in a scientific way. One can indeed show and proof technical / theoretical (show meaning either audible or (mostly) in statistics) differences between different formats, both in dynamic range (bit depth) as in bitrate (the sampling).
  
 Fact is that for real life use there are very little musical works that could benefit from any of these differences, because of a bunch of limitations in the chain (production, reproduction, sensitory system and psyche of the listener).
  
 My concerns were at the stated "subharmonics and overtones, 30K cymbal stuff and some more of those together with the influence of the sampling into the lower frequencies (teh last one been shown to be able to sometimes even worsen the final result).
  
 Considering phono... there's more to it than just LP DR figures: RIAA curves and preamp rumble filters with low frequency rolloff have an impact on the final sound shaping. Anyhow, a good player attached to a nice set of preamp-amp-LS has very little hissing or noise (same goes as above: crank up the volume and hear the noise, and next when the music kicks in hear a beep...  ).
  
 As for commercial reasons: I got a recording from the library last week, SA-CD. So I was figuring: let's give it a go for some hi-res testing. No need to, because after a simple check on my laptop (Audacity) it turned out to be 16/44 material.
  
 So I'll leave it at looking for good quality material (starting at good recording and mastering and decent pressing/burning/ripping) and enjoy music at 16/44, or eventually - but not necessarily - higher.


----------



## azteca x

> As for commercial reasons: I got a recording from the library last week, SA-CD. So I was figuring: let's give it a go for some hi-res testing. No need to, because after a simple check on my laptop (Audacity) it turned out to be 16/44 material.


 
 Your computer was reading the CD layer. SACDs (which are encoded as DSD) can only be played in authorized SACD players and ripped by early PS3s. So yes, of course the files you got were 16/44.1.


----------



## Dark_wizzie

audiosampling said:


> Just a little intrusion here, to tell that the 16-bit v/s 8-bit blind test page that has been often cited in this thread, has been updated. Now using Neil Young's own music. I am such a bad guy indeed...
> 
> http://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php


 
 It's a convenient test but somebody criticized the test because it ain't high DR content. (Or something.) But that obviously didn't get the guy to conduct his own Foobar abx test.


----------



## RRod

dark_wizzie said:


> It's a convenient test but somebody criticized the test because it ain't high DR content. (Or something.) But that obviously didn't get the guy to conduct his own Foobar abx test.


 
  
 The page itself says that you can readily hear the effect of 8-bit, but that the effect lessens with much popular music, exactly because it often isn't high DR content.  It would be interesting to have an example of, say, the ending of Mahler's 6th in 16-bit-noise-shaped vs. 24bit.


----------



## Hudson

audiosampling said:


> Just a little intrusion here, to tell that the 16-bit v/s 8-bit blind test page that has been often cited in this thread, has been updated. Now using Neil Young's own music. I am such a bad guy indeed...
> 
> http://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php


 
  
 Is the 8-bit sample dithered and noise shaped?


----------



## audiosampling

No... because noise shaping would add distinctive noise in this case (during the fade out at the end of the sample... one would fade into noise)... and make the blind test too easy to succeed


----------



## jcx

that seems backwards - truncation or rounding adds distinctive spitting, hissy amplitude jumps during fades, good dither gives smooth music signal fades below the lsb - but can add audible but constant noise that we can easily "listen through"
  
  http://audio.rightmark.org/lukin/dither/dither.htm
  
 noise shaping is applied with dither, changes the dither noise spectrum to be lower where our hearing is most sensitive, reducing dither noise audibility
  
  
 I bet few would ever pass a 8/192 shaped dither comparisons with music playing at reasonable levels (if their setup didn't distort, create audible IMD difference products with the ultrasonic dither)
  
http://www.meridian-audio.com/w_paper/Coding2.PDF


----------



## Hudson

audiosampling said:


> No... because noise shaping would add distinctive noise in this case (during the fade out at the end of the sample... one would fade into noise)... and make the blind test too easy to succeed


 
  
 Are you sure? Is it your website? So no dithering either? I just want to clarify as I would have expected noise shaping to reduce the amount of audible noise during the fadeout, not increase it. Cheers.
  


jcx said:


> that seems backwards - truncation or rounding adds distinctive spitting, hissy amplitude jumps during fades, good dither gives smooth music signal fades below the lsb - but can add audible but constant noise that we can easily "listen through"
> 
> http://audio.rightmark.org/lukin/dither/dither.htm
> 
> ...


 
  
 This makes more sense to me.


----------



## stv014

No dither is not always necessarily a problem, especially compared to simple non-shaped dither. While it obviously makes a major difference with artificial test signals, complex music that is never very quiet (e.g. during a long fade-out to silence at the end) may have enough entropy in the least significant bits that it essentially becomes "self-dithered". Basic dithering always adds a few dB of extra noise floor (which can make an audible difference at 8-bit resolution), but it is "safer" because it guarantees that the quantization error is uncorrelated white (if not shaped) noise. By the way, the Meyer&Moran Red Book vs. SACD tests did not use dithering either.
  
 You can test dithering to various resolutions vs. no dithering with the samples from this older thread. The original 24-bit sample has been quantized from 16 to 8 bits with a very simple "shaped" triangular dither, and the 10-bit version is also available with white noise TPDF dither, uniform dither, and no dither. Try listening to the files, or extract the residual with an audio editor. Ideally, it should sound like pure noise, with no recognizable bits of the original signal.


----------



## Hudson

stv014 said:


> No dither is not always necessarily a problem, especially compared to simple non-shaped dither. While it obviously makes a major difference with artificial test signals, complex music that is never very quiet (e.g. during a long fade-out to silence at the end) may have enough entropy in the least significant bits that it essentially becomes "self-dithered". Basic dithering always adds a few dB of extra noise floor (which can make an audible difference at 8-bit resolution), but it is "safer" because it guarantees that the quantization error is uncorrelated white (if not shaped) noise. By the way, the Meyer&Moran Red Book vs. SACD tests did not use dithering either.
> 
> You can test dithering to various resolutions vs. no dithering with the samples from this older thread. The original 24-bit sample has been quantized from 16 to 8 bits with a very simple "shaped" triangular dither, and the 10-bit version is also available with white noise TPDF dither, uniform dither, and no dither. Try listening to the files, or extract the residual with an audio editor. Ideally, it should sound like pure noise, with no recognizable bits of the original signal.


 
  
 I see. I understand the theory but I'm just curious to see how low a resolution I need to go to before I'm able to perceive an audible difference.
  
 I will take a look at those files and abx when I get a chance. For no real reason I always expected quantization noise on a non-dithered low res sample to be more noticeable, but I guess not. I can't tell the difference between those files on the audiocheck website.
  
 Cheers


----------



## castleofargh

I guess at some point it ends up being a question of listening volume levels and how quiet your room is. I know that with 16bit what I usually end up hearing first is noise from the amp section, and going 24bit doesn't help a bit
	

	
	
		
		

		
			





.


----------



## audiosampling

hudson said:


> Are you sure? Is it your website? So no dithering either? I just want to clarify as I would have expected noise shaping to reduce the amount of audible noise during the fadeout, not increase it. Cheers.
> 
> 
> This makes more sense to me.


 

 Yes, this is my website. And yes, I am 100% sure: there is no dithering applied. Actually, I did apply dithering first, but the fade out (at the end of the extract) would then fade out into audible noise (with or without shaping). So, it would have been very easy for people to find out which was the 8bit version, by simply listening to the fade out. That's what I wanted to avoid and why I removed the dithering.
  
 I am very aware of what dithering does and how it works. BTW, I wrote the following page, providing audio examples and audible tests:
  
 http://www.audiocheck.net/audiotests_dithering.php
  
 It shows how dithering, applied to an 8-bit file, increases the dynamic range by 18dB!


----------



## Hudson

audiosampling said:


> Yes, this is my website. And yes, I am 100% sure: there is no dithering applied. Actually, I did apply dithering first, but the fade out (at the end of the extract) would then fade out into audible noise (with or without shaping). So, it would have been very easy for people to find out which was the 8bit version, by simply listening to the fade out. That's what I wanted to avoid and why I removed the dithering.
> 
> I am very aware of what dithering does and how it works. BTW, I wrote the following page, providing audio examples and audible tests:
> 
> ...


 

 Thanks for confirming. I also took a look at the other page. Yeah, fading into audible noise would be more obvious i guess, I was just suprised by how inaudible the quantization noise was in the tracks.
 It's an interesting website so thanks, I'm curious to know how many people can tell pass the abx test using those tracks though.


----------



## bigshot

Ethan Winer covers dithering in his Audio Myths video. See my sig for a link.


----------



## k00zk0

Check out these couple tests. One is from Reddit to which I replied to elaborate on and summarize:
  
 From:

 https://www.reddit.com/r/audiophile/comments/2n5zwe/beating_the_24bit_horse/



> Do you think you can hear the difference between 24- and 16- bit audio? In that case, I have prepared a torture test for you. Here is a zip file (stored with -0, so no compression) with set of two 24-bit wav files (16MB).
> 
> In it are two 24-bit audio named x[six-digit number].wav that contain the following:
> 
> ...


 
  
  
 This is incredible. Let me explain what was done to make this test with another file you can use to test the theory even more.



> Here is a 24-bit flac, where the data has been attenuated so it roughly fits into the 9 bits. This file was posted by Arve earlier.


 This file has a song with the music volume turned so low that there is only data in the least significant 9 out of 24 bits of the sound file. If it was a 16-bit file at the same volume, we would only have the last bit of audio data present which would encode basically the presence of what would sound like a tick or a pop - the quietest possible sound that could be in the file. Any lower is pure silence. In the 24-bit file though, the last 9 bits allow an actual recognizable song to be hidden in there instead of just ticks, so we can use it to check if we are able to actually hear a real song buried deeper than what's possible in a 16 bit file.

 Normally I run my receiver at level 50 (I am not sure if this corresponds to db) for medium-loud near-field listening (I sit about 1 meter from the speakers). Often I use it at about 43-45 for medium volume listening. The receiver is turned up so high because it gets a quiet input due to an EQ that turns the total level down by approx 15db.

 When I play the file at my normal loudish volume, I hear nothing since it is so quiet. If I turn the receiver up to max which is level 74, AND disable the DSP (so it is total 40-45db higher than normal level), and all of this with the computer output volume as high as possible, I at first hear nothing from the speakers except noise from the receiver being turned to max. And my quiet fridge humming away 7 meters away across the room. This is absolutely as loud as this 2x80W system goes.

 If I put my ear up to the speakers mid/tweeter, I can hear the song through the speaker hiss. But when I sit at a normal position, 1 meter away from the speakers, I can't hear anything but the noise that's a side-effect of my room and having the speakers so loud.

 There's another file with the music in the 8 least significant bits, with the music half as loud, as expected. I could barely pick the song out of the hiss with my ear right up to it. Mind you my receiver is a cheap 8 year old and hisses at max volume like a running water tap would sound from down the hall. Some other receivers have lower noise floor and in a silent room with the volume maxed, you'd hear the track more clearly. But, you would never run your receiver at max!

 What Arve has done here in the post is take this barely audible file and put it into and under another track that is at a normal level. The only way you can hear the quiet sound is to play the regular-volume song to ear-destroying levels.

 Basically what this test shows is that in order to have not only the 8 extra bits that 24-bit gives you audible, but even the least significant bit of the 16 bit file audible, I have to have my stereo so loud that my ears would literally burst, if the speakers wouldn't go first.

*If our ears and speakers could handle it, you couldn't make out the barely audible least significant 9 bits through the hurricaine of noise anyway, due to multiple mechanisms:*


Your brain very likely wouldn't be sensitive to it with all the much louder sound present. But if it was;
There would be mechanical noise generated on the eardrum from the extremely high SPL that would be louder than the sound present in the 9 least significant bits, completely overpowering it. But, even if we had perfect eardrums;
The ear has a mechanism that engages with loud sounds where it adjusts the bones to dampen incoming sound, possibly to the point where we then wouldn't hear the quietest fraction even if was present on it's own without all the other loud sound.

 There is no question that unless you have a special application that requires enormous dynamic range, there is no hearing the extra quiet details granted in a 24 bit track. An example application where it would be necessary, would be to have the ability to encode explosions or flashbang grenades at _lifelike_ levels, along with quality quiet whispers, in the same track. For example, you could use the least significant 16 bits of the 24 bit stream to encode a regular movie soundtrack, and you use the loudest 8 bits as headroom to be able to actually literally blow the audience's ears out, or to encode bass so loud so as to represent an earthquake or the earth's atmosphere splitting in half.

 When is 24-bit necessary?

 The use of 24-bit by engineers or musicians for recording and working with audio is explained elsewhere but here are other applications for why we may want 24 or higher:

 Recordings of underwater or space phenomena, for example, where there can actually be sounds thousands or millions of times quieter than other sounds, and we want all of them to be in high quality. With fewer bits you either cut out the loudest stuff, or cut out the quietest stuff. If you want it all, you need 32, 64, 128 bits or more. Your microphone or measuring instrument and electronics better be that accurate though, limiting the application to the highest of scientific instruments/research.

 This also opens the door to new technology. Assuming the microphones used were sensitive enough, the 24-bit file can contain sounds so quiet, such as the musician's breathing, their heart beating, the engineer turning a knob in the other room, or things in the room rattling due to the instruments. In the future with new processing algorithms, we may be able to extract these details to create more realism in the experience, or figure out more information about recordings.

 If we develop a brain-audio interface to send the audio data directly into our neuropsychology, our brain can then potentially perceive the data to any level of detail, and with any amount of dynamic range with no potential for hearing damage. It is closer to "awareness" data about every frequency or sound and its level sent into our perception. In this case, the higher quality, the better, whether it's bit depth or sample rate. More digital data is directly proportional to more thought information. So, obtaining/storing files at 24 bit may be useful later even if we can't hear it now, but by then our current storage and delivery methods may be obsolete anyway.


----------



## The Walrus

Hey guys,
 Been reading this thread for a few days. I'm not a recording expert or a sound engineer to make any comment, but from what I read and hear, is it safe to claim that the labels are selling snake oil, especially with the remastered 24 bit versions of ancient albums like "Led Zeppelin IV" ?
  
 I wonder who makes the most profit from the 24 Bit  audio business. Is it the record labels or the DAP's with a big "Hi Res" sticker plastered on the box 
  
 cheers


----------



## Head Injury

the walrus said:


> Hey guys,
> Been reading this thread for a few days. I'm not a recording expert or a sound engineer to make any comment, but from what I read and hear, is it safe to claim that the labels are selling snake oil, especially with the remastered 24 bit versions of ancient albums like "Led Zeppelin IV" ?
> 
> I wonder who makes the most profit from the 24 Bit  audio business. Is it the record labels or the DAP's with a big "Hi Res" sticker plastered on the box
> ...


 

 24-bit copies of the same mastering? Yes, that might as well be snake oil.
  
 If it's a brand new master, there may be actual appreciable differences between the new 24-bit masters and old remasters. But, it's not because it's 24-bit, and the differences aren't necessarily for the best.


----------



## Greenears

kaffeemann said:


> Back to topic. Benchmark Media Systems published this article on the bit depth stuff:


 
  
 I thought that article had some correct stuff and some incorrect.


----------



## The Walrus

Hi guys,
 I have a question: How can I convert 24 bit flac files to 16/44.1 flac without introducing any artifacts in the conversion process? Sony's Media Go software can convert 24 to 16. Can I trust its accuracy?
 Thanks.


----------



## RRod

the walrus said:


> Hi guys,
> I have a question: How can I convert 24 bit flac files to 16/44.1 flac without introducing any artifacts in the conversion process? Sony's Media Go software can convert 24 to 16. Can I trust its accuracy?
> Thanks.


 
  
 I use Sox, and the incantation is:
  
 sox -G yourfile.flac -b 16 newfile.flac rate 44100 dither -s
  
 But you can also just try the Sony and see how it sounds, or do an ABX by upsampling the 16/44.1 file back up.


----------



## The Walrus

rrod said:


> I use Sox, and the incantation is:
> 
> sox -G yourfile.flac -b 16 newfile.flac rate 44100 dither -s
> 
> But you can also just try the Sony and see how it sounds, or do an ABX by upsampling the 16/44.1 file back up.


 
 Thanks. Just did the conversion. It sounds good. Will do an AB test later.
 cheers.


----------



## castleofargh

the walrus said:


> Hi guys,
> I have a question: How can I convert 24 bit flac files to 16/44.1 flac without introducing any artifacts in the conversion process? Sony's Media Go software can convert 24 to 16. Can I trust its accuracy?
> Thanks.


 

 media go is really window media player with another skin(in fact you can't use media go if you removed WMP from your computer).
 anyway I didn't find anything audible when I tried the software(except that the interface sux ^_^). but I agree that ultimately you're doing it for yourself, so you should just ABx the original file and the converted one for your own peace of mind. it's always conforting to fail an ABX instead of just put your trust into some random internet guys 
	

	
	
		
		

		
			





.
  
 now if you just want the probable best even though you can't hear it, SOX has a very nice reputation. probably the preferred solution for free.


----------



## The Walrus

OK,this is weird... I bought the 24 bit version (for some reason) of Lara Fabian's "Le Secret" from qobuz.com  I noticed very significant distortion in Lara's vocals especially in high notes. After scratching my head for a week, I downloaded the 16/44.1 flac version from the same site, and it was way cleaner (don't ask me to describe it in technical terms) and no distortion whatsoever.
 This is not a placebo effect (at least I don't think so). How can you explain this? How can 24 bit version sound worse? Could it be that they used different masters?
 (BTW, the 24 bit was 24/44.1)


----------



## RRod

the walrus said:


> OK,this is weird... I bought the 24 bit version (for some reason) of Lara Fabian's "Le Secret" from qobuz.com  I noticed very significant distortion in Lara's vocals especially in high notes. After scratching my head for a week, I downloaded the 16/44.1 flac version from the same site, and it was way cleaner (don't ask me to describe it in technical terms) and no distortion whatsoever.
> This is not a placebo effect (at least I don't think so). How can you explain this? How can 24 bit version sound worse? Could it be that they used different masters?
> (BTW, the 24 bit was 24/44.1)


 
  
 Are you sure there's no on-the-fly down-conversion going on?


----------



## The Walrus

Actually I'm not sure. I listened to both with Media Go. Could it be


rrod said:


> Are you sure there's no on-the-fly down-conversion going on?


 
 Actually I'm not sure. I listened to both with Media Go. Could it be that the software is down sampling it before playing? What player should I use for PC to listen to 24 bit flac?


----------



## RRod

the walrus said:


> Actually I'm not sure. I listened to both with Media Go. Could it be
> Actually I'm not sure. I listened to both with Media Go. Could it be that the software is down sampling it before playing? What player should I use for PC to listen to 24 bit flac?


 
  
 If it's trying to on-the-fly go from 24-bit to 16-bit, it's probably just tossing off the last few bits from the file (truncating), which shouldn't be too taxing on the system. But you never know. I'm a Linux user, so someone will have to help you get Windows right (or search the forums).
  
 p.s. You an also try converting the 24bit file beforehand to 16bit and see if the distortions remain.


----------



## The Walrus

rrod said:


> If it's trying to on-the-fly go from 24-bit to 16-bit, it's probably just tossing off the last few bits from the file (truncating), which shouldn't be too taxing on the system. But you never know. I'm a Linux user, so someone will have to help you get Windows right (or search the forums).
> 
> p.s. You an also try converting the 24bit file beforehand to 16bit and see if the distortions remain.


 
 Thanks.


----------



## castleofargh

the walrus said:


> Actually I'm not sure. I listened to both with Media Go. Could it be
> 
> 
> rrod said:
> ...


 

 media go has an output option for ASIO for bit perfect, but you need to have asio or at least some kind of asio for all or a name like that(I'm a wasapi fanboy for totally non audio related reasons so I can't really tell).
 else a good start could be to check that your windows options for sound aren't 16/44.
  
 but as RRod was saying, I doubt that adding or removing the last bits would change anything to the music. it's a pretty innocent process involving zero calculation. maybe media go is the culprit, I never tried listening to music with it. I use foobar, other often recommend Jriver. other seem to like musicbee, but I would think that it has to do with the file managing options more than any special playback specialty.


----------



## stv014

the walrus said:


> How can 24 bit version sound worse? Could it be that they used different masters?


 
  
 It is not impossible, sometimes if the 24-bit version is released later, it can be louder and more compressed and distorted than it was on an older CD. If this is the case, the tracks may be visibly different in an audio editor.


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## Greenears

Alright everyone .... drumroll ... I've finally done some ABX testing.  It was relatively easy you can try the same thing yourselves.
  
 I downloaded HDtracks 2014 sampler (free), and played track 2, Vivaldi's Spring on headphones.
  
 I used SoX to convert it to straight 16 (no dither), I did not convert the sample rate so it was 96/24 vs 96/16.
  
 Ran Foobar2000 ABX plugin on 30 trials and got 19/30 probability of guessing 10%.
  
 I thought I was getting better and maybe with more trials I could get it lower, but decided to quit while I was ahead. I started well and then had a rough patch in the middle, but later on had a good streak.  After 10-15 trials I honed the technique to just focus on a short section from 1:11-1:21 (use the blue slider to narrow).  As the violin crescendos you listen for a certain "roughness" for the B (16-bit).  Also the strings sound more "beautiful" but maybe a bit flatter in A (24 bit).  Flat sounding is characteristic of more even harmonics, some may say "pure" or "warm" or "subdued"  There is a little distortion in both A and B at the crescendo peak but it's just a little less grating in A.  I also tried listening for "air" and "bass" earlier on but had more luck with the second even though I was sure 24 bit had more "air" at the start.
  
 After a while I just set the volume on medium (reasonably loud for that passage) and didn't change it didn't even bother re-listening to A and B just jump between X and Y a few times and as soon as I heard "it" clearly I made the decision.  Maybe 3 or 4 back and forths each decision.  Every 5th pair or so I just rechecked A and B to remind myself.
  
 Note I am quite a skeptic you can hear anything more than 16/44 (read my posts).  This is my first ever ABX test. I should say that when you first hear the 24 bit you are quite impressed. I think it's a very good recording, and it reveals some technical brilliance of the players.  It's quite startling in realism.  However .... when you listen in 16 bit you realize it is not so much the bits as the recording and what feels like wider dynamic range than typical.
  
 What did I prove? Hmmm. Not sure.  10% seems good but I was closer to 40% a lot of the time.  I haven't used dither yet.  I was quite impressed at my first 24 bit samples at first, but the more trials you fail you start to realize how close they are and many things you believe are "better" are actually identical. Still listening to the whole thing while I type this I could swear the 24 bit had a little more "emotional grab" or "immediacy" than the 16 bit. Or maybe I just want to hear that? hmmm.


----------



## stv014

greenears said:


> I thought I was getting better and maybe with more trials I could get it lower, but decided to quit while I was ahead.


 
  
 Note that doing many trials and looking for a run that happens to be below 5% (or whatever threshold is chosen) is a statistically biased method. X% chance of guessing means just that, and with enough trying it can eventually be achieved by luck. Do you still have the scores for all the trials you have ever done for that track ? Over a large number of trials with no selective inclusion of results, real audibility should produce a combined score that converges towards zero chance of guessing (for example, 69/100 is already less than 0.01% chance).


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## castleofargh

Greenears, it's ok to show the result when you're doing it as a way to try and learn how to succeed. so you know when you successfully identified a difference and can practice finding it again. I did that a lot to learn identify mp3.
 but when you're doing the real ABX, you should mask the results. because that's a bias like any other and you shouldn't have it in the test.
  
 often when I look for something and let the answer visible, I will see that I randomly got right the first 3 or 4times(not that hard to get statistically) and that in itself will bias me into thinking I heard something. and I start making up cues in my head that sometimes were never real. it's just my playful brain that decides to give me what I ask for, without any regard to reality and what I'm really hearing.
  
  


greenears said:


> .... when you listen in 16 bit you realize it is not so much the bits as the recording and what feels like wider dynamic range than typical.
> ....


 
 changing the bit depth will not change anything in the dynamic of the recorded music.there are very little chances that the 24bit record actually uses more than 70db of dynamic.
 and anyway each sample will be at the exact same loudness on both encodings. you must imagine it as chopping down the lowest sounds of the record, removing sounds from -96db to -144db not as changing the dynamic. from 0 to -96db the sound is really 100% identical. so what you describe as wider dynamic range is 100% bias. you went in your head from "changing the maximum possible dynamic range of the medium" to "changing the dynamic of the music". ^_^ 
 I guess it's the most common misconception one can do in audio, so there really is no shame to it, we all went that road. actually some dudes with 30years of very active audiophile life are still going at it with that very misunderstanding. just like the max power of an amp being bigger never meant the amp would actually deliver more into a given headphone.


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## Greenears

There is nothing wrong with looking at the results as you go.  If you can discern the difference, then it is audible.   if there is really truly is no difference then no help you get is going to matter.  I didn't practice either just went straight in.
  
 I don't know if foobar automatically logs the results, but for those that want the exact sequence I'd be happy to post it if you tell me where to look.  I'll be doing some more tests anyway tomorrow.  Generally I remember i started with 2/3 and then my % chance of guessing hovered in the 30-50% range. I don't think i went over 50 much if at all.  Then it dropped to 10%.  Yes 19/30 is hardly convincing I think I'm pretty clear about that in my post. I can do it again tomorrow when well rested and have time.  Honing in one one passage definitely helps and as I used the blue bars my score got better.
  
 Regarding dynamic range, to be clear my comment was with respect to the recording, not the bit depth.  What I am saying is I listened to several tracks and chose the Vivaldi because I thought it had a wider dynamic range and more detail of loud and soft passages than most recordings.  It was also recorded with a fair bit of echo and a "live" room/soundstage feel.  I was hoping those soft echos and slight distortions would give me something to latch onto. 
  
 16 and 24 bit has nothing to do with the recorded dynamic range.
  
 For those that claim this track does not have the range, please measure it.


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## castleofargh

ok I misunderstood your sentence about dynamic.
  
 about looking at the result while doing the test. the only condition where it isn't a bias is if you determine how many trials you will do before starting and stand by it. I find that not knowing makes it more honest between me and myself(also I can just stop when I'm bored ^_^).
  
 about the track having huge dynamic, as long as it has less than 96db(and I would be very surprised if it didn't, the maximum I found in my library was somewhere around 65/70db.), then 16bit is enough by definition.


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## RRod

greenears said:


> There is nothing wrong with looking at the results as you go.  If you can discern the difference, then it is audible.   if there is really truly is no difference then no help you get is going to matter.  I didn't practice either just went straight in.
> 
> I don't know if foobar automatically logs the results, but for those that want the exact sequence I'd be happy to post it if you tell me where to look.  I'll be doing some more tests anyway tomorrow.  Generally I remember i started with 2/3 and then my % chance of guessing hovered in the 30-50% range. I don't think i went over 50 much if at all.  Then it dropped to 10%.  Yes 19/30 is hardly convincing I think I'm pretty clear about that in my post. I can do it again tomorrow when well rested and have time.  Honing in one one passage definitely helps and as I used the blue bars my score got better.
> 
> ...


 
  
 Can't download it because HDTracks requires a downloaded that doesn't work in either Linux or Wine. But I've heard enough Vivaldi to know that it doesn't have the dynamic range of Mahler, and Mahler works on CD just fine. The proper way to do the test is to pick a set number of trials, and do all the trials without looking at results. Also, here's what you should be doing in Sox, just in case:
 sox -V4 24bit.flac -b 16 24to16bit.flac dither -s
 sox -V4 24to16bit.flac -b 24 16to24.flac
  
 Then you compare 24bit.flac to 16to24.flac, with your sound card / dac set to 24/96.


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## bigshot

If you look at your results as you go, you aren't doing a blind test.
  
 I don't think that there is a piece of recorded music with a dynamic range that gets close to needing 24 bit, so it really doesn't matter what music you use.


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## cjl

bigshot said:


> If you look at your results as you go, you aren't doing a blind test.


 
 That isn't quite true. It's still a blind test because while you're trying to decide whether X is A or B, you do not have any knowledge (or way of knowing) which one it is, aside from the sound quality. That having been said, it is true that looking at the results while you're testing and then quitting "while you're ahead" does bias the test statistically. The more valid and rigorous method would be to decide the number of trials in advance, and do that number of trials regardless of intermediate outcomes.


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## bigshot

If you consider them a bunch of different tests it may be considered blind. But if you are considering it all part of the same test, it isn't blind.


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## Greenears

Thanks all for interesting responses.  I'd like to get back to the meat of the testing, so let me address them collectively:
  
 Maybe I should back up first and let you know the purpose of me posting as I am going along is to (a) get tips for improvement (b) encourage others to reproduce my results.  Maybe with your ears and/or equipment you can do better.  I will address a few objections as a courtesy, but I don't want to get too sidetracked from (a) and (b).  I would encourage more ideas on whether to listen to short clips or long, what to look for, better tracks to try (as long as normally accessible in US over regular network), etc etc etc.  
  
 1. Number of trials and looking at the results
  
 So I hear your input, and I can see where you are coming from.  Actually I never put any thought into this it was the default setting and off I went.  Plus it was late and I wanted success quicker rather than slower, if it was possible to discern.  However, I have taken a couple of 2nd year courses in probability enough to know the binomial distribution and its application.  It seems the ABX plugin is doing a straight binomial distribution calculation on the success/trials ratio.  It doesn't care what you are thinking or doing or intent or whether you are looking at the result or not.  That's the beauty of ABX testing.  The only thing that would be invalid is to throw out failing trials but the tool won't let you do that .... you can start again from the beginning or continue but that's it.  Note that 3/4 yields a very different % than 30/40 - that is baked into binomial. The % is valid no matter what you do or when you quit.  I promise you.  However, note that 10% means exactly that - 1 in 10 chance I was flipping coins and using that to decide. 
  
 To humor everyone if I get what I consider solid results (5% or better) I'll redo it both ways and post the logs.  See how nice I am?  But I did want to set the record straight for our dear readers.
  
 2. Dynamic range, and whether this Vivaldi is a good track to get a positive result
  
 Firstly, people often confuse available dynamic range (eg. 96 dB for redbook CD) vs actual dynamic range of a section of music (ratio of loudest to softest part).  I was trying to select a piece of music with a high value for the second. I didn't measure it (someone provide me a SoX incantation and I'll gladly do it).  Whether Spring can have a dynamic range depends on the music to an extent, yes I agree, but also how it was recorded.  With a sensitive mic really close to a violin and if the player plays very softly and very loud, you could get a range.  The rest is the mix and how much dynamic range compression is applied.  My expectation is that at the most this is 50 dB, more likely less  Compared to 15 db for a lot of popular music I'm told.  I heard one engineer claim 60 dB on a big orchestra, that is the highest claim I've ever read.  The reason I want high range is to look for softer passages, were the relative delta of each step of quantization is highest.  This is where 24 bit might shine.
  
 3. Difficulty downloading to reproduce
  
 I'm really really sorry HDtracks doesn't support Linux.  But if you find another track we can both download legitimately I'm game.  I was suggested to use HD Tracks 24 bit Random Access Memories but I thought Vivaldi was a better start point.  I can't get into Pono store yet, the European place only starts in January.  Where else?
  
 4. Dither and reconverting to 24 bit
  
 I intentionally started without dither.  I want the best chance for success first, then I'll add dither and see if it makes me fail.
  
 >sox -V4 24to16bit.flac -b 24 16to24.flac  -- interesting suggestion, but I don't think think upconverting is legitimate is it?  The dither will be on the 8th LSB and not 0 LSB when you get back to 24.  The DAC natively handles 16 and 24 and converts to multi-segment Sig-Delt anyway so I think my method is legit. This incantation was suggested after some discussion.  
  
 One word about success: I am an admitted skeptic about 24 bit.  My intellectual bias says that 16 bit redbook may be the be-all and end-all of music, all we need is better reproduction hardware.  But, I'm not 100% sure of my position.  So why do I want to succeed in telling them apart?  The weakness I find in ABX testing is when someone does a whole series of comparison and they all come up negative.  It is easily attacked, and hard to prove the negative (ie if you had just done X you would have passed).  But if you pass one and gradually change parameters till you fail, it shows where the knee in the curve is for your ears and setup. IMHO.


----------



## sonitus mirus

With respect to the number of trials and immediately seeing the results of each, the fallacy of the maturity of chances might create bias.


----------



## Roly1650

greenears said:


> Thanks all for interesting responses.  I'd like to get back to the meat of the testing, so let me address them collectively:
> 
> Maybe I should back up first and let you know the purpose of me posting as I am going along is to (a) get tips for improvement (b) encourage others to reproduce my results.  Maybe with your ears and/or equipment you can do better.  I will address a few objections as a courtesy, but I don't want to get too sidetracked from (a) and (b).  I would encourage more ideas on whether to listen to short clips or long, what to look for, better tracks to try (as long as normally accessible in US over regular network), etc etc etc.
> 
> ...



And of course you've checked that the HD Tracks file is genuine 24/96 and not upsampled 16/44 right? Being upsampled wouldn't be a first for that outfit.


----------



## RRod

greenears said:


> Thanks all for interesting responses.  I'd like to get back to the meat of the testing, so let me address them collectively:
> 
> Maybe I should back up first and let you know the purpose of me posting as I am going along is to (a) get tips for improvement (b) encourage others to reproduce my results.  Maybe with your ears and/or equipment you can do better.  I will address a few objections as a courtesy, but I don't want to get too sidetracked from (a) and (b).  I would encourage more ideas on whether to listen to short clips or long, what to look for, better tracks to try (as long as normally accessible in US over regular network), etc etc etc.
> 
> ...


 
  
 Resposes above.


----------



## limpidglitch

greenears said:


> I'm really really sorry HDtracks doesn't support Linux.  But if you find another track we can both download legitimately I'm game.  I was suggested to use HD Tracks 24 bit Random Access Memories but I thought Vivaldi was a better start point.  I can't get into Pono store yet, the European place only starts in January.  Where else?


 
  
 People have been using free download samples from 2L in the past. Haven't listened to any of them myself, so I can't comment on recording and musical qualities.


----------



## Greenears

I understand the padding with zeros. It is just weird with dither. I'm going to first try get to 5% then I'll try to reverse that by converting to 24,then dither 16 then dither 24. 

I'm not sure how familiar you are with the details of the foobar abxy plugin current version? I used the default setting. 

Forgot about linn. Good idea. 

I think we agree to try high dr tracks listening to quiet section. Best chance of many leading zero where 24 shines. Anyway to measure that with sox?


----------



## castleofargh

greenears said:


> Thanks all for interesting responses.  I'd like to get back to the meat of the testing, so let me address them collectively:
> 
> Maybe I should back up first and let you know the purpose of me posting as I am going along is to (a) get tips for improvement (b) encourage others to reproduce my results.  Maybe with your ears and/or equipment you can do better.  I will address a few objections as a courtesy, but I don't want to get too sidetracked from (a) and (b).  I would encourage more ideas on whether to listen to short clips or long, what to look for, better tracks to try (as long as normally accessible in US over regular network), etc etc etc.
> 
> ...


 

 1/ if you do more trials, you just have to add the previous results to them. that way you get rapidly a much better statistical relevance and avoid the problems of seeing the results live, or simply picking the one test that looked the best out of several. it's not a problem for you and it's very much reassuring for us ^_^.
  
 2/


> My expectation is that at the most this is 50 dB, more likely less  Compared to 15 db for a lot of popular music I'm told.  I heard one engineer claim 60 dB on a big orchestra, that is the highest claim I've ever read.  The reason I want high range is to look for softer passages, were the relative delta of each step of quantization is highest.  This is where 24 bit might shine.


 
 agreed with about 60db, the max we encounter on albums is in that ballpark.
 about the quantization delta, maybe I don't get what you're saying again, but to me it's nope. with the lowest signal let's say at -70db(if the audio wasn't topped at 0db but at -10instead), even then you would still have the quantization noise much lower, so no masking, no nothing from 16bit IMO.
 and of course the values of each samples are strictly the same with the same precision in 16 and 24bit, only the added zeros unused, and the overall noise floor will change on 24bit.
  
 3/ the track that we pick doesn't mean a lot. obviously for the purpose of the test it would seem logical to get the most dynamic piece we can, but as we all agree, the most dynamic track is still far from reaching 16bit limits. the track dynamic will not change, no part of the music will be truncated of compressed, so all in all if you want to abx 16 and 24bit of a metallica track, it would give pretty much the same results I think. because if there is an audible difference between the 2 bit depth, the tracks dynamic won't really have much to do with it.
 at least that's how I see it.
  
 4/dither or no dither, well it doesn't really matter much. just know that almost any 16bit album is dithered, so it's not such an unfair game to dither you track.  my own trials have shown that unless I push the volume to the point where I can hear the noise(not safe at all!), I can't seem to distinguish dither or no dither, or even different dithering methods.
  
 and of course converting back to 24bit is just a harmless way to make sure you don't get external hints. on a computer I wouldn't think so, and with foobar abx I believe both files are turned into 32bit wave or something before being played.  so it doesn't matter.


----------



## stv014

> Originally Posted by *Greenears* /img/forum/go_quote.gif
> 
> 1. Number of trials and looking at the results
> 
> ...


 
  
 Restarting the test when the result starts to look bad (discarding the previous trials as if they never happened) is definitely a "cheating" method, as is doing multiple runs and choosing the best score while ignoring the rest. In general, deciding whether a trial is to be included in the "final" statistics *after* knowing if it was a correct guess introduces bias. That is why I suggest computing an overall score from all trials ever done, except those that were chosen to be excluded in advance before doing them (e.g. for "training" purposes). Looking at the score after every trial, and stopping when it is "good enough" is also a problem, particularly when only looking for a relatively high p-value like the usual 0.05. In any case, with biased methods, the real chance of guessing is higher, for example one 5% best result cherry picked from 10 runs can actually be achieved with ~40% chance just randomly guessing. The various biases can be compensated by requiring a much lower p-value, like 0.0001 (0.01% chance of guessing).


----------



## RRod

greenears said:


> I understand the padding with zeros. It is just weird with dither. I'm going to first try get to 5% then I'll try to reverse that by converting to 24,then dither 16 then dither 24.
> 
> I'm not sure how familiar you are with the details of the foobar abxy plugin current version? I used the default setting.
> 
> ...


 
  
 I'm on Linux, so I can't advise on foobar. One other thing to check for is that you're not getting messages about clipping from sox (don't know if this happens with non-dithered bit-depth conversions). My first step is typically to normalize the volume of the 24-bit track to -3 or -6dB, which is legit because even the best audio equipment doesn't actually get 24 bits above noise, so 23-23.5bits isn't losing anything. But this gives you headroom for the calculations sox has to do. Also, since there is 0% chance that any real music track is taking up all 24bits, normalization brings up the soft parts exactly so they aren't lost in any bit truncation.
  
 The incantation:
 sox 24bitfile.flac normed.flac gain -n -3


----------



## vid

rrod said:


> I'm on Linux, so I can't advise on foobar.


 
  
 Works fine on Linux through Wine, for the record.


----------



## RRod

vid said:


> Works fine on Linux through Wine, for the record.


 
  
 No doubt, but I save Wine for things that have no legit equivalent on Linux.


----------



## Greenears

castleofargh said:


> 1/ if you do more trials, you just have to add the previous results to them. that way you get rapidly a much better statistical relevance and avoid the problems of seeing the results live, or simply picking the one test that looked the best out of several. it's not a problem for you and it's very much reassuring for us ^_^.
> 
> 2/
> agreed with about 60db, the max we encounter on albums is in that ballpark.
> ...


 
 This maybe difficult for those without background in signal theory to understand, I'll try a different way.  Think of quantization as levels, or rungs on a ladder.  Let's say there are 100 equal space levels to use round numbers.  If the loudest peak is at level 90, the lowest peak at level 40, and the combined noise floor (recording noise plus quantization) is at 10,  then there are 80 levels between the loud parts and floor, but only 30 levels between the soft parts and the floor.  You can see that the level gaps are twice as big relative to the amplitude of the soft part compared to the loud part.   
  
 Yes I've read most CD are mastered with peaks at -10 to -15 dBFS.   So let's say this one peaks at -10 dBFS and with 50dB DR then the bottoms are at -60dBFS and let's say the noise is at -90.  You only have 30 dB SINAD for the soft parts compared to loud parts 80 dB.  It's exactly the same thing I just said before, expressed in dB.  Since I expect 80 dB is not audible, I focus on the 30. 
  
 Note though I started just trying to get a "Feel" for the opening loud passage.  My score improved when I focused on a narrow medium soft passage.  That is why I think there is something to it.


----------



## Greenears

FWIW I did another run.  Same files same procedure, nothing touched since the other day.  Still no dither, no upconvert (I'll get to that later if I can get to 5% or better). I got the first trial wrong, but then had a streak of 4 in a row to get me to 5/6 and about 10% or so.  Then I kept going and ended up with 5/10 62.3%.  Interestingly I felt I was starting to fatigue into the 2nd half and I was also distracted maybe somewhere around trial 6 I had to get up for a moment.
  
 Other than that I just focused on that one passage from 1:10-1:21.1 at volume 48.  I listened to A & B for a few minutes to get warmed up, but then only XY for the first 5 trials .... I checked back with A & B a little at maybe trial 7 but it didn't help.   I also felt less certain about my choices later.
  
 I'm going to try again a longer run when I'm better rested and no distractions.  I'm prettty confident in what I'm listening for, but you play this passage 5 or so times each trial, so after play 50 your brain doesn't want to focus.   Another tactic may be to do 5 trials, take a break and come back for another 5 (on the same test panel) and so on.
  
 If someone has a passage were it is way easier to pick it out ... I'm all ears please suggest.  I would prefer a short passage, 10 seconds or less.
  
 So far I have never seen less than 10%, not even for a fleeting moment.


----------



## RRod

greenears said:


> FWIW I did another run.  Same files same procedure, nothing touched since the other day.  Still no dither, no upconvert (I'll get to that later if I can get to 5% or better). I got the first trial wrong, but then had a streak of 4 in a row to get me to 5/6 and about 10% or so.  Then I kept going and ended up with 5/10 62.3%.  Interestingly I felt I was starting to fatigue into the 2nd half and I was also distracted maybe somewhere around trial 6 I had to get up for a moment.
> 
> Other than that I just focused on that one passage from 1:10-1:21.1 at volume 48.  I listened to A & B for a few minutes to get warmed up, but then only XY for the first 5 trials .... I checked back with A & B a little at maybe trial 7 but it didn't help.   I also felt less certain about my choices later.
> 
> ...


 
  
 See my comment on normalization above, as that will affect results across the board. You can save interim results to a file, then compile all the results at the end if you want to spread your testing out over several days to avoid fatigue. Just ignore the file results and hold yourself to the preset number of trials.


----------



## castleofargh

greenears said:


> castleofargh said:
> 
> 
> > 1/ if you do more trials, you just have to add the previous results to them. that way you get rapidly a much better statistical relevance and avoid the problems of seeing the results live, or simply picking the one test that looked the best out of several. it's not a problem for you and it's very much reassuring for us ^_^.
> ...


 

 ok so I understood your previous post right and just don't get why that would make any difference. unless you're meaning to say that you're hearing the noise floor?
  
 also it's kind of funny that you're thinking I can't follow digital theory when I started writting to you because I thought you didn't understand the basics ^_^.


----------



## The Walrus

roly1650 said:


> And of course you've checked that the HD Tracks file is genuine 24/96 and not upsampled 16/44 right? Being upsampled wouldn't be a first for that outfit.


 
 This is the response to my query, by the genius at HDTracks tech support: 
  
 "I'm going to try and clear things up for you. To understand the bit depth (also referred to as the bit rate) of an audio file, you need to understand sample rates and the process by which audio is captured digitally using modern recording techniques. With Digital Audio Workstations (DAW) audio is captured at a certain sample rate, usually being 44.1k, 48k, 88k, 96k, 176k, or 192k. This represents the amount of "samples," or digital snapshots, taken *per second *of the analog audio source. So obviously 192,000 samples per second will be much greater audio quality than 44,100 samples per second because more audio data is being recorded. 
  
 Now onto bit depth. Bit depth is the *quality of each individual sample*. This means that a 24 bit sample is greater quality than a 16 bit sample. You can liken this to megapixels on a camera. The more megapixels your camera has, the greater the image quality.
  
 If you would like to learn more about sample rates and bit depth, there are countless articles online. Please note that not everybody can hear the difference between hi-res audio and a regular 44/16 physical CD, which is why we recommend that everyone try our hi-res free sampler to find out.
  
 I hope this information is helpful."


 Go to their website and check the FAQ page. There they say that the tracks that they sell is CD QUALITY. Nowhere on their website you can find a single word about the benefits of HD audio. Isn't it strange that these people aren't promoting the very product they are selling?


----------



## castleofargh

the walrus said:


> roly1650 said:
> 
> 
> > And of course you've checked that the HD Tracks file is genuine 24/96 and not upsampled 16/44 right? Being upsampled wouldn't be a first for that outfit.
> ...


 

 what's nice is "To understand the bit depth (also referred to as the bit rate) of an audio file"
	

	
	
		
		

		
			




 yeah bit rate, expressed in bit per second is obviously the same as bit depth... it's probably just an attention mistake from the guy, but that certainly doesn't look pro.
 I would bet that most hirez providers have started to become expert in the art of not clearly saying that hirez is always hirez quality. because else they get in trouble each time the track is actually 16/44 turned into 24/96. some put on a warning like "we take the files as given by the studio, so don't sue us, we're innocent and we don't have the money to buy a free copy of audacity and check ourselves", others just avoid the question altogether.


----------



## Greenears

rrod said:


> See my comment on normalization above, as that will affect results across the board. You can save interim results to a file, then compile all the results at the end if you want to spread your testing out over several days to avoid fatigue. Just ignore the file results and hold yourself to the preset number of trials.


 
 So I mucked around for 10 minutes trying to find out where the log files are, looking online, searching my drives.  Finally I figured it out by accident, it offers you a log when you exit.  But it wasn't obvious that was the log.  Next time I'll post a log.  I'll have to figure out how to join logged sessions.
  
 I hear you on normalization, and I guess if I keep failing I'll try it later. The problem is once you manipulate the file in any way beyond chopping the 8 bits from 24 to 16 I think you open yourself to criticism (beyond leveling the volume between the tracks which is of course required).  By turning up the volume on a soft section I'm sort of doing the same thing.


----------



## Greenears

castleofargh said:


> ok so I understood your previous post right and just don't get why that would make any difference. unless you're meaning to say that you're hearing the noise floor?
> 
> also it's kind of funny that you're thinking I can't follow digital theory when I started writting to you because I thought you didn't understand the basics ^_^.


 
 I don't understand why you don't understand!  What is the air-speed velocity of an unladen swallow!  Argh.
  
 If you take my ladder analogy to the extreme, you can make the signal so small that there are only two rungs.  At that point you won't hear the signal.  So yes obviously amplitude matters and makes a difference.  The word "noise" means many things since there are many different types.  Quantization noise could be called quantization artifacts or whatever you want.  I am simply saying on soft passages I think you have a better chance of hearing Q artifacts.  However, the q noise may be below the background noise of the recording room, in which case you can't hear it whatever you do and I'll fail the ABX testing.  Unless there are other artifacts at bigger amplitudes, such as subtle phase shifts (frequency shifts) that you can hear.  I don't know.


----------



## bigshot

You're a lot more likely to reach the noise floor of the recording venue or your amp by turning up the volume, than you are to reach the noise floor of a CD.


----------



## RRod

greenears said:


> So I mucked around for 10 minutes trying to find out where the log files are, looking online, searching my drives.  Finally I figured it out by accident, it offers you a log when you exit.  But it wasn't obvious that was the log.  Next time I'll post a log.  I'll have to figure out how to join logged sessions.
> 
> I hear you on normalization, and I guess if I keep failing I'll try it later. The problem is once you manipulate the file in any way beyond chopping the 8 bits from 24 to 16 I think you open yourself to criticism (beyond leveling the volume between the tracks which is of course required).  By turning up the volume on a soft section I'm sort of doing the same thing.


 
  
 I wouldn't call it a manipulation, it's simply the business of mapping the 24bits into 16bits. If you can agree with me that the musical content of the track doesn't actually use all 24bits, then there are 2 extreme cases:
 1) The recording is set to have a peak near 0dB. In this case, the softest musical material will be well above the noise floor and near 16bits anyway. Here is where truncation doesn't hurt much, as we're not throwing away that much musical content in the lower sig figs.
  
 2) The recording is set to have the softest musical content just above the noise floor. In this case, the peaks won't get anywhere near 0dB, but since we have musical content near the floor, truncation will throw away lots of good stuff. This is where normalization comes in; you raise up the peaks to near 0dB to get the music away from the floor.
  
 Normalization doesn't just turn up the soft parts, it turns up all track to get the peaks to a certain level. So it's a straight up proportional change, no tricks.
  
 p.s. It's quite true that there comes a signal level where we only have 1 bit with which to encode it. But the thing is that you can still recover dynamics from this via dithering and noise shaping, which is exactly why DSD can work at all.


----------



## Greenears

rrod said:


> I wouldn't call it a manipulation, it's simply the business of mapping the 24bits into 16bits. If you can agree with me that the musical content of the track doesn't actually use all 24bits, then there are 2 extreme cases:
> 1) The recording is set to have a peak near 0dB. In this case, the softest musical material will be well above the noise floor and near 16bits anyway. Here is where truncation doesn't hurt much, as we're not throwing away that much musical content in the lower sig figs.
> 
> 2) The recording is set to have the softest musical content just above the noise floor. In this case, the peaks won't get anywhere near 0dB, but since we have musical content near the floor, truncation will throw away lots of good stuff. This is where normalization comes in; you raise up the peaks to near 0dB to get the music away from the floor.
> ...


 
 Hrmmmphhh mmm arrr ergh. ... which is technical jargon for "I'm still mulling it over".  I understood what you meant right at the word "normalization" but it's still a subtle point.  The problem is the gain also amplifies the noise.  The ratio (dB) stays the same .... I'm actually no longer sure if it will help me pass or fail or make no difference.  To make the math simple, just think about a gain example as multiplying by power of 2 (shift left, pad with trailing zero, ~6dB).  When you truncate to 16 bits you only lose 7 bits of information vs 8 previously due to the trailing zero.  I know what you're saying but I think the volume control does the same thing.  Any noise wiggling around on the 7th bit will still be there in the same ratio.  
  
 Do yo really think doing this will help me pass? I want to pass first.  After I get to 5% I'll dial it back and try more options. So far the best instantaneous number I've seen is 10%.  Today I mucked with the Linn tracks in poor listening conditions, my results were waay worse clearly guessing land.  Need to try more from HD tracks but so far that one passage from Spring is my best candidate since I've had 5+ streaks more than once. 
  
 It just occurs to me there is a guy I know that I can ask some of these DSP questions.  He happens to be an awesome jazz musician (keyboards) and a wizard with FFTs.  Next week.


----------



## limpidglitch

greenears said:


> Do yo really think doing this will help me pass? I want to pass first.  After I get to 5% I'll dial it back and try more options. So far the best instantaneous number I've seen is 10%.  Today I mucked with the Linn tracks in poor listening conditions, my results were waay worse clearly guessing land.  Need to try more from HD tracks but so far that one passage from Spring is my best candidate since I've had 5+ streaks more than once.


 
  
 If you continue doing it like this, you'll have to start applying Bonferroni corrections. But as you hove some basic statistics knowledge, you probably already know of this pitfall.


----------



## tdog

RRod et al. The only decent way to turn 24-bit audio into 16-bit is to dither it properly. For information on this see:
  
 http://xiph.org/video/vid2.shtml
 https://www.youtube.com/watch?v=cIQ9IXSUzuM - same as above if you don't have the codec.
 https://www.youtube.com/watch?v=PydUQrka4pc
  
 You could also try these (in rough order of sophistication):
 http://en.wikipedia.org/wiki/Dither
 http://www.audiocheck.net/audiotests_dithering.php
 http://productionadvice.co.uk/no-stair-steps-in-digital-audio/
http://www.ece.rochester.edu/courses/ECE472/resources/Papers/Lip*s*h*i*tz_1992.pdf - I think I fixed the underlying link, if not remove the *s manually - it's well worth reading.
 http://drewdaniels.com/dither.pdf
  
 Turning 24-bit audio into 16-bit by truncating the last 8 bits is possible but not recommended, you will lose resolution as shown above..
  
 Normalisation is the process of linearly multiplying all samples by the same amount to shift the peak value to some pre-determined (usually high) value. An unnormalised file might peak at -4.5dB, normalising it will bring it up to whatever value you choose, usually something close to 0dB, for example -0.05dB. This does not change the resolution at all, it does change your peak and average levels.
  
 Hope that helps.


----------



## tdog

Oh the STOOPID editor has starred out Stanley Lip-shi-tz name in the link above because it has crap in it - GROAN!
  
 Here is the link http://www.ece.rochester.edu/courses/ECE472/resources/Papers/Lip*s*h*i*tz_1992.pdf - remove the *s manually...


----------



## limpidglitch

tdog said:


> Oh the STOOPID editor has starred out Stanley Lip-shi-tz name in the link above because it has crap in it - GROAN!
> 
> Here is the link http://www.ece.rochester.edu/courses/ECE472/resources/Papers/Lip*s*h*i*tz_1992.pdf - remove the *s manually...


 
  
 Or you can: http://bit.ly/1w4R1ii


----------



## bigshot

The goal is not necessarily to pass. It's to find out the truth.


----------



## RRod

greenears said:


> Hrmmmphhh mmm arrr ergh. ... which is technical jargon for "I'm still mulling it over".  I understood what you meant right at the word "normalization" but it's still a subtle point.  The problem is the gain also amplifies the noise.  The ratio (dB) stays the same .... I'm actually no longer sure if it will help me pass or fail or make no difference.  To make the math simple, just think about a gain example as multiplying by power of 2 (shift left, pad with trailing zero, ~6dB).  When you truncate to 16 bits you only lose 7 bits of information vs 8 previously due to the trailing zero.  I know what you're saying but I think the volume control does the same thing.  Any noise wiggling around on the 7th bit will still be there in the same ratio.
> 
> Do yo really think doing this will help me pass? I want to pass first.  After I get to 5% I'll dial it back and try more options. So far the best instantaneous number I've seen is 10%.  Today I mucked with the Linn tracks in poor listening conditions, my results were waay worse clearly guessing land.  Need to try more from HD tracks but so far that one passage from Spring is my best candidate since I've had 5+ streaks more than once.
> 
> It just occurs to me there is a guy I know that I can ask some of these DSP questions.  He happens to be an awesome jazz musician (keyboards) and a wizard with FFTs.  Next week.


 
 Oh, no way it will help ^_^ Sorry, didn't know you wanted to bias towards that. This would only make it harder. You're right in that the noise moves up too, but then it gets hit by truncation and replaced by new artifacts from that.
  


tdog said:


> RRod et al. The only decent way to turn 24-bit audio into 16-bit is to dither it properly. For information on this see:
> 
> …
> Turning 24-bit audio into 16-bit by truncating the last 8 bits is possible but not recommended, you will lose resolution as shown above..
> ...


 
 Normalization helps to prevent the problems of truncation, and is a separate issue from dithering. But yes, if you need every last inch of the Redbook dynamic range, you'll need dither, though in any non-high-range-dynamic setting, it won't really matter.


----------



## bigshot

rrod said:


> But yes, if you need every last inch of the Redbook dynamic range, you'll need dither


 
  
 Redbook is overkill.


----------



## RRod

bigshot said:


> Redbook is overkill.


 
  
 Yes, whence the if.


----------



## tdog

rrod said:


> Normalization helps to prevent the problems of truncation, and is a separate issue from dithering. But yes, if you need every last inch of the Redbook dynamic range, you'll need dither, though in any non-high-range-dynamic setting, it won't really matter.


 
  
 The "problems of truncation" are completely solved by dither - learn the theory and put it into practice properly, as per the papers I linked to.
  
 I did not confuse normalisation and dithering just because I put them in the same post, as you have suggested. Truncating 24-bit audio to 16-bit is something that no one should do these days, even for relatively compressed and low dynamic range pop music - the difference is audible on good playback systems, as per the links in my previous post.


----------



## RRod

tdog said:


> The "problems of truncation" are completely solved by dither - learn the theory and put it into practice properly, as per the papers I linked to.
> 
> I did not confuse normalisation and dithering just because I put them in the same post, as you have suggested. Truncating 24-bit audio to 16-bit is something that no one should do these days, even for relatively compressed and low dynamic range pop music - the difference is audible on good playback systems, as per the links in my previous post.


 
  
 I didn't say you confused them, I said they were two different things, and the simple fact is that the default bit-depth down-conversion in SoX is truncation, and that's what we're talking about. Give it a try yourself if you want. Make a 24bit sine wave down around -100dB, tell SoX to convert to 16bit, and you'll get nothing but zeros in the output.
  
 FWIW, here's my usual resampling script for doing ABX. Where would you like me to improve it?
  
 #!/bin/bash
 #1 = file
 #2 = lo bit
 #3 = lo rate
 #find bits, rate, and # of samples
 bits=$(sox --i "$1" 2>&1 | grep "Precision" | cut -d " " -f 8 | tr --delete "\-bit")
 rate=$(sox --i "$1" 2>&1 | grep "Sample Rate" | cut -d " " -f 7)
 samps=$(echo $(sox "$1" -n stat 2>&1| grep read |cut -d " " -f 11)/2 | bc)
 #normalize xhi-res
 peak=$(sox "$1" -n stats 2>&1 | grep "Pk lev" | sed 's/[ ]\+/ /g' | cut -d ' ' -f 4)
 gain=$(echo "(("$peak")>-3)*(-3-("$peak"))"|bc)
 sox -V4 "$1" xhi.wav gain -n "$gain"
 sox -V4 xhi.wav -b "$2" xres.wav rate "$3" dither #-s
 sox -V4 xres.wav -b "$bits" xlo.wav rate -v "$rate" trim 0s "$samps"s
 sox -m xhi.wav -v -1 xlo.wav xdiff.wav
 sox xhi.wav -n spectrogram -x 1920 -Y 1080 -o xphi.png
 sox xlo.wav -n spectrogram -x 1920 -Y 1080 -o xplo.png
 sox xdiff.wav -n spectrogram -x 1920 -Y 1080 -o xpdiff.png
 echo "Read "$samps" samples"
 echo "Processed as "$bits"-bit/"$rate"Hz"
 echo "Applied "$gain" gain"


----------



## tdog

If this was a SOX list, that might be meaningful. A software package's default action is usually the easiest, not the best. I have posted enough links to verified research - where would you like to challenge that?


----------



## RRod

tdog said:


> If this was a SOX list, that might be meaningful. A software package's default action is usually the easiest, not the best. I have posted enough links to verified research - where would you like to challenge that?


 
  
 I'm asking where in my workflow am I going against the principles in your research. And I am deliberately not using the default action of the software, precisely to get better results. And it's meaningful because it's the software being used to do the conversion that is currently under examination.
  
 It's also interesting that your first link is to a xiph video, when Monty himself at one point says in the other video "no one ever ruined a recording by not dithering the final master."


----------



## tdog

Using dither is best practice. It's virtually free these days and there is no penalty for using it. You can choose not to use it and get inferior results. Many recordings are pretty poorly made, that is no reason to advocate poor practice just because most people won't notice.


----------



## bigshot

It's easy enough to do. But I really don't think it makes a whole heck of a lot of difference. (see Ethan Winer's tests in my link below)


----------



## tdog

Well, it makes a difference _sometimes_, and that's the point. I've chatted with both Monty and Ethan over email about this (and other issues) just last year and they did agree that it's something that is so good and free that there's no reason not to do it. 16-bit resolution is indeed excellent, better than most people need for most compressed music, but for no cost you can have better within the same bit-depth (i.e. 24-bit dithered to 16-bit) and it makes a difference sometimes and to some people, so why not?


----------



## limpidglitch

Besides, Greenears wanted to give 24bit every reasonable favour, so dither goes out the window, precisely because it's 'best practice'.


----------



## RRod

tdog said:


> Well, it makes a difference _sometimes_, and that's the point. I've chatted with both Monty and Ethan over email about this (and other issues) just last year and they did agree that it's something that is so good and free that there's no reason not to do it. 16-bit resolution is indeed excellent, better than most people need for most compressed music, but for no cost you can have better within the same bit-depth (i.e. 24-bit dithered to 16-bit) and it makes a difference sometimes and to some people, so why not?


 
  
 I don't think anyone disagrees.


----------



## Don Hills

greenears said:


> ... Yes I've read most CD are mastered with peaks at -10 to -15 dBFS.   So let's say this one peaks at -10 dBFS and with 50dB DR then the bottoms are at -60dBFS and let's say the noise is at -90.  You only have 30 dB SINAD for the soft parts compared to loud parts 80 dB.  It's exactly the same thing I just said before, expressed in dB.  Since I expect 80 dB is not audible, I focus on the 30.


 
  
 You must have read a very old book. Very few CDs are mastered with peaks that low nowadays. Certainly no popular ones.
 Also, when you are doing your listening tests, remember that you aren't supposed to turn up the volume for the quiet parts. You set the volume to your normal listening level for the genre of music you're listening to and leave it at that.


----------



## macshooter

I'm don't have time to read all 134 pages of this thread to see if someone else has already tossed this out there (I would be surprised if no one had) but this article really opened my eyes a bit on this topic. Some reviews of some interesting audio equipment on that site as well.


----------



## stv014

tdog said:


> Using dither is best practice. It's virtually free these days and there is no penalty for using it.


 
  
 Actually, there is some penalty for using dither, since it increases the noise floor, and with real music (that is, not a single pure tone, or something so quiet that the quantized version will only have information in a small number of bits) that does not have a very high dynamic range, the undithered quantization error will be mostly noise anyway. See also these older posts. With 16-bit samples, it usually just does not matter either way, because neither the quantization nor the dither noise is audible under normal conditions. The perceived noise level can be reduced with noise shaping, although at low sample rates like 22050 Hz it can do more harm than good.
  
 Dithering may be best practice, as it allows for a more consistent and "analog-like" performance (i.e. a guaranteed uncorrelated noise floor), but do not assume that an undithered sample is necessarily "broken", or even sound worse overall (other than e.g. when listening a very quiet section like a long fade-out to silence at the end at a volume setting that would be unbearable for the rest of the track) than one with simple non-shaped dither.
  


tdog said:


> Truncating 24-bit audio to 16-bit is something that no one should do these days, even for relatively compressed and low dynamic range pop music - the difference is audible on good playback systems, as per the links in my previous post.


 
  
 I doubt you would be able to hear the difference with compressed pop music without some form of cheating (which includes any use of volume settings unsuitable for listening to the entire track, and/or pre-attenuating the sample before quantizing it so that the signal will have very little entropy left). In the well known Meyer and Moran tests, no dithering was used, and, while there was some criticism regarding the samples chosen, they were generally better than compressed pop music.


----------



## stv014

limpidglitch said:


> Besides, Greenears wanted to give 24bit every reasonable favour, so dither goes out the window, precisely because it's 'best practice'.


 
  
 In this case it might not matter that much, though, since ABX only tests transparency, rather than which sample sounds "better". In cases where non-dithered quantization is not transparent, the noise floor of non-shaped dither has good chances of being audible as well, and therefore lead to a positive ABX result, even if it is aesthetically less objectionable than distortion.
  
 The main problems seem to be statistical bias (doing multiple runs until one produces a p-value <= 0.05, which will sooner or later happen by simple luck), and listening to quiet sections at unrealistically high volume.


----------



## limpidglitch

stv014 said:


> In this case it might not matter that much, though, since ABX only tests transparency, rather than which sample sounds "better". In cases where non-dithered quantization is not transparent, the noise floor of non-shaped dither has good chances of being audible as well, and therefore lead to a positive ABX result, even if it is aesthetically less objectionable than distortion.
> 
> The main problems seem to be statistical bias (doing multiple runs until one produces a p-value <= 0.05, which will sooner or later happen by simple luck), and listening to quiet sections at unrealistically high volume.


 
  
 In fact we expect him to manage that in 1 out of 20 tries. Statistics is weird like that


----------



## Greenears

Thanks all for the interesting SoX incantations. I really appreciate it.
  
 I'm certainly not the OP (134 pages into this thread!) but I'm the OP on this mini sub-thread I guess.  Let me kindly restate my purpose:
  
 I am not a fan of ABX tests where someone does a series and gets fail-fail-fail.  I have seen many many such logs online and they leave me cold.  This is just my perspective.  Because it begs the question whether the tester has good enough ears, good enough transducers etc.  IMO you have to pass something first, and then gradually dial up the difficulty until you fail.
  
 So my purpose here is to do something relatively pure - you'll notice I converted the 24 bit myself rather than relying on the HD tracks 16 bit version, since you don't know how that was done.  But after that, give 24 bit every chance to win.  Look at the results as you go.  Crank the volume on soft passages.  Listen to a very short clip over and over. Everything other than looking at the electronic signature.  I know you consider this "cheating" but it is not.  If your ears can detect it, fair game.  I'm not trying to reproduce "natural listening" in my first pass.  I promise once I get a pass I'll start to dial back on these things.  Then people who view the logs can make their own decision whether it matters to them.  
  
 Please realize I fully understand the mathematics behind dither, normalization, sampling, FIR filters, and all kinds of other semi-useless information.    None of your comments are invalid, I'm just saying not yet. I am still struggling to see the magic 5% first.
  
 Which me luck! I hope to try again later today.  I also set up and converted some Linn tracks yesterday.  I encourage anyone interested to try to reproduce.   All the tracks and software I'm using is 100% free.  If anyone has a different passage or track that passes for them please let me know.


----------



## bigshot

greenears said:


> IMO you have to pass something first, and then gradually dial up the difficulty until you fail.


 
  
 I'm learning to fly like that. I started by standing on a chair and flying to the ground. I'm working my way up to jumping off the top of the Empire State Building.


----------



## Don Hills

bigshot said:


> I'm learning to fly like that. I started by standing on a chair and flying to the ground. I'm working my way up to jumping off the top of the Empire State Building.


 

 He does have a point though. The test should have positive and negative controls. Samples where the test subject should be able to tell a difference, and samples where he should not.


----------



## Greenears

bigshot said:


> I'm learning to fly like that. I started by standing on a chair and flying to the ground. I'm working my way up to jumping off the top of the Empire State Building.


 
 Oh how humorous we are .... but thanks for proving my point.  If you practice flying by increasing 3 feet at a time, at least we learn how high you can fly.  With ABX testing all I see is a bunch of people jumping from different skyscrapers - all with the same predictable result.  That is why you don't learn that much, except what you already expected. 
  
 Remember the only thing you can absolutely learn from ABX testing is that if you pass you were absolutely NOT guessing (down to extremely unlikely probability).  A fail tells you close to nothing - A could have been Bach and B Mozart and maybe the listener was just not paying attention.


----------



## RRod

greenears said:


> Oh how humorous we are .... but thanks for proving my point.  If you practice flying by increasing 3 feet at a time, at least we learn how high you can fly.  With ABX testing all I see is a bunch of people jumping from different skyscrapers - all with the same predictable result.  That is why you don't learn that much, except what you already expected.
> 
> Remember the only thing you can absolutely learn from ABX testing is that if you pass you were absolutely NOT guessing (down to extremely unlikely probability).  A fail tells you close to nothing - A could have been Bach and B Mozart and maybe the listener was just not paying attention.


 
  
 It all comes down to the listeners intentions. If I want to fail an ABX deliberately, I just pick A every time. But if someone said "I will pay you $100 if you can pass", then I'm listening intently to find the difference. It's fine to futz around with improper variations of a test to understand its inner workings, but eventually you have to actually take it in earnest.


----------



## castleofargh

greenears said:


> bigshot said:
> 
> 
> > I'm learning to fly like that. I started by standing on a chair and flying to the ground. I'm working my way up to jumping off the top of the Empire State Building.
> ...


 
 I like how you went from first timer in ABX, to telling us what abx can and cannot do in a week.
	

	
	
		
		

		
			




 I would think that most people here have a little understanding of statistical significance and have looked at most of Arny's explanations about the ups and downs of abx on his website or now on hydrogen.
  
 anyway the logical "solution" for you then if you're looking for positive samples and progressive evaluation, would be to try abx starting with 6 or 8bits and go up. if all you're asking for is when does the quantization noise is audible or when is the track eaten too much. but we already have a pretty good idea about that, it would be a little under 10 or 12bits for most people at normal listening level. which was about what we had when 16bit dacs where only 16bit.
  
 but as RRod said, it a matter of what you're trying to achieve with abx. looks like you're trying to find any way to succeed, your not trying to know if you would succeed under normal conditions. just get the volume loud enough on a calm enough passage and you'll win 100%. the end. 
 but that's as fake as trying to show you can hear 20khz by pushing the tone at 140db. sure you can succeed the test, but what did that tell you that you didn't already knew know?


----------



## bigshot

I guess it depends on what the purpose of the test is in the first place... whether it's just to find out what does and doesn't make a difference in practical real world use, or whether it's that you want to beat the test.


----------



## sonitus mirus

I went through a similar issue when attempting to find an acceptable bit rate with a lossy codec to use for my music.  After reading about a dozen or more different descriptions of what an artifact sounded like and what types of audio differences that might occur with lossy files, I simply made a very low bit rate mp3 that had obvious artifacts so that I could hear these for myself.


----------



## RRod

bigshot said:


> I guess it depends on what the purpose of the test is in the first place... whether it's just to find out what does and doesn't make a difference in practical real world use, or whether it's that you want to beat the test.


 
  
 Yeah, we're in "Kirk taking the Kobayashi Maru Test" phase, it seems.


----------



## sonitus mirus

rrod said:


> Yeah, we're in "Kirk taking the Kobayashi Maru Test" phase, it seems.


 
  
 Clearly Kirk cheated and lacks ethics. 
	

	
	
		
		

		
		
	


	




 I thought the point of the test was to determine how one deals with failure.  I was the only freak in an entire business team of over 30 people that felt this way when this was used as a group exercise at some seminar.  Everyone else thought it was innovative thinking.


----------



## Krutsch

sonitus mirus said:


> Clearly Kirk cheated and lacks ethics.
> 
> 
> 
> ...


 

 You attended a business seminar where one of the tasks was the Kobayashi Maru test from the fictional Star Trek universe?  
 Did they build a working replica of the bridge with simulated explosions?


----------



## sonitus mirus

One of the team building exercises was to see how we felt about the actions that Kirk took in order to pass a test that was designed to fail.    We didn't actually take the Kobayashi Maru test.   The idea was that it was supposed to bring about a discussion, all in good fun.  I was the only one in the group that thought what Kirk did was wrong.


----------



## Krutsch

sonitus mirus said:


> We didn't actually take the Kobayashi Maru test.


 
  
 But... everyone in that group knew to what that referred? Was this at Comic-con?


----------



## sonitus mirus

krutsch said:


> But... everyone in that group knew to what that referred? Was this at Comic-con?


 
  
 I didn't have a clue to what it was before.  It was a Power Point presentation; just a paragraph describing what it was all about.  We answered some question along the line of whether we thought Kirk's action was commendable or wrong, and then shared our results and discussed it in the group.  It was not that detailed.  It was a simple exercise about a no-win situation and creative thinking.  My employer sent 30 of us to this seminar, and it was just one small exercise of many.
  
 Nobody had to be a Trekkie to grasp the concept and participate.


----------



## RRod

sonitus mirus said:


> Clearly Kirk cheated and lacks ethics.
> 
> 
> 
> ...


 
  
 "it's a test of character", Kirk says, and it wouldn't surprise me if the character of business is to cheat to win :3 The thing with ABXing hi-res vs. Redbook is that those of us who have tried it ourselves and verse ourselves, to varying degrees, in the theory (both audio and statistical) know what the outcome will be when the test is done correctly. So when a Kirk "passes" the test, we know something is up. Once the truth comes out, there's always some "cheat" that occurred, and if it "had the virtue of never having been tried", as with Kirk, we might give a pat on the back. But we've seen this stuff too many times now for that


----------



## Greenears

Wow. Tough crowd. All I did was use minimal flags and default settings and dared to share interim results. Before I could shake my audiophile tail I'm jumping off skyscrapers into a fictional universe. Nice to see everyone has an open mind.


----------



## Greenears

The track is the 2nd clarinet track test192 from Linn sampler (I put it in the HD tracks directory with the others).
  
 I mainly focused on the 6th note, listening for a little distortion or texture on the start of that note that I thought was in the 24 bit.
  
 sox -G test192.flac -b 16 test192_16.flac
 C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sa
 mpler>sox --i test192.flac
 Input File     : 'test192.flac'
 Channels       : 2
 Sample Rate    : 192000
 Precision      : 24-bit
 Duration       : 00:00:29.28 = 5621672 samples ~ 2195.97 CDDA sectors
 File Size      : 17.2M
 Bit Rate       : 4.69M
 Sample Encoding: 24-bit FLAC

 C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sa
 mpler>sox --i test192_16.flac
 Input File     : 'test192_16.flac'
 Channels       : 2
 Sample Rate    : 192000
 Precision      : 16-bit
 Duration       : 00:00:29.28 = 5621672 samples ~ 2195.97 CDDA sectors
 File Size      : 6.25M
 Bit Rate       : 1.71M
 Sample Encoding: 16-bit FLAC
 Comment        : 'Comment=Processed by SoX'

 C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sa
 mpler>c
  
 foo_abx 1.3.4 report
 foobar2000 v1.3.6
 2014/12/15 22:48:27
 File A: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\test192.flac
 File B: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\test192_16.flac
 22:48:27 : Test started.
 22:50:09 : 00/01  100.0%
 22:51:17 : 00/02  100.0%
 22:51:37 : 00/03  100.0%
 22:51:58 : 01/04  93.8%
 22:53:31 : 02/05  81.3%
 22:54:06 : 03/06  65.6%
 22:55:00 : 04/07  50.0%
 22:56:28 : 04/08  63.7%
 22:56:50 : 05/09  50.0%
 22:57:17 : 06/10  37.7%
 22:58:59 : 07/11  27.4%
 22:59:27 : 08/12  19.4%
 23:00:42 : 08/13  29.1%
 23:01:11 : 09/14  21.2%
 23:01:52 : 09/15  30.4%
 23:02:45 : 10/16  22.7%
 23:03:17 : 11/17  16.6%
 23:03:52 : 11/18  24.0%
 23:04:46 : 12/19  18.0%
 23:05:34 : 13/20  13.2%
 23:06:23 : 14/21  9.5%
 23:07:54 : 15/22  6.7%
 23:09:07 : 15/23  10.5%
 23:09:27 : 15/24  15.4%
 23:09:50 : 16/25  11.5%
 23:10:25 : 17/26  8.4%
 23:10:45 : 18/27  6.1%
 23:11:20 : 18/28  9.2%
 23:11:57 : 18/29  13.2%
 23:12:23 : 19/30  10.0%
 23:12:43 : 20/31  7.5%
 23:13:20 : 20/32  10.8%
 23:13:48 : 21/33  8.1%
 23:14:47 : 21/34  11.5%
 23:15:48 : 22/35  8.8%
 23:17:17 : 23/36  6.6%
 23:18:04 : 23/37  9.4%
 23:18:17 : 23/38  12.8%
 23:18:43 : 23/39  16.8%
 23:19:17 : 24/40  13.4%
 23:20:14 : 24/41  17.4%
 23:20:35 : 25/42  14.0%
 23:21:04 : 25/43  18.0%
 23:21:24 : 26/44  14.6%
 23:21:54 : 26/45  18.6%
 23:23:02 : 26/46  23.1%
 23:23:37 : 27/47  19.1%
 23:24:06 : 28/48  15.6%
 23:24:32 : 28/49  19.6%
 23:25:24 : 29/50  16.1%
 23:26:02 : 30/51  13.1%
 23:26:46 : 30/52  16.6%
 23:27:39 : 31/53  13.6%
 23:28:07 : 31/54  17.0%
 23:28:28 : 32/55  14.0%
 23:28:47 : 33/56  11.4%
 23:29:14 : 33/57  14.5%
 23:29:40 : 34/58  11.9%
 23:30:15 : 35/59  9.6%
 23:31:11 : 36/60  7.8%
 23:32:03 : 37/61  6.2%
 23:32:43 : 37/62  8.1%
 23:34:02 : 38/63  6.5%
 23:35:13 : 38/64  8.4%
 23:36:07 : 39/65  6.8%
 23:36:54 : 40/66  5.4%
 23:37:16 : 41/67  4.3%
 23:38:17 : 42/68  3.4%
 23:41:16 : 42/69  4.6%
 23:42:33 : 42/70  6.0%
 23:42:52 : 43/71  4.8%
 23:43:35 : 43/72  6.2%
 23:44:12 : 43/73  8.0%
 23:44:31 : 44/74  6.5%
 23:44:57 : 44/75  8.3%
 23:45:01 : Test finished.
  ---------- 
 Total: 44/75 (8.3%)
  
 Do you think I'm guessing?


----------



## stv014

> Do you think I'm guessing?


 
  
 Maybe you can hear something (it is not impossible when listening to some quiet part at high volume), but if you do, it is a very minimal difference with all those failed trials. If you continue the test, combining the results with this run, the p-value should keep going down on average if you do actually hear a difference, so you should eventually get well under 1% (keeping the 44/75 ratio constant, it would be 2%, 0.56%, and 0.16% for 88/150, 132/225, and 176/300, respectively). Also, do not discard any runs just because the score was bad. Selective inclusion guarantees "winning" over enough time, even with pure random guessing.


----------



## Greenears

This one the volume wasn't so pumped up - I would say I listened at a little higher than medium volume.  I could listen to the whole piece at that.  
  
 It's very subtle and maybe just one note.  I thought maybe there was a slightly different texture on the first note but very difficult to place it with A or B so I gave up on that and focused on the 6th.
  
 I seem to do better with short passages. I've also tried just to sit back and "feel" the whole experience without focusing on specific passages but not seen much success.
  
 Are others posting more convincing logs for 16 vs 24 (down to 1%?)


----------



## RRod

greenears said:


> Wow. Tough crowd. All I did was use minimal flags and default settings and dared to share interim results. Before I could shake my audiophile tail I'm jumping off skyscrapers into a fictional universe. Nice to see everyone has an open mind.


 
  
 Oh the extended Kirk commentary wasn't about you, just the general problem with the ABX MMO. Do you have a link for that Linn track?


----------



## castleofargh

greenears said:


> This one the volume wasn't so pumped up - I would say I listened at a little higher than medium volume.  I could listen to the whole piece at that.
> 
> It's very subtle and maybe just one note.  I thought maybe there was a slightly different texture on the first note but very difficult to place it with A or B so I gave up on that and focused on the 6th.
> 
> ...


 

 but it's all about how loud you go. the difference is the noise floor and should be only the noise floor, so as long as you can go loud enough to make the noise floor audible on the 16bit you'll win. there really is nothing else to this test I think. if you were listening at 90db, the noise floor would be below 1db loud in a room probably saturated with noises around 30db or more+the music itself.
 so you either go too loud on a silent passage just so you can win, and we're back to what we were talking about. or else that would suggest another problem not directly related to the track being in 16bit.
 that's why loudness is all but a trivial matter.


----------



## Greenears

rrod said:


> Oh the extended Kirk commentary wasn't about you, just the general problem with the ABX MMO. Do you have a link for that Linn track?


 
 Oh, the commentary is definitely about me.  I'm Kirk.  Since those with superior audio intellect already know the outcome of the KM test, the only way I can get a different outcome is to cheat by reprogramming the ABX tester or similar.
  
 http://download.linnrecords.com/test/flac/test192.aspx
  
 FYI I also tried the shorter Linn http://download.linnrecords.com/test/flac/recit24bit.aspx  but I couldn't find anything audible to latch onto and my scores were 60-100%.  I gave up after a couple short tries at it.  If you want my logs I can dig them up.


----------



## RRod

> Originally Posted by *Greenears* /img/forum/go_quote.gif
> 
> Do you think I'm guessing?


 
  
 All depends on your acceptance criterion. Basically, at the beginning of the test you have to specify two of three things:
 .the total number of trials
 .the false positive rate
 .a false negative rate for a given outcome
  
 If you chose 75 trials and a 10% false positive rate, then yes you can say "I reject the notion that I am guessing." But if you set a 5% false positive rate, then you can't reject. It is breaking the rules to select your rate after you see results. Me personally, I require 19/25 for myself, since I want <=1% false positives and get fatigued with many more tests, plus it has decent power for picking up true results when the actual probability of hearing a difference is >80%.
  


greenears said:


> Oh, the commentary is definitely about me.  I'm Kirk.  Since those with superior audio intellect already know the outcome of the KM test, the only way I can get a different outcome is to cheat by reprogramming the ABX tester or similar.
> 
> http://download.linnrecords.com/test/flac/test192.aspx
> 
> FYI I also tried the shorter Linn http://download.linnrecords.com/test/flac/recit24bit.aspx  but I couldn't find anything audible to latch onto and my scores were 60-100%.  I gave up after a couple short tries at it.  If you want my logs I can dig them up.


 
  
 Oh the first Kirk post was about you definitely, but in the sense that it's fine to people to try to figure out how to beat the system. Still, though, at some point you have to get it set up all right. I'll try the Linn recording myself when I get home.


----------



## bigshot

greenears said:


> Since those with superior audio intellect already know the outcome of the KM test, the only way I can get a different outcome is to cheat by reprogramming the ABX tester or similar.


 
  
 It might not be cheating. You might just be not setting up the test right, or you may be unconsciously allowing bias to alter your results.


----------



## limpidglitch

greenears said:


> Oh, the commentary is definitely about me.  I'm Kirk.  Since those with superior audio intellect already know the outcome of the KM test, the only way I can get a different outcome is to cheat by reprogramming the ABX tester or similar.
> 
> http://download.linnrecords.com/test/flac/test192.aspx
> 
> FYI I also tried the shorter Linn http://download.linnrecords.com/test/flac/recit24bit.aspx  but I couldn't find anything audible to latch onto and my scores were 60-100%.  I gave up after a couple short tries at it.  If you want my logs I can dig them up.


 
  
 I gave it a go, doing it like this:
  
 sox test192.flac test9624.wav rate -v 96k #I don't think my soundcard supports 192k output.
 sox test9624.wav -b 16 -t wav - -D | sox -t wav - -b 24 test9616.wav
  
 Result was 6/10.
 Only noise I could hear was the background noise of the recording itself, and I could hear no distortion or other artefacts.
  
 I then made a diff file and checked the stats; everything looked normal.
 I know my own physiological limitations well enough to say that the differences are indeed inaudible.
  

  
 Left
 Right
 Peak Amplitude:
 -95,87 dB
 -95,87 dB
 True Peak Amplitude:
 -90,72 dBTP
 -90,67 dBTP
 Maximum Sample Value:
 135
 135
 Minimum Sample Value:
 -134
 -134
 Possibly Clipped Samples:
 0
 0
 Total RMS Amplitude:
 -101,26 dB
 -101,25 dB
 Maximum RMS Amplitude:
 -100,82 dB
 -100,88 dB
 Minimum RMS Amplitude:
 -132,53 dB
 -132,43 dB
 Average RMS Amplitude:
 -102,15 dB
 -102,15 dB
 DC Offset:
 0
 0
 Measured Bit Depth:
 24
 24
 Dynamic Range:
 31,71 dB
 31,55 dB
 Dynamic Range Used:
 31,30 dB
 31,20 dB
 Loudness:
 -101,32 dB
 -101,28 dB
 Perceived Loudness:
 -∞ dB
 -∞ dB
 ITU-R BS.1770-2 Loudness: (selection too short)
  
  
  
  
 0dB = FS Square Wave
  
  
 Using RMS Window of 50,00 ms
  
  
 Account for DC = true


----------



## Greenears

@limp: Thanks for running that.  I really appreciate it.  Real runs and results I'm a fan of - whatever the outcome or shortcomings.  Inuendo about my Starfleet cadet history .... not so much.  
  
 To be clear for everyone, referenced by you and several other posts, I am not hearing any noise floor.  There was speculation about cranking the volume to the level where I could hear noise.  I have not done that, and no matter what volume I tried I did not hear noise or any other non-music artifact in any of the 12 clips I have, in either 16 or 24.
  
 The way I got the result on the above Linn test192 was listening for a certain quality I heard on the  6th note played by the clarinet.  It is difficult to describe, but it is a certain roughness on the front end of that held note, just the first little bit, in the 24b version.  The 16b sounds a little thinner or purer. I'm not sure whether it is the reed or actually distortion within the instrument as that note is played louder.  I also tried listening to the reed "shhh" on the first note, and the sounds of the key pads closing which I can hear quite easily in both the 16 and 24.  Unfortunately I could not zone in on any difference and that probably accounts for about 10 trials.
  
 Give it a try.  Maybe you have better equipment. 
  
 So the question is am I guessing?  Well I don't really know.  I feel I can recognize that detail, but sometimes it is very hard to pick up.  Then I start to hear it well, and I get a streak of successes, often with quick decisions.  Then it eludes me for a bit.  I saw one poster recommend taking a break every 5th trial.  Good idea but I havnt had the time yet.
  
 @RRod:  Honestly I don't have a set criteria.  I'm open for input.  Many posters suggest at least 5% or better 1%.  If I saw any run of >10 trials with 1% result I would call that absolutely not guessing.  I did hit 5% a couple of times with lots of trials so I'm really on the fence.  Note my better runs correspond to passages and pieces where I've heard something specific.  I have other runs on different pieces that look very different, starting at 60% and heading straight to 100% and stayin gin the range.  I can post those logs if someone wants.  The two passages I have some success start at 50 and head to 10 and then stay in the 30-40 range.   I consider any run valid as long as you are trying really really hard to win on every trial. 
  
 My current leaning is that chances are high that I am hearing something but it is really subtle.  But my mind may change with more testing.  I did not expect this.  I thought if I heard something once I learned what to look for I would get a 9/10 run.  But that never happened. 
  
 And no I am not acting like Kirk.  Kirk reprogrammed because he couldn't face failure.  I can accept failure but I still really really want to know the truth.  I am very curious.


----------



## bigshot

The difference between 16 bit and 24 bit, all things being equal, is the depth of the noise floor. There is no difference in resolution up in the audible range. If you are hearing differences in tone on a clarinet note, it's probably just unconscious expectation bias at work. I'm sure if you did enough tests, it would come out as random chance.


----------



## Krutsch

^^ What about sample rate? I know he is knocking down the bit-depth from 24 to 16, and leaving the sample rate at 96 kHz, but what if the opposite were true: downsampling to 44.1 (or 48) from 96, but leaving the bit-depth alone for the sake of argument. Would it be reasonable to find a passage, section or notes where these subtleties might be audible?
  
 I'm sincerely curious, not trolling...


----------



## bigshot

Only if you can hear frequencies humans can't hear.


It's pretty safe to say that if you hear a difference between redbook and high bit/sampling rate, it's the difference in the way your equipment is playing the file that makes the difference, not the sound in the recording.


----------



## RRod

greenears said:


> @limp: Thanks for running that.  I really appreciate it.  Real runs and results I'm a fan of - whatever the outcome or shortcomings.  Inuendo about my Starfleet cadet history .... not so much.
> 
> To be clear for everyone, referenced by you and several other posts, I am not hearing any noise floor.  There was speculation about cranking the volume to the level where I could hear noise.  I have not done that, and no matter what volume I tried I did not hear noise or any other non-music artifact in any of the 12 clips I have, in either 16 or 24.
> 
> ...


 
  
 Really didn't mean to get you so riled up with the Star Trek ref. Anyway, here are my test results:

  
 Here is the spectrogram of the difference between my two files:

  
 Using Sox, I did a convert to 16-bit with shaped dither and re-upsampled to 24-bit. No normalization was needed as the peak is only at -6.74.


----------



## Greenears

bigshot said:


> The difference between 16 bit and 24 bit, all things being equal, is the depth of the noise floor. There is no difference in resolution up in the audible range. If you are hearing differences in tone on a clarinet note, it's probably just unconscious expectation bias at work. I'm sure if you did enough tests, it would come out as random chance.


 
 75 is a lot of trials.  It's hard to say I need to do more trials.
  
 I'm sure you are familiar with the fact that quantization noise is a nonlinear process that cannot be mathematically described with differential equations. The partial analysis shows that at some specific frequencies (harmonics of Fs) the noise energy can get concentrated.  I'm sure you are familiar with the details but http://qtwork.tudelft.nl/~schouten/linkload/adc-tutorial.pdf is a reasonable overview that is repeated in all the textbooks at length, for those that aren't.  Describing Q-noise as something close to Gaussian noise floor works for most of the time, but are there could be exceptions.
  
 So if I am hearing something I believe it is most likely subtle phase shifts.  I say phase and not amplitude since I believe that we are not as sensitive to amplitude changes.  Think of playing a major third on a piano.  If you hit the 2 keys with equal pressure compared to 1 key twice as hard as the other, both still sound like a major third and may actually be very difficult to distinguish at all.  But if the piano tuner turns the key even a little on one of the strings, the color of the chord changes immediately, even very subtle shifts can be picked up right away.  We are wired to detect harmonies and dissonances, I don't know why.  
  
 It the effect is psychological I am well aware I can fool myself, but I can't fool the machine.  The log is there, that's the test run.  Is it consistent with guessing?  I don't think it's that easy to dismiss.


----------



## limpidglitch

greenears said:


> @limp: Thanks for running that.  I really appreciate it.  Real runs and results I'm a fan of - whatever the outcome or shortcomings.  Inuendo about my Starfleet cadet history .... not so much.
> 
> To be clear for everyone, referenced by you and several other posts, I am not hearing any noise floor.  There was speculation about cranking the volume to the level where I could hear noise.  I have not done that, and no matter what volume I tried I did not hear noise or any other non-music artifact in any of the 12 clips I have, in either 16 or 24.
> 
> ...


 
  
 I haven't seen much Star Trek, so I've got absolutely no idea what that Kirk stuff is about. My Trek trivia knowledge begins and ends with the popular knowledge that being a red is a risky life, because so many of them die. But in fact the opposite is true, their mortality rate is the lowest of all the groups on board.
  
 When listening to that sample I concentrated on the very beginning, thinking that if there was a difference it would be in the form of a slight change in the noise profile. But while the track is pretty quiet, it isn't dead quiet, and what little difference there might have been probably got swamped.
  
 I usually take a two-tier approach to these tests. First 10 trials, p-value=0.05. If I pass with good margin (say 1%) I call it a day, but if it's a close call I do 20 or 30 new trials (in two or three sittings), with a stricter criterion.


----------



## RRod

greenears said:


> 75 is a lot of trials.  It's hard to say I need to do more trials.
> 
> I'm sure you are familiar with the fact that quantization noise is a nonlinear process that cannot be mathematically described with differential equations. The partial analysis shows that at some specific frequencies (harmonics of Fs) the noise energy can get concentrated.  I'm sure you are familiar with the details but http://qtwork.tudelft.nl/~schouten/linkload/adc-tutorial.pdf is a reasonable overview that is repeated in all the textbooks at length, for those that aren't.  Describing Q-noise as something close to Gaussian noise floor works for most of the time, but are there could be exceptions.
> 
> ...


 
  
 The tuner is listening for periodic variations in amplitude caused by cancellation between the untuned notes, with period defined by the difference (beat) frequency. And yes, your log is consistent with guessing if you're getting a p-value of 8%.


----------



## Greenears

rrod said:


> Really didn't mean to get you so riled up with the Star Trek ref. Anyway, here are my test results:
> ......
> Using Sox, I did a convert to 16-bit with shaped dither and re-upsampled to 24-bit. No normalization was needed as the peak is only at -6.74.


 
 Consider me duly riled.  It wasn't just you.  Think about it: The whole world is on the cusp of going 24-bit crazy (well the audiophile world anyway).  Someone comes along and tries to do some honest ABX testing.  And within nanoseconds I'm accused of constructing some kind of devious Kobayashi Maru no-win test (because of course everyone _knows_ what the right answer is) and if I somehow pass I must have rigged it.  Really.  
  
 Anyway back to the testing - so you got 15/25 on the Linn test192 clarinet, right? Do you think you heard anything specific?
  
 I just did another run, this time with Cassandra Wilson from HD Tracks sampler.  Same Sox -b 16 not dither no upsampling.  This time however I unchecked "look at the results".  I also discovered that when you click that it doesn't tell you how many trials (?) and then I got interrupted. I definitely intended to do 10 or more.
  
 The biggest change was on this I just played A and B a couple times each, then played X then Y from the start and normal med-low volume extended listening conditions.  I just closed my eyes and played about 16-24 bars from time 0 until I had the feel taking in the whole groove & rythm and not trying to focus on anything specific.  Played X once, Y once, whichever I liked better I picked.  Frankly I'm surprised how well I did.  I got interrupted at this point, and I know it is not enough trials.  But still - interesting.  Totally different method than the previous "successes".   This method I would say is much more akin to "natural" listening.  Hmmm.  
  
 The ABX is not co-operating.  It would be much more convenient just to get results all over the place and conclude it's a wash and go back to fighting clingons.
  
 foo_abx 1.3.4 report
 foobar2000 v1.3.6
 2014/12/16 17:00:38
 File A: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\08-Another Country.flac
 File B: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\cassandra16.flac
 17:00:38 : Test started.
 17:03:43 : 01/01  50.0%
 17:04:33 : 02/02  25.0%
 17:05:41 : 03/03  12.5%
 17:06:34 : 04/04  6.3%
 17:07:21 : 04/05  18.8%
 17:08:42 : 05/06  10.9%
 17:09:34 : 05/07  22.7%
 17:09:43 : Test finished.
  ---------- 
 Total: 5/7 (22.7%)


----------



## bigshot

I think either your equipment is introducing noise somewhere that wouldn't be there if the conversion and playback were being handled properly, or you are genuinely trying to fairly administer the test to yourself, but are unable to.


----------



## RRod

greenears said:


> Consider me duly riled.  It wasn't just you.  Think about it: The whole world is on the cusp of going 24-bit crazy (well the audiophile world anyway).  Someone comes along and tries to do some honest ABX testing.  And within nanoseconds I'm accused of constructing some kind of devious Kobayashi Maru no-win test (because of course everyone _knows_ what the right answer is) and if I somehow pass I must have rigged it.  Really.
> 
> Anyway back to the testing - so you got 15/25 on the Linn test192 clarinet, right? Do you think you heard anything specific?
> 
> ...


 
  
 Here was my method:
 1) Do the conversion, don't look at anything
 2) Set everything to 24/192k
 3) Do the trials
 4) Look at spectrograms and see if everything looks right
  
 For 3), I marked a number of passages (breathiness of clarinet on first long note, quiet piano parts, fade out, etc.), and tried my best to hear a difference in a couple of marks for each sample. I really couldn't latch on to anything, and the difference spectrogram concurs: the only difference is some noise, and that's already at -100dB or so and that's in the noise-shaping region. I went ahead and downloaded Linn's recording of the Poulenc organ concerto, and it's the same thing: just a bit of noise is the difference.


----------



## Krutsch

bigshot said:


> Only if you can hear frequencies humans can't hear.


 
  
 And that's because the increased sample rate contributes to the range of frequency response (e.g. 96 kHz for a sample rate of 192 kHz), as opposed to adding greater resolution to the audible Redbook range (i.e. 20 - 20 kHz), correct?
  


> bigshot said:
> 
> 
> > It's pretty safe to say that if you hear a difference between redbook and high bit/sampling rate, it's the difference in the way your equipment is playing the file that makes the difference, not the sound in the recording.


 
  
 Ignoring the possibility that the high bit/sampled version is from a different master than the Redbook version, of course; I might expect the higher bit-rate/depth tracks to place additional load on the playback system (i.e. the files are much larger, requiring more of everything to play them back: CPU load to decode and convert, network throughput, HDD activity).
  
 So, what's the motivation for standards like DTS-HD and Dolby True HD, et al., that support 24-bit depth and high sample rates (outside of additional surround channels and volume leveling)?
  
 I can see recording / processing sound tracks that way to mitigate rounding errors (big difference in this regard between 24 and 16 bit, from a numerical analysis standpoint). But when presented to the listener, it's all marketing spin (i.e. "as the artist intended" or as the "director originally recorded" for ease of post-production)?  I suppose, at the point, why not just release as-is and use it to push a new format?
  
 Anyway... thanks.


----------



## Greenears

rrod said:


> For 3), I marked a number of passages (breathiness of clarinet on first long note, quiet piano parts, fade out, etc.), and tried my best to hear a difference in a couple of marks for each sample. I really couldn't latch on to anything, and the difference spectrogram concurs: the only difference is some noise, and that's already at -100dB or so and that's in the noise-shaping region. I went ahead and downloaded Linn's recording of the Poulenc organ concerto, and it's the same thing: just a bit of noise is the difference.


 
 Yup, I tried "breathiness" of first clarinet note as well.  At first my mind was sure it sounded more resolved or "real" on the 24 but when I tried to use that to ABX I did horrible.  So I abandoned that pretty quickly.  
  
 I don't consider white noise (or even slightly pink noise) to be an issue with 16.  Humans have an amazing ability to tune out random backgrounds, go back and listen to tapes from the 80s compared to CDs.  I never hear or notice noise on redbook, and haven't heard on any testing.  It is too far down.  So I don't consider it valid to find a short gap, mark that and amplify the heck out of it until you can hear noise and ABX that to pass.  Yes I may be Kirk, but even he wouldn't do that.


----------



## bigshot

krutsch said:


> And that's because the increased sample rate contributes to the range of frequency response (e.g. 96 kHz for a sample rate of 192 kHz), as opposed to adding greater resolution to the audible Redbook range (i.e. 20 - 20 kHz), correct?


 
  
 Yes. Sampling rate is frequency response. Bit depth is noise floor. If you know how digital audio works, you know what to expect. Inaudible is inaudible. If someone comes up with a way to hear the unbearable, you give them the benefit of the doubt and figure that they are doing something wrong without realizing it. The truth is that none of it makes a lick of difference. If you have to sweat and strain and fudge your numbers to be able to hear something, it just doesn't matter anyway.
  


krutsch said:


> So, what's the motivation for standards like DTS-HD and Dolby True HD, et al., that support 24-bit depth and high sample rates (outside of additional surround channels and volume leveling)?


 
  
 Marketing?


----------



## castleofargh

greenears said:


> rrod said:
> 
> 
> > For 3), I marked a number of passages (breathiness of clarinet on first long note, quiet piano parts, fade out, etc.), and tried my best to hear a difference in a couple of marks for each sample. I really couldn't latch on to anything, and the difference spectrogram concurs: the only difference is some noise, and that's already at -100dB or so and that's in the noise-shaping region. I went ahead and downloaded Linn's recording of the Poulenc organ concerto, and it's the same thing: just a bit of noise is the difference.
> ...


 

 but noise floor is or at least should be the only difference from changing the bit depth. so by dismissing the noise floor difference, I see it like dismissing the factual difference to look for something that shouldn't exist. I never said you were not hearing something, but you have to admit that you're mindset for going at it is strange.
 it's always possible that the conversion somehow changed something it shouldn't, then the abx prog will turn the 2 files into 32bit pcm tracks for the test, those 2 pcm streams will then be turned down to whatever you have set on your computer's output(foobar is in 24bit? are you using direct input? ...). it's all a matter of adding zeros and then cut them out, it shouldn't change the value of the music sample themselves as they are well above 16bit in value. but still that's a lot of ups and downs for a file and maybe somehow somewhere, something goes wrong for one of them? but even if that happens, how can you attribute the difference to the track being in 16bit, or even to the first conversion to 16bit?
 by turning the 16bit back to 24bit, you add yet another change, but at least you know that the computer will treat both files the same way. that's why even though that should not change a thing, most of us will do that and put the files we want to test back into a common resolution/file format. just to be sure we're testing the track and not the computer or the DAC.
 but when we make suggestions, apparently if it's not RRod you decide you know better. (yeah I'm jealous
	

	
	
		
		

		
			





).
 same for dithering, if the noise floor isn't audible why should that matter to add dither? CDs are dithered, so it would seem like a more honest comparison.
  
 as for us knowing the result in advance and trying to see you fail (or whatever), there had been quite a few tests done before you, a few AES papers,and of course tests we did for ourselves. this isn't really cutting edge experiment and anybody can do it with free and easy to use tools. so yeah we tend to feel like we already know the end of the movie. I really don't see what's wrong with that? people getting positive results are marginal, and usually when they don't run away insulting us for doubting them, it ends up that the files were different, one way or another. you can feel offended when we suspect something done wrong in your trials if it pleases you, but it's not because we secretly hate you or that we're all members of the 16bit lobbying cult. it's because you offer us an unlikely result that goes against what is mostly recognized(by science and engineers, maybe not so much by "audiophiles" 
	

	
	
		
		

		
		
	


	




) for abx at normal listening levels.
  
 if you come telling me that you can abx a flac from a mp3@96kb I will not try to find a reason why you succeed, because it is expected for you to do so. it has nothing to do with egos or knowing better, it's about you saying that you can identify 16bit from 24bit when pretty much any controlled tests resulted in people unable to tell 16/44 from any superior resolutions whatever the file format or the resolution. DVD, DSD, PCM they all failed to show audible differences one after the other. and that's why we look for a reason for you more than guessing results that isn't the track being 16bit. maybe RRod can send you his converted file or you send yours(short sample else copyright police will strike us all dead) so we can start by making sure the conversion went ok? that would be one less possible bias in the way.


----------



## RRod

Don't be jelly, brah!


----------



## Danz03

Signal to noise ratio for 24bit is 144dB, signal to noise ratio for 16bit is 96dB. One of the best analogue 2 track master recorders was the Studer A820, with 77dB signal to noise ratio at 1/2" 30ips and crosstalk of 65dB at 1kHz, most digital recordings in the 80s and 90s were mastered in 16bit; so unless the 24bit file was recorded within the last 15 years, the 2 track master that it was mastered from would unlikely to have signal to noise ratio better than 96dB anyway. And if it was remastered using some kind of denoisng plugins, why would anyone need to pay for the privilege? Anyone could have done it himself with a denoiser plugin.
  
 Quote:


castleofargh said:


> but noise floor is or at least should be the only difference from changing the bit depth. so by dismissing the noise floor difference, I see it like dismissing the factual difference to look for something that shouldn't exist. I never said you were not hearing something, but you have to admit that you're mindset for going at it is strange.


----------



## castleofargh

rrod said:


> Don't be jelly, brah!


 

 I know it's because I'm bald.


----------



## Krutsch

> Originally Posted by *castleofargh* /img/forum/go_quote.gif
> 
> <snip, snip>
> 
> ...


 
  
 Do you have a link you can share? I've read examples of controlled tests for audibility of various bit rates of compressed music, but I haven't seen examples of the above. I can Google, too, but if you have links handy, I would like to read/learn more. I'm more interested in the testing methodology than the actual results, to be frank.


----------



## RRod

krutsch said:


> Do you have a link you can share? I've read examples of controlled tests for audibility of various bit rates of compressed music, but I haven't seen examples of the above. I can Google, too, but if you have links handy, I would like to read/learn more. I'm more interested in the testing methodology than the actual results, to be frank.


 
  
 Here's one:
 http://www.drewdaniels.com/audible.pdf


----------



## castleofargh

krutsch said:


> > Originally Posted by *castleofargh* /img/forum/go_quote.gif
> >
> > <snip, snip>
> >
> ...


 
 wow not having access to AES I'm probably not the most reliable source. I just come across one from time to time, but I can't say I'm organized enough to even pretend I have a list ^_^.
 the first that comes to mind is indeed moran&meyer linked by RRod. I think it's the one that was both recognized and meaningful. I think it closed the book for many people about the purpose of hires. as a listener at least, because we all understand why sony would want to get new support patents from time to time.
 one I've posted myself not long ago https://www.gearslutz.com/board/attachments/high-end/6491d1114045260-why-didnt-dsd-catch-reshaping_digital_-audio.pdf  with a relatively small number of participants so not really conclusive but following the general idea.
 a good deal of websites or groups of audiophiles have at some point conducted some sort of test like that with more or less controls. the latest being probably at archimago's bog. but that's totally uncontrolled, the results are based on trusting people not to open the files in an audio editor.
  
 if it's only for methodology of tests, then I guess you can also look for hires vs hires.


----------



## Krutsch

^^^ Thanks for the suggestions.


----------



## limpidglitch

If you're interested in how they do it within the industry, this paper on testing codecs for use in DAB might be interesting.
 The testing standards discussed can be found here, and here.


----------



## Greenears

rrod said:


> Here's one:
> http://www.drewdaniels.com/audible.pdf


 
 Thank you RRod for actually providing a link, as compared to Castle's long-winded hand-waving at mountains of evidence.....  Surely I jest Castle, your input has been valuable RRod just has a certain way with posts....    
  
 I had seen reference to this paper but never were able to see the content of it until now.  I read it with apt attention.
  
 My comments:  I like the general test setup.  I think they did a good job of proving for their listening setup (state of the art at the time) and for the SACD/DVDA programming available at the time of testing in 2007 that the material could just as well been recorded on Redbook since it was not distinguishable by a large number of test subjects.
  
 The shortcomings I think are that this needs to be updated for 2014 to test 192/24 FLAC, and I would have liked to seen a list of the material tested.  There have been advances in DAC design in the last 7 years, with relatively inexpensive 24 bit DACs multi-segment SD architectures with amazing 130 dB performance.  I doubt this would have been in the equipment tested at the time.
  
 To me there were two very interesting notes in the paper.  In section 1 they note there was no published paper testing SACD vs Redbook up until then.  Which is amazing considering that high res formats had already been going for a decade.  This is my chief complaint: Lots of people today claim "there are mountains of testing" but I never get to see the notes of the actual test.  I have looked.  The second even more interesting thing is in section 4.  They point out that most of the SACD/DVDA "sounded better" than a CD, even through the CD loop.  They actually say these recordings should be released on Redbook!  This is something I've already posted about, that it is all in the mastering. But it is very hard to get information on the mastering and mixing of any tracks let alone 24 bit.


----------



## bigshot

If it's transparent at 44.1/16 it doesn't matter how high you go, it's still going to be transparent. No point reinventing wheels. Better to focus on things that actually make sound better.


----------



## Greenears

bigshot said:


> If it's transparent at 44.1/16 it doesn't matter how high you go, it's still going to be transparent. No point reinventing wheels. Better to focus on things that actually make sound better.


 
 I don't agree.  This discussion has been had with tape, whether redbook is enough for tape.  So let's say pro tape is your source at 70 dB and you loop in redbook codec at 96 db.  Testing yields transparent.
  
 Now you come with some new system at 130 dB source, maybe it becomes detectable? It's hard to know the end-to-end ADC-to-DAC full system resolution of the SACD/DVDA played (in part because they weren't listed in the paper).  What if they were just upsampled redbook in reality?
  
 So you need to read the paper carefully because they didn't make sweeping claims.  Their point was the SACDs and DVDAs tested could just have well been recorded on Redbook.  And as you see, I granted their point in my first sentence.  And for that I think it's a good and useful test.
  
 Someone simply needs to redo a similar test on 192/24 and publish it.  With the state-of-the art recording done with current best-in-class ADC and DAC.  If nobody can find a recording that's detectable then that tells you something.  Pono is going to have some (or already has some) 24 bit - try it.  I can't get at that site.


----------



## bigshot

The problem is that you are looking at numbers and specs as abstract things, not representations of *sound* that people can or can't hear. You are entirely focused on the specs of sound reproduction formats, but you haven't done any research into the specs of audibility thresholds for human hearing. Until you put the numbers into context with what your 100% human ears can actually hear, you will keep chasing down the rabbit hole of "bigger numbers equals better".
  
 For instance, do you know the dynamic range that humans can hear *in music*?


----------



## Greenears

bigshot said:


> The problem is that you are looking at numbers and specs as abstract things, not representations of *sound* that people can or can't hear. You are entirely focused on the specs of sound reproduction formats, but you haven't done any research into the specs of audibility thresholds for human hearing. Until you put the numbers into context with what your 100% human ears can actually hear, you will keep chasing down the rabbit hole of "bigger numbers equals better".
> 
> For instance, do you know the dynamic range that humans can hear *in music*?


 
 Is this directed at me?  I have already posted at length that I am well aware that 100 db SPL is the sound of a jackhammer at 1m, and that 10 dbSPL is a mosquito in the corner of a sound proofed room.  So yes, I am well aware that humans are not supposed to be able to hear more than 100 dB without pain.  If that is your question. 
  
 So if we feed a 130 dB source and our ears limit to 100 dB and pass it though a prallel 100 dB redbook codec the ABX should fail on a large sample and large number of ear-pairs.  
  
 So where is the link to that test?
  
 If I could find a convincing study with updated equipment I would run away.  The only thing I found so far was a guy on Anand tech who in the last couple yeasr (I think it was last year) did some ABX'ing on DACs.  It was a great test, I learned stuff from it, but it didn't fully answer the format question and that wasn't what he was investigating.  I have seen a number of others but they all were flawed for the question I was asking, even if they were perfect for the question they were asking.
  
 I (and others) are mystified why this is so hard to answer.


----------



## bigshot

Our ears can only hear about 40-50dB of dynamic range at a time. Any more than that and our ears need a few minutes to adjust to the different volume level. That means that if you are listening to music with peaks normalized in redbook, the whole bottom 50dB or so is completely inaudible. Not only that, the room you are in right now has a noise floor of at least 30dB. Anything in the recording below that is going to be masked by the room tone anyway. Dynamic range in recordings extends *downwards* not upwards. Peak level is peak level no matter what format you are listening to. So if you are playing a high bit depth recording at a volume of 90dB, which is plenty loud, it is going to sound EXACTLY the same as a redbook recording at 90dB, because you are only hearing the top 40-50dB anyway.
  
 You can go ahead and call for tests, but I don't see why anyone should want to waste their time or money on conducting listening tests on things that are clearly inaudible. You might as well do tests to find out if people can see X rays or feel the earth rotating on its axis.
  
 I'm not trying to be argumentative here. I'm just pointing out that human hearing has its limits, and unless you fully understand what humans can and can't hear, you are going to waste a whole lot of time chasing down things that don't make a bit of difference. If you know how hearing works and how sound reproduction works, you can focus on the things that actually makes music sound better.
  
 If you are just interested in theory, and not practice, feel free to ignore my comments. It's fine to play intellectual games if you enjoy it. But it won't make your stereo sound any better.


----------



## Greenears

bigshot said:


> Our ears can only hear about 40-50dB of dynamic range at a time.


 
 Where is the study/paper that proves that by experiment? It's so hard to find this stuff.


----------



## Greenears

Alright, the good and the ugly.  I skipped over bad completely.
  
 Back to the clarinet, I market just 3 seconds up to that 6th high note.  I've had success there before. I skipped all other sections.
  
 I took more time listening to ABXY until I felt sure. Volume was medium to low - I found higher volume fatiguing.  I would say it was equivalent to listening to a live clarinet played medium loud about 6-10 ft away. I'm listening for a certain roughness or natural distortion like you get playing in a wood-floor room right at the front end of that note. I'm NOT hearing or listening for noise.  Note the streak of 7 right out the box.  Note all my other runs on this track are all way below 50%.  If I'm really guessing I'd expect to see mean reversion and some time over 50% and also some long streaks of wrong guesses.
  
 Cassandra on the other hand eluded me yesterday.  I did that one with "look at results" unchecked.  The runs before were all over the place, but if I hit 6% I swung to 75% pretty quickly.  I'm guessing on Cassandra. I've never really found a note to latch onto. 
  
 For test192 (clarinet) the thing that impressed me is I never had a losing streak of more than 2 (only 3 times). Yet I had a winning streak of 7, and the other day I think 5 and some others.  I really should see the 5+ losing after so many trials if guessing.  Mean reversion. Fatigue is definitely a problem.  You hear it then listen more and your mind switches it around on you. It's there then switched.  But this time I paused and took time when the switch happens, longer trials I'm battling. I only committed when i felt sure.
  
 Conclusion? Hmmm. I'm 80% sure I'm not guessing on Vivaldi and Clarinet.  The effect I hear is very subtle and susceptible to fatigue.  But there is something audible.
  
 foo_abx 1.3.4 report
 foobar2000 v1.3.6
 2014/12/17 21:51:24
 File A: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\test192.flac
 File B: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\test192_16.flac
 21:51:24 : Test started.
 21:54:00 : 01/01  50.0%
 21:54:44 : 02/02  25.0%
 21:55:43 : 03/03  12.5%
 21:56:20 : 04/04  6.3%
 21:57:19 : 05/05  3.1%
 21:58:47 : 06/06  1.6%
 22:00:28 : 07/07  0.8%
 22:02:40 : 07/08  3.5%
 22:09:00 : 07/09  9.0%
 22:09:51 : 08/10  5.5%
 22:11:48 : 08/11  11.3%
 22:14:34 : 09/12  7.3%
 22:15:08 : 10/13  4.6%
 22:16:53 : 10/14  9.0%
 22:17:27 : 10/15  15.1%
 22:19:06 : 11/16  10.5%
 22:19:44 : 11/17  16.6%
 22:21:06 : 12/18  11.9%
 22:23:27 : 13/19  8.4%
 22:24:13 : 13/20  13.2%
 22:24:38 : 14/21  9.5%
 22:25:24 : 15/22  6.7%
 22:26:35 : 16/23  4.7%
 22:28:02 : 16/24  7.6%
 22:29:08 : 16/25  11.5%
 22:29:35 : 17/26  8.4%
 22:30:09 : 18/27  6.1%
 22:30:57 : 18/28  9.2%
 22:33:43 : 18/29  13.2%
 22:34:14 : 19/30  10.0%
 22:34:43 : 20/31  7.5%
 22:35:29 : 20/32  10.8%
 22:36:07 : Test finished.
  ---------- 
 Total: 20/32 (10.8%)
  
 foo_abx 1.3.4 report
 foobar2000 v1.3.6
 2014/12/16 21:05:07
 File A: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\08-Another Country.flac
 File B: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\cassandra16.flac
 21:05:07 : Test started.
 21:07:24 : 01/01  50.0%
 21:08:32 : 01/02  75.0%
 21:09:11 : 01/03  87.5%
 21:09:50 : 02/04  68.8%
 21:10:35 : 02/05  81.3%
 21:11:15 : 03/06  65.6%
 21:12:00 : 03/07  77.3%
 21:12:42 : 04/08  63.7%
 21:13:24 : 04/09  74.6%
 21:14:14 : 05/10  62.3%
 21:14:52 : 05/11  72.6%
 21:15:32 : 06/12  61.3%
 21:16:14 : 06/13  70.9%
 21:16:52 : 07/14  60.5%
 21:17:27 : 08/15  50.0%
 21:18:02 : 08/16  59.8%
 21:18:57 : 09/17  50.0%
 21:19:33 : Test finished.
  ---------- 
 Total: 9/17 (50.0%)


----------



## adisib

10% generally isn't considered conclusive, and HDTracks is known to have audibly different masters for the higher resolution files. Or did you resample it yourself?


----------



## Greenears

adisib said:


> 10% generally isn't considered conclusive, and HDTracks is known to have audibly different masters for the higher resolution files. Or did you resample it yourself?


 
 Scroll up ... 15 or 20 pages haha.  I resampled the 24 using SoX -16.  Resampling is a must, I've seen a number of 16 v 24 ABXs but as soon as I poked my nose in to learn about the files it became clear they were different masters.  
  
 My HDtracks sampler didn't give me 16 bit versions. I guess I would have to try a paid version? Someone actually recommended Random Access Memories. 
  
 The funny part would be if the HD Sampler is actually just upsampled 16 and that is why I'm having trouble ... or if I do hear something it is an artifact from the 44-192 conversion.  
  
 But note today I'm having the most success with the Linn track.  I guess I could try their 16 vs my 16 for grins.
  
 I agree 10% isn't conclusive, but the overall experience is hard to totally dismiss.  It's particularly the lack of mean reversion on that track that puzzles me.  If I saw 23/30 I would consider that conclusive, but I never have.


----------



## RRod

greenears said:


> Scroll up ... 15 or 20 pages haha.  I resampled the 24 using SoX -16.  Resampling is a must, I've seen a number of 16 v 24 ABXs but as soon as I poked my nose in to learn about the files it became clear they were different masters.
> 
> My HDtracks sampler didn't give me 16 bit versions. I guess I would have to try a paid version? Someone actually recommended Random Access Memories.
> 
> ...


 
  
 Look back at my spectrogram of the difference file from the clarinet track. With a proper conversion that is all you should be hearing different: noise at around -100dB. Perhaps you should look at such a difference for your files. Also, don't get too caught up in your runs. I made a random draw of 500 Bernoulli trials (p=0.5), and got the following runs:

```
1  2  3  4  5  6  7  8 11   0 55 34 15 12  4  2  1  0  0   1 63 34 14  7  1  1  2  1  1
```
  
 So if you happen to get the 8 and 11 runs early on, you might think "hey why am I not reverting to the mean", but it's still really just luck.


----------



## castleofargh

castleofargh said:


> greenears said:
> 
> 
> > rrod said:
> ...


 
  
 mea culpa. I had at least one other big ass trial in head(and a few small 2 or 3 people stuff) and was pretty confident, it so happens that it was the stuff about stradivarius that I seem to have mixed up somehow in my sad and poor memory... so there seem to be no other trial done with big number of participants and exhaustive controls. sorry about that. I hate it when people claim false stuff, so I'm slightly mad at myself right now. but then again, I'm expecting to be wrong from time to time so I accept the idea readily.
 now I still believe there is a general... maybe not consensus because that doesn't ever exist in the audiophile world, but close enough, about what bit depth and sample rate do and don't do to a track. and it's usually conclusive with measurements of noise and disto. so the mystery isn't really one.
  
 @ greenears, I also think that you're still dismissing the one and only logical reason for a difference. maybe I'm on a mistake streak, but to me without dither the only thing you do when changing bit depth(admitting that it doesn't get low enough to crop the audio signal recorded), is adding or removing zeroes to the sample word length. so I even have a hard time getting how a software could do that wrong. and even if it did, that would still have zero impact on the first 16bit of sound. meaning that you're noticing something at -96db or whatever error we can get from the LSB.
 we're really factually only changing the voltage values/loudness of quantization errors (unless the DAC actually uses a really different way to deal with 16 and 24bit? but that isn't a concern with foobar's abx as both files will end up with the same bit depth). so the very first thing below -96db that you'll notice is and will be the noise you decided to dismiss(plz anybody if I'm saying something wrong tell me).
  what is your hypothesis is what I'm asking I guess?
  
 that's only the third time I mention it, but did you try with added dither to see if it becomes less easy to abx or not? if the noise is involved, shaping it should have an impact(at same loudness for the tests).
 did you try turning the 16bit back to 24bit to abx 2tracks with the same resolution? so we know for sure that it's not the abx pluging failing at one of the most simple task ever when going 32bit pcm. I really don't believe it could be an issue, but if we're looking for the improbable, why not there?
 for the tracks where you have better results, did you check if the song peaks were far from 0db and thus helping to rise the noise floor. this was already suggested with normalization by others. its very obvious that a track with peak loudness at -20db will make the noise more likely to be heard than some track reaching 0db often.
  
 except repeating abx on the same basis, that can only really hope to improve the statistic significance if you decide to add all the results, are you willing to try zeroing in on the possible cause of difference?
 all in all what are you looking for with those abx?
  
  


greenears said:


> bigshot said:
> 
> 
> > Our ears can only hear about 40-50dB of dynamic range at a time.
> ...


 
 Steve Eddie talked about like 100db range for sounds with a significant delay between listening to both extremes(don't know if it's like say, 20 and then 120db in a normal environment, or if it is in an anechoic chamber). and 60db for instantaneous dynamic.
 lossy formats considered "transparent" seem to have no difference above -60db


 I know I pass mp3 192 with only a little effort so I find the 60db idea consistent(I fail mp3 and aac @320). I also tried adding tones and musics at different loudness inside other songs and check if I could hear them. and decided the -80db was a very very safe limit for me to stop carring with my listening volume(obviously the song weren't all stuck to 0db).
  
 then there is the fact that we can only go so loud before it's hurtful to us, and that the ambient noise in a room will probably be at least 30 or 40db, I don't remember for our own body noise but I think it was about 10 to 20db(really not sure about that one).
 and all in all albums rarely pass 60db anyway(because it's our limit? IDK, chicken or the egg kind of problem probably).


----------



## bigshot

greenears said:


> Where is the study/paper that proves that by experiment? It's so hard to find this stuff.


 

 It's easy to find. Just google "human hearing thresholds of perception" and add the particular aspect of sound that you are searching for.
  
 Here is info on how the ear adjusts to louder sounds. http://hyperphysics.phy-astr.gsu.edu/hbase/sound/protect.html#c1


----------



## Greenears

@RRod
  
 Chance of long runs of success:  
 As I said a few pages back I'm well familiar with binomial distribution and I know that with increasing number of trials the chance of seeing any given run length increases asymptotically towards 100%.  But my total runs on test192 are maybe high double digits, and I gravitate towards 10% hitting 5% and 1% in places.  And the better success (in both tracks) corresponds to tracks where I heard a specific note.  Remember there are other tracks where I couldn't find anything to latch onto and I did much worse.  To say it could still be chance is always true - anything can always be chance.  But at this point I am starting to believe I may be hearing something.
  
@castleofargh
  
 Lack of previous tests:
 I'm glad the error of your ways you have learned, my young padawan.  (insert maniacal laugh).   But seriously, THIS is the biggest problem.  I (and others) have posted similar questions of various forums of knowledgeable people.  Predictably the same thing always happens, the answers quickly sort out into the two camps.   The camps agree on absolutely nothing, EXCEPT that they are both so sure of their position they consider testing is a waste of time and any test that is contrary to their position must be flawed.  So this is IS a Kobayashi Maru situation for the tester (that was the best reply to my whole thread IMO).  It's exactly like the expectation of listening - people that read a test result once a while back remember it's conclusions the way they want.  If you go back and actually read the test results, it didn't actually test for that conclusion.  I have looked hard and challenged many, and I am coming to believe there is no 24 v 16 test that has proper results posted.  That is a problem for the industry.
  
 No dither:
 The reason for no dither is simply testing time. I wanted to pass the first one and then make it harder.  But the first one was hard enough. I'm going to take a two week hiatus from testing and posting, then I'll be back and probably try dither.  Frankly I don't think I stand a chance with dither added, but I need to buck up my courage.
  
 Noise:
 I have said repeatedly, but people refuse to believe me, I am not rejecting any of the noise theories.  I am well versed in the theory of noise, I know it is around 100 dB etc etc.  I'm not willfully ignoring it - I just don't hear it.  I don't hear it on CDs I don't hear it on MP3s and certainly not on any of these HD tracks, even with the volume up.  I'm using headphones and there is a limit to how loud I am comfortable with.  
  
 So what am I hearing?
 That's a 64 bit question.  I don't know, I think there is scant research on the very edges of audible quantization errors.  I don't know why, but I suspect for communications research in ADC they experiment with shaping the noise in different ways to get a better bit error rate, which is relatively easy to measure.  Although a lot of the techniques developed for Comms get used in Audio, it's not the same end game.  For audio my impression is that most of the research dollars in the 90s was in the psychoacoustic models for compression.  To test that, you put 100 average subjects with average or typical consumer listening situations and fiddle the model until they are can't discern.  It's not exactly about finding the boundary in the best uncompressed conditions.  For people making high end equipment, they all need their secret voodoo sauce or marketing and they have scant interest in funding tests that may prove there is no voodoo and no sauce.
  
 So still what am I hearing?
 My best guess lies in a misconception that many have about quantization noise.  Please open any standard EE textbook on signals and systems.  The first thing you will read is that Quantization errors are a non-linear process and cannot be completely analyzed mathematically.  The ~6db per bit idea (which is where you get your 100 dB and 60 dB) is an _approximation_.  I'm not making this up it says it right there in the textbook.  I saw a U of Waterloo paper (google) that had a good intro summarizing Q noise, google it.  The analogy is similar to FM and AM radio.  In EE you learn how to completely analyze AM using Fourier and Laplace.  Every detail can be described by nice equations with precise answers.  Then the next thing you learn is that FM is a non-linear process that has no equivalent equation.  But it sounds better.  This revelation is very frustrating to young padawans, but you get over it after a few weeks.  A few tricks and approximations and maybe computer simulations are used to analyze FM to enough extent to be able to use it.  Same with Q noise - and actually it has some similarities to FM with repeated short spikes throughout the spectrum.
  
 FFT:
 You also can't say conclusively you looked at the FFT and didn't see anything.  While I agree it's true that you aren't going to miss some 50 dB spike, there are limitations with FFT.  Signals move in time, FFT is a slice in time.  To convert between domains you need a window like Hann or Blackman and the windows have artifacts.  I think everyone that has worked with this stuff hands on knows this.  
  
 Possible Theory:
 So remember that a frequency shift and phase shift are the same thing (while they're shifting).  My suspicion is that human hearing is incredibly attuned to minute frequency differences, which make up what we call "tone".  Note how well we pick out peoples voices, or a Stradavarius, or a Gibson.  I'm sure you can pick out Mick Jagger or Bono or Bruce Springsteen in the first syllable.  No tones are pure, they all have distortion and we can pick out the slight differences in the higher order harmonics.  It may be that at some resonant frequencies the quantization introduces just enough frequency shift that you can detect it, or mucks with the relative amplitude of certain harmonics.  
  
 Conclusions:
 Well as I said I'm taking a 2 week hiatus but I'll be back with my phasers fully energized to take one last dig at this. I only found one poster that did similar tests, and also inspired me to try.  His results seem to match mine.  He also said it was very very hard to discern.  He could only detect on one of his headphones, the other was pure guess. I may be limited in equipment.  I've also thought about other possibilities.  The first is that the sample tracks weren't recorded in 24 bit, they were upsampled from 16. That would account for a fail, but not a pass, unless I'm hearing artifacts of the upsampling.  Since I am well familiar with upsampling algorithms, I think that is even less likely to be audible than 16 bits, but not non-zero. The other possibility is simply a minor bug or flaw in the 16 to 24 upsampling (if they did that) - also very remotely unlikely but probably more likely than the other theories.  Or a bug or resonance point in the multi-segment converter in the DAC that is triggered by either the 16 or the 24.  Probably more likely than the rest, but till very unlikely to be audible. 
  
 I would very much like to conclude that 24 bit is hooey and move on. However four big things give me pause:  (a) lack of solid testing data published on this problem, and other anecdotes similar to mine (b) Engineers and architects at TI, Cirrus/ESS, and Wolfson are not idiots. I've met some of these people they are very serious ridiculously smart and educated people.  Why would they all invest so heavily in 24 bit architectures for the last N years and push performance another 30 dB past 100, if there was absolutely no use for it whatsoever.  It could be Marketing, I know, but I still pause.  I've read their papers they seem to be trying seriously to make the DAC and ADC better.  And they are amazing already.  (c) People inside Dolby in the 90s, who had no dog in this particular hunt at the time, told me the be all and end all was somewhere in the 18-20 bit range.  This was when Dolby AC3 was being commercialized for DVDs.  Dolby AC3 on DVD is 16 bit, but you may not know there were 18 and 20 bit AC3 options that were only available on the Pro equipment that Dolby made for the theaters and studios.  A friend of mine left Dolby and joined me at another company and we talked about it later and it seems they were doing serious science it wasn't fluff. (d) why would anyone bother with dither if 96 dB was 30 dB too much?  This one doesn't seem to be marketing because there is not "dither sticker" on the label of a CD.
  
 I also note that apparently Neil Young posted on the Pono website in the last few days that Warner Brothers has a huge catalog of 24 bit material and he is pushing them to release it.  I don't know anything more than that, but if true, 24 bit mania may be upon us soon.  It would be nice to know the answer 
  
 I take the hiatus with the tentative guesstimate conclusion that the be-all and end-all is somewhere in the 18 bit ENOB area, give or take a bit.  Well implemented dither may get you some of the way there, and dither may get you to the point that detection will require such an exotic combination of gear and ears as to render it such that 1 in 1000 people can detect so it probably becomes moot.  Certainly a good recording trumps all of this by 100 country miles.


----------



## bigshot

I'm beginning to think that you guys enjoy scientific testing better than listening to music.


----------



## Stereodude

He's going to prove all of you wrong.  Just you wait.


----------



## RRod

greenears said:


> @RRod
> 
> Chance of long runs of success:
> As I said a few pages back I'm well familiar with binomial distribution and I know that with increasing number of trials the chance of seeing any given run length increases asymptotically towards 100%.  But my total runs on test192 are maybe high double digits, and I gravitate towards 10% hitting 5% and 1% in places.  And the better success (in both tracks) corresponds to tracks where I heard a specific note.  Remember there are other tracks where I couldn't find anything to latch onto and I did much worse.  To say it could still be chance is always true - anything can always be chance.  But at this point I am starting to believe I may be hearing something.


 
  
 We'll just see how things go with your reboot. Kudos for putting so much time into it. My only suggestion is to come back with a fixed statistical goal in mind, and just go with it.
  
 As far as FFT, yes it is limited, but I'm willing to bet I could futz around with all the window and length options in Sox and still not get anything to pop out (I'll give it a try later). As far as the industry: it's always best for the people at the start of the chain to have the best specs, because anything they introduce is carried along the whole chain and can build up into actual audible error; that doesn't mean the end-user needs the same precision.
  
 TBH, I'd be fine if we just accepted 24/176.4 or whatever as a new standard. The problem is that this wouldn't solve anything, because there will always be people who believe in "more is always better" with money to spend. Where would it stop? The second we'd standardize 32/352.8, some corporate-lackey researcher will come out with something about how MHz signals cause a little signal to pop up on a brain scan, and then the horses are off again: "We need 64/5.6448MHz and a flux capacitor to enjoy our rock master-tapes made in 1964!" Meanwhile guys like bigshot are sitting like the Maxwell man in front of properly set-up 5.1 x 16/44.1 systems and having a grand old time, instead of wondering if he accidentally just sat on his clip-on headphone pico-tweeter. (rant over)


----------



## bigshot

I think instead of chasing better numbers, it's a good idea to chase down better sound. I'm always astonished to find people who have spent tens of thousands of dollars and spent years of their life working on improving numbers on a sheet of paper, and yet they haven't taken the most fundamental steps yet to improve the sound of their systems.


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## bigshot

I've heard audiophiles who collect SACDs and order from HD Tracks wax rhapsodic about the sound of Donald Fagen's "The Nightly". Whenever you ask audiophiles what their reference recording is, it's usually near the top of the list. I agree. It actually *is* one of the best sounding rock albums ever made. Audiophiles pay high prices for 180 gram audiophile vinyl copies and Japanese SACDs... ...but it was recorded in the early days of digital audio at 16/48.


----------



## castleofargh

Spoiler: Warning: Spoiler!






> Lack of previous tests:
> I'm glad the error of your ways you have learned, my young padawan.  (insert maniacal laugh).   But seriously, THIS is the biggest problem.  I (and others) have posted similar questions of various forums of knowledgeable people.  Predictably the same thing always happens, the answers quickly sort out into the two camps.   The camps agree on absolutely nothing, EXCEPT that they are both so sure of their position they consider testing is a waste of time and any test that is contrary to their position must be flawed.  So this is IS a Kobayashi Maru situation for the tester (that was the best reply to my whole thread IMO).  It's exactly like the expectation of listening - people that read a test result once a while back remember it's conclusions the way they want.  If you go back and actually read the test results, it didn't actually test for that conclusion.  I have looked hard and challenged many, and I am coming to believe there is no 24 v 16 test that has proper results posted.  That is a problem for the industry.
> 
> No dither:
> ...


 
  


  
  
  
  
  
  
 I don't even need 2 weeks to answer, /me so strong.
  
 no major test for hires vs 16bit is certainly not a problem for the industry ^_^. it's only one for us and they probably have done all they could to keep it that way. (conspiracy theory !!!!!!!!!!)
  
 I'm pushing for dither because it the one simple way to lower the perceptual noise while not changing the bit depth. so if it's enough to significantly do worst in the abx, then for me it's proof that the 16bit noise was indeed responsible for the good answers. at least that's how I see it. I guess hearing a difference can be easy, and being able to pinpoint the cause a lot less so. I know I'm stuck on noise floor, but I just don't see anything else. 
  
 for DACs and quantization, the only model I clearly understand is with good old R2R ladder. with sigma delta I pretend the basics are still the same and do as if the pulse modulated signal is just a good old smooth stuff ^_^. pusle modulation to me is just another noise on the output and on a much lower level than quantization so who cares. anyway if the principles of a R2R can't be applied then it's beyond my ability to make a model. I'm limited to each new bit opens a path with half the voltage, that way all the bits below never add up to the total voltage of the bit just above and I can justify why adding bits only add lower sounds and lower noises.
 to me the fact that we can't really make a model of quantization is because the noise values come from manufacturing precision issues (true or not IDK, but that's how I self explained it). so when you go back to my R2R model, the resistor values can only be precise to a point, each being just very slightly different from the other, and while the error doesn't matter in regard to the resistance value, it may add up (thanks to being still my R2R ladder), and have significant value for the least significant bits.
  
 anyway error on a wave is just noise, and voltage is loudness so error of small voltage values are quiet noises. thinking like that it explains in a very basic way why the louder sounds are still perfect when the output signal has more errors. because it's a wave and they add up the way you add 2 instruments on a record. they don't mix into another thing, they just play at the same time. the perfect music up to -96db, and the quantization noise from whatever errors at 16bit. and the same way going to 24bit can only ever help adding sounds below -96db because the louder sounds are already what they are. be it music or noise it's the same and that's why we keep saying that adding bits to the album is useless if we agree that we already can't hear -96db.
  
 this is what I came up with when trying to write some layman digital stuff in french last year. I don't even know how much is true, but the model works with everything I've learned in digital audio up till now and people don't have to know about Fourier, 1st or 2nd degree equations, or really understand any of what is done in a DAC(or ADC). but again I have zero idea if something like a sabre chip still has a part where the signal is dealt with in a R2R way. I just suppose so.
 trying to make oversimple models like that would probably get me excommunicated, burned and then have my ashes locked up on several continents if I posted this on hydrogen audio ^_^.
 last time I heard about Laplace, it was in a gundam anime(seriously). that's how much math has helped me throughout in my real life.
  
  
 I agree with amplitude and phase being somehow linked because it's a wave, but I can't see why bit depth should affect any of them.
  
 and why have DACs in 24bit? well that's because 16bit DACs can't do 16bit ^_^. to me it's more than enough a reason. and for the manufacturers, I would think that having a 32bit DAC is a great solution to have volume control implemented in the DAC chip for anything portable at least. or DSP/EQ and anything where using the strict limit of bits would risk clipping the signal.
  
 and warner or sony or anybody are in need to get new products and new ways to control those products to get money of them. one way is remastering, but they did so much crap that remaster now rhymes with cheap romanian remaster and lesser quality in most consumers mind(the low fi ones at least). the other way is to come up with the craziest support. and what cries out upgrade and quality louder than going mad over resolution numbers?
 I have zero hope that they're doing this for the sake of sound quality, we all know how much they care about sound quality with autotune, loudness war, and some techno beats added to any ok modern song so that it can get a chance to pass in clubs. they are marketing and only marketing.
 now if 24/96 end up being the common standard, I really couldn't care less and would use that. but we all know that when we'll be there, they will throw at us some new mega much better support than last time, that was already 110% pure fidelity. and once again you'll have to buy a new gear to read it, and it will be hard to copy for some time, etc. history repeating.
  
 I'm the optimistic kind.


----------



## Greenears

bigshot said:


> I'm beginning to think that you guys enjoy scientific testing better than listening to music.



Hmmmm fifty fifty.  Actually it was more fun and I learned more about the enjoyment of music than I thought I would. Anyone that has ever cared about sound quality or equipment should ABX at least once.


----------



## sonitus mirus

greenears said:


> Hmmmm fifty fifty.  Actually it was more fun and I learned more about the enjoyment of music than I thought I would. *Anyone that has ever cared about sound quality or equipment should ABX at least once.*


 
  
 Yes, already had my mind blown.
  
 Between 16-bit AAC 256 or Lame 320 and a lossless version, any ABX logs that seem to suggest someone can tell a difference are often only with a few sections of some songs, or an instrument playing a certain note or passage. Even then, few people ever get it perfect, with some misses on occasion, despite extremely critical listening and the ability to quickly change from one file to another.   I'm convinced that only a smattering few individuals would ever be able to actually tell that the source of their music was coming from a well-encoded lossy format in a normal listening environment. 
  
 I'm an audiophile.  I listen to lossy music files with capable, yet inexpensive DACs and amplifiers, and I expect to hear the best sound quality.


----------



## bigshot

sonitus mirus said:


> I'm an audiophile.  I listen to lossy music files with capable, yet inexpensive DACs and amplifiers, and I expect to hear the best sound quality.


 
  
 Exactly!


----------



## Greenears

sonitus mirus said:


> Yes, already had my mind blown.
> 
> Between 16-bit AAC 256 or Lame 320 and a lossless version, any ABX logs that seem to suggest someone can tell a difference are often only with a few sections of some songs, or an instrument playing a certain note or passage. Even then, few people ever get it perfect, with some misses on occasion, despite extremely critical listening and the ability to quickly change from one file to another.   I'm convinced that only a smattering few individuals would ever be able to actually tell that the source of their music was coming from a well-encoded lossy format in a normal listening environment.
> 
> I'm an audiophile.  I listen to lossy music files with capable, yet inexpensive DACs and amplifiers, and I expect to hear the best sound quality.




Yup no surprise. I did some crude AB testing a decade ago which opened my eyes on compression. But I'll probably give it a serious retry in January. My hunch says something like 330 kbps vbr 18 bit is the last audio codec that's ever needed. Subject to downward revision.


----------



## adisib

greenears said:


> My hunch says something like 330 kbps vbr 18 bit is the last audio codec that's ever needed. Subject to downward revision.


 
  
 There isn't really a bit depth in the same sense for lossy compression. To quote wikipedia,
  


> Bit depth is only meaningful in reference to a PCM digital signal. Non-PCM formats, such as lossy compression formats like MP3, AAC andVorbis, do not have associated bit depths. For example, in MP3, quantization is performed on PCM samples that have been transformed into the frequency domain.


 
  
 Then 320 kbps vbr in a high quality codec is all that is ever needed, for listening that is. There are still other reasons to have lossless files.


----------



## jcx

I believe I've seen the claim that lossy codecs use ~7 bits per critical band - but don't know how much more goes into the full calculation - is that just the mantissa in a float?, what's the typical fft length vs bit rate...


----------



## Greenears

Traditionally the algorithms were floating point and convert to fixed point at the output.


----------



## Greenears

Has anyone actually read this paper?
  
 http://www.aes.org/events/137/papers/?ID=4058
  
 Seems like they are supporting my result but it's  convoluted.  
  
 I also saw somewhere to another paper that found 24 bit could be heard, but I wasn't able to get back to the reference and find it.


----------



## RRod

greenears said:


> Has anyone actually read this paper?
> 
> http://www.aes.org/events/137/papers/?ID=4058
> 
> ...


 
  
 All three authors from Merdian, and who knows what kind of vetting convention papers go through. But I may spend the 20 large and see what the deal is.
  
 See also:
 http://www.hydrogenaud.io/forums/index.php?showtopic=107124&st=0


----------



## sonitus mirus

rrod said:


> All three authors from Merdian, and who knows what kind of vetting convention papers go through. But I may spend the 20 large and see what the deal is.
> 
> See also:
> http://www.hydrogenaud.io/forums/index.php?showtopic=107124&st=0


 
  
  
 Save your money.  Been discussed at HA recently.  
  
 http://www.hydrogenaud.io/forums/index.php?showtopic=107124


----------



## RRod

sonitus mirus said:


> Save your money.  Been discussed at HA recently.
> 
> http://www.hydrogenaud.io/forums/index.php?showtopic=107124


 
  
 Yeah sorry, I edited in that very link after a bit of Googling. Thanks!


----------



## castleofargh

yup it was pretty interesting topic, even if you have to swim between the usual arny vs amirm fights.
 from the little parts and indications from those who bought the paper, it's crap. they played the game of filtering at 16/44, used the only one dither method everybody said was bad for many years. had a very dynamic track obviously, with several moments with room sound and no music played. and it was with peaks around 100db or something.
  
 so the trial was pretty much to know if we could identify 16/44 when everything was done wrong on a specific track. just the idea of filtering at 16/44 is such a joke. anyway, it's not the paper that will change the world, that much seems clear.


----------



## RRod

castleofargh said:


> yup it was pretty interesting topic, even if you have to swim between the usual arny vs amirm fights.
> from the little parts and indications from those who bought the paper, it's crap. they played the game of filtering at 16/44, used the only one dither method everybody said was bad for many years. had a very dynamic track obviously, with several moments with room sound and no music played. and it was with peaks around 100db or something.
> 
> so the trial was pretty much to know if we could identify 16/44 when everything was done wrong on a specific track. just the idea of filtering at 16/44 is such a joke. anyway, it's not the paper that will change the world, that much seems clear.


 
  
 And I'm sure the quietest musical sound of the ensemble was well above the room noise. I mean, seriously, having played in them and listened to them live many times, string quartets aren't what should push 16 bits.


----------



## castleofargh

rrod said:


> castleofargh said:
> 
> 
> > yup it was pretty interesting topic, even if you have to swim between the usual arny vs amirm fights.
> ...


 

 maybe they had big strong fingers? ^_^.
 but sure I also don't expect the track to be extraordinary by itself. they say what it is somewhere but I can't recall.


----------



## Greenears

I read the other forums and it's a lot of back and forth about whether it was peer reviewed, very little meat.  I think few (none?) of the posters had actually read the paper.
  
 Why on earth are they filtering?  BTW, I already looked in-depth at the anti-alias filters used in modern 24 bit DACs last week.  The are rock solid.  Bottom line - with a 512 tap linear phase FIR which nowadays is nothing to implement in 180 or 65nm, the filters are rock solid.  Less than 0.1 dB passband ripple then falls off a cliff at 0.45 Fs.  No chance of any audible aliasing in the stop band (I think it was -130 dB or better).  
  
 Anyhow, moving on.  One last try:
  
 Some guy posted these files and a log showing 9/10 on Foobar ABX.  So I just did my regular SoX conversion 24 to 16 bit, still no dither (sorry Castle) and here we go again.  Still tantalizingly close.  Not as good as some other runs, since it shows some signs of mean reversion and longer runs of wrong guesses.  But still, after 40 trials, I'd at least like to be in double digits to be sure.  He had better phones so maybe that is the difference.  Sigh.  I would really like to see a conclusive test.  Maybe a selection of 5 top rated phones, 5 DACs covering all three 24-bit DAC makers and one or two computer DACs like Realtek, and maybe 100 listeners.  Best to record some content fresh to know the provenance.  It's a bit of work but not a major thing for pros, compared to all the dollars and effort put into marketing 24 bit audio.  
  
 If anyone gets near 1% please post.

http://www.nordicaudiolabs.com/samples/24.wav
http://www.nordicaudiolabs.com/samples/16-hpt.wav

EDIT: Another 16 bit version made with Ditherbox (Airwindows Audio Unit Plugins) uploaded here for easy access:

http://www.nordicaudiolabs.com/samples/16-avd.wav

  
 foo_abx 1.3.4 report
 foobar2000 v1.3.6
 2014/12/19 10:31:23
 File A: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\24.wav
 File B: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\24to16.wav
 10:31:23 : Test started.
 10:35:41 : 01/01  50.0%
 10:36:32 : 02/02  25.0%
 10:38:02 : 02/03  50.0%
 10:38:43 : 02/04  68.8%
 10:39:00 : 02/05  81.3%
 10:42:01 : 03/06  65.6%
 10:42:26 : 04/07  50.0%
 10:42:58 : 04/08  63.7%
 10:43:37 : 04/09  74.6%
 10:45:42 : 05/10  62.3%
 10:46:54 : 06/11  50.0%
 10:47:28 : 06/12  61.3%
 10:49:01 : 07/13  50.0%
 10:49:22 : 07/14  60.5%
 10:50:21 : 08/15  50.0%
 10:50:39 : 09/16  40.2%
 10:51:00 : 10/17  31.5%
 10:51:24 : 11/18  24.0%
 10:51:56 : 12/19  18.0%
 10:52:53 : 12/20  25.2%
 10:53:11 : 12/21  33.2%
 10:54:12 : 12/22  41.6%
 10:54:43 : 13/23  33.9%
 10:55:08 : 14/24  27.1%
 10:55:56 : 15/25  21.2%
 10:56:07 : 15/26  27.9%
 10:56:42 : 15/27  35.1%
 10:57:04 : 16/28  28.6%
 10:57:33 : 17/29  22.9%
 10:57:57 : 18/30  18.1%
 10:58:52 : 19/31  14.1%
 10:59:24 : 20/32  10.8%
 10:59:58 : 20/33  14.8%
 11:00:46 : 20/34  19.6%
 11:01:11 : 21/35  15.5%
 11:01:32 : 22/36  12.1%
 11:02:09 : 23/37  9.4%
 11:03:34 : 24/38  7.2%
 11:03:52 : 25/39  5.4%
 11:04:26 : 25/40  7.7%
 11:04:34 : Test finished.
  ---------- 
 Total: 25/40 (7.7%)


----------



## RRod

castleofargh said:


> maybe they had big strong fingers? ^_^.
> but sure I also don't expect the track to be extraordinary by itself. they say what it is somewhere but I can't recall.


 
  
 Presto from Haydn Op.76 No.5 (one of the Erdödy quartets).


----------



## sonitus mirus

Spoiler: Warning: Spoiler!






greenears said:


> I read the other forums and it's a lot of back and forth about whether it was peer reviewed, very little meat.  I think few (none?) of the posters had actually read the paper.
> 
> Why on earth are they filtering?  BTW, I already looked in-depth at the anti-alias filters used in modern 24 bit DACs last week.  The are rock solid.  Bottom line - with a 512 tap linear phase FIR which nowadays is nothing to implement in 180 or 65nm, the filters are rock solid.  Less than 0.1 dB passband ripple then falls off a cliff at 0.45 Fs.  No chance of any audible aliasing in the stop band (I think it was -130 dB or better).
> 
> ...


 
  


  
 You should have someone provide 2 tests for you, one with identical files and another with a 24bit and 24-16bit, just to see what your results might look like when the files are truly the same.  If you had no idea which test was which, it might help further support the notion that you're hearing a difference.


----------



## Don Hills

greenears said:


> Where is the study/paper that proves that by experiment? It's so hard to find this stuff.


 

 Have you seen this?
  
 http://audioskeptic.blogspot.com/
  
 And these?
 http://www.aes.org/sections/pnw/ppt.htm
 They should keep you occupied for a couple of weeks...
  
 JJ Johnston arguably knows more than anyone else alive about the way the human auditory system works.
 https://home.comcast.net/~retired_old_jj/


----------



## Greenears

I do





don hills said:


> Have you seen this?
> 
> http://audioskeptic.blogspot.com/
> 
> ...



I don't see any test results in there. Did I miss something?


----------



## Greenears

don hills said:


> Have you seen this?
> 
> http://audioskeptic.blogspot.com/
> 
> ...


 huh? These are double blind tests in foobar. The method is well established. The only thing I want to see is that the tester creates the 16b file himself so we know the provenance. His post says he did.


----------



## RRod

Here's a little example I made for myself just now:
 https://drive.google.com/file/d/0BwmVtb5IwniEVTlzWExvNW1QTlE/view?usp=sharing
  
 The files are:
 pluck.flac = 24bit
 pluck16t.flac = 16bit truncated
 pluck16d.flac = 16bit triangular dither
 pluck16s.flac = 16bit shaped dither
 pluck.sh = shell script that made the files
  
 Basically I made a sequence of 1/2second-long synth plucks using sox, starting with 5s of them at full-scale (well more like -3dB) then doing a 1min long sequence of them fading to 0 at 24bits. I then resampled them 3 ways to 16bits. I did a few throw-away ABX trials and it wasn't looking good even for the truncated version, even if I cheated and pretended like the initial loud part didn't exist. But give it a try if you want; your results might vary.
  
 I chose plucks b/c the make it easy to decide when you stop hearing the fade, which for me was at 41s for all of them if I set the pot to a decently loud level at the start (at 41s the levels are about -60dB peak/-72 rms). But if you then jack the dial again, you can start hearing the tone again (note: RRod takes no responsibility if you blow out your eardrums if you skip back to the beginning after you do this!).
  
 Anyway, enjoy.


----------



## Don Hills

greenears said:


> I do
> I don't see any test results in there. Did I miss something?


 
 http://www.aes.org/sections/pnw/ppt/jj/fund_of_hearing.ppt
 Approximately 30 dB.
  
 No test results, it was likely pre-war research.
 I have access to the Bell Labs archives at work, I'll take a look...


----------



## Hellkitchen

Great read! Thank you


----------



## RRod

don hills said:


> http://www.aes.org/sections/pnw/ppt/jj/fund_of_hearing.ppt
> Approximately 30 dB.
> 
> No test results, it was likely pre-war research.
> I have access to the Bell Labs archives at work, I'll take a look...


 
  
 Agreed, thanks for that link. I like that he mentions realistic things like most of us don't listen to music in sound-deprivation tanks. On that note, he quotes a 114dB range as encapsulating what humans would need, even if we did listen to music in 6dBSPL environments. Are there any accepted notions of maximal dB range for the CD when dither is added to the mix? I've seen the 19bit range in searches often, and in the first post in this thread it says that 150 is possible in certain bands.


----------



## Greenears

don hills said:


> http://www.aes.org/sections/pnw/ppt/jj/fund_of_hearing.ppt
> Approximately 30 dB.
> 
> No test results, it was likely pre-war research.
> I have access to the Bell Labs archives at work, I'll take a look...


 
 Interesting read.  But still I think he makes my point:
  
 "To the best of my knowledge, nobody has run the sensitivity test to determine how much is audible, and at what frequency"
  
The stuff about pre-ringing is confusing even though I know what pre-ringing and linear phase is.  Also unclear if it is actually audible in practical music.  If it is already there in the recording what can be done about it during playback?


----------



## BackToAnalogue

greenears said:


> So still what am I hearing?
> My best guess lies in a misconception that many have about quantization noise. Please open any standard EE textbook on signals and systems. The first thing you will read is that Quantization errors are a non-linear process and cannot be completely analyzed mathematically. The ~6db per bit idea (which is where you get your 100 dB and 60 dB) is an approximation. I'm not making this up it says it right there in the textbook. I saw a U of Waterloo paper (google) that had a good intro summarizing Q noise, google it. The analogy is similar to FM and AM radio. In EE you learn how to completely analyze AM using Fourier and Laplace. Every detail can be described by nice equations with precise answers. Then the next thing you learn is that FM is a non-linear process that has no equivalent equation. But it sounds better. This revelation is very frustrating to young padawans, but you get over it after a few weeks. A few tricks and approximations and maybe computer simulations are used to analyze FM to enough extent to be able to use it. Same with Q noise - and actually it has some similarities to FM with repeated short spikes throughout the spectrum.
> 
> FFT:
> ...


 


I think that you are spot on here in your analysis of why some people can hear a difference. It is about the range of frequencies and how they change. Harmonics are of most importance. What you haven't quite realised is that the information about frequency shift is carried in the shape of the side of a SIN wave, so when you think about all of the SIN waves added together in every sample you will get more 'information' about the sides of waves if you have a more accurate reading and if you have more samples. 24 bit is actually only giving us 20.5 bits whereas 16 bit does give us 15 or 15.5. Assuming that the ADC used for recording was able to provided more than 21 bits of resolution then that is around 25% better resolution. So not massive and CDs sound a lot better through a good DAC anyway. So, it is reasonable to conclude that we are fairly close to getting as good as we can with this technology.


----------



## stv014

> Originally Posted by *Greenears* /img/forum/go_quote.gif
> 
> Chance of long runs of success:
> As I said a few pages back I'm well familiar with binomial distribution and I know that with increasing number of trials the chance of seeing any given run length increases asymptotically towards 100%.  But my total runs on test192 are maybe high double digits, and I gravitate towards 10% hitting 5% and 1% in places.  And the better success (in both tracks) corresponds to tracks where I heard a specific note.  Remember there are other tracks where I couldn't find anything to latch onto and I did much worse.  To say it could still be chance is always true - anything can always be chance.  But at this point I am starting to believe I may be hearing something.


 
  
 With the number of tests you did (I guess it is a total of hundreds of trials by now), getting occasional <0.05 p-values over short sections of long runs can easily happen by chance. And a 7/7 score at the beginning of a long run makes the entire rest of it look better. For example, without the first 4 trials, test192 would randomly vary between 12.5% (3/3) and 50% over the remaining 28 trials, with an overall score of 28.6% chance of guessing. When a 5/5 or 7/7 run does not occur, there is always the option of restarting the test until it does.
  


> Originally Posted by *Greenears* /img/forum/go_quote.gif
> 
> So still what am I hearing?
> My best guess lies in a misconception that many have about quantization noise.  Please open any standard EE textbook on signals and systems.  The first thing you will read is that Quantization errors are a non-linear process and cannot be completely analyzed mathematically.  The ~6db per bit idea (which is where you get your 100 dB and 60 dB) is an _approximation_.


 
  
 The absolute peak level of the error is predictable. With simple undithered rounding, it is guaranteed to be within +/- 0.5 LSB, or -96.3 dBFS with 16-bit samples. Dithering increases the overall level of the error, but - if implemented correctly, which is simple without noise shaping - eliminates the non-linear distortion, and the error is now noise at a predictable RMS level as well. If this did not work in practice, you would not be able to play your 24 or even 16 bit PCM samples at an acceptable quality on your very likely multi-bit delta-sigma DAC that has an actual "hardware" resolution of only a few bits.
  
 If you do not use dither, and listen to quiet sections of the music at a high volume, it is possible that you can hear the distortion, and it depends on the type of sound being played. Dither noise can of course become audible at sufficiently high volume, too, its advantage is that there are no narrow band distortion products that depend on the input signal and have a higher peak level in the frequency domain than the noise floor of dithered quantization.
  


> Originally Posted by *Greenears* /img/forum/go_quote.gif
> 
> FFT:
> You also can't say conclusively you looked at the FFT and didn't see anything.  While I agree it's true that you aren't going to miss some 50 dB spike, there are limitations with FFT.  Signals move in time, FFT is a slice in time.  To convert between domains you need a window like Hann or Blackman and the windows have artifacts.  I think everyone that has worked with this stuff hands on knows this.


 

 The spectrogram uses overlapped "slices", so no part of the input is missed. Different window sizes and types allow for trade-offs between frequency and temporal resolution, but this problem also exists in the case of human hearing. You can experiment with the window parameters, but any remotely reasonable settings will show a constant noise floor (if dithered) that decreases by 6.02 dB for each bit of resolution added.
   
 

 If you do not find that convincing enough, you can also examine the residual in the time domain, and see that its peak level is always limited to +/- 1.5 LSB with the commonly used triangular dither. You can also listen to it at a high volume, it should sound like noise that is not audibly modulated by the original input signal in any way.


----------



## Greenears

backtoanalogue said:


> greenears said:
> 
> 
> > So still what am I hearing?
> ...


  

 Haven't heard the term "side of a sine wave" before.  You mean the first derivative?  Where do you get 15.5 / 20.5 bits from?  if you are saying that quantization errors near the zero crossing cause subtle phase shifts I'm listening but the counter theory is that as long as the INL of the codec is good those errors get filtered at the output and the fundamental comes out unchanged with the errors pushed to higher frequencies at inaudible levels.  But tell me more....


----------



## Greenears

@stv014
  
 I hear you on all the amplitude analysis and 0.5 LSB.  But are you sure there are no subtle frequency shifts? The reason I am hung up on frequency/phase is that I am pretty convinced I can't hear 0.5 LSB anything at 16 bit. Most music is compressed to 20 dB DR and I don't know about the pieces I used but even if they are an extreme 50 dB it's still 40 dB above anything.  If you scroll back many many pages people assumed I was hearing noise and again to confirm I am hearing none whatsoever on any clip at any volume. It's got to be phase or I don't know what.


----------



## RRod

greenears said:


> @stv014
> 
> I hear you on all the amplitude analysis and 0.5 LSB.  But are you sure there are no subtle frequency shifts? The reason I am hung up on frequency/phase is that I am pretty convinced I can't hear 0.5 LSB anything at 16 bit. Most music is compressed to 20 dB DR and I don't know about the pieces I used but even if they are an extreme 50 dB it's still 40 dB above anything.  If you scroll back many many pages people assumed I was hearing noise and again to confirm I am hearing none whatsoever on any clip at any volume. It's got to be phase or I don't know what.


 
  
 You can gradually pin down the samples that are causing the phenomenon by trimming around where you hear the difference. If you trim the difference file at the same time, then you'll be able to see exactly what has been "taken away" from the 24bit file. I would happily post the files I've got, but I would think a complete difference file might be against the rules.


----------



## castleofargh

greenears said:


> @stv014
> 
> I hear you on all the amplitude analysis and 0.5 LSB.  But are you sure there are no subtle frequency shifts? The reason I am hung up on frequency/phase is that I am pretty convinced I can't hear 0.5 LSB anything at 16 bit. Most music is compressed to 20 dB DR and I don't know about the pieces I used but even if they are an extreme 50 dB it's still 40 dB above anything.  If you scroll back many many pages people assumed I was hearing noise and again to confirm I am hearing none whatsoever on any clip at any volume. It's got to be phase or I don't know what.


 

 I fail to understand how you can thumb up stv014's post(well I do agree with all that he said) and then post this?
 how loud do you expect some phase shift distortions caused by a  24 to 16bit conversion to stand?
  
  
  you didn't try to see if there were statistical changes(assuming that your posted abx weren't only your best runs) with and without dither to try and really rule out quantization noise. that's the very first thing I would have done if I happened to get more than guessing. instead now you're hunting for phase.


----------



## sonitus mirus

greenears said:


> @stv014
> 
> I hear you on all the amplitude analysis and 0.5 LSB.  But are you sure there are no subtle frequency shifts? The reason I am hung up on frequency/phase is that I am pretty convinced I can't hear 0.5 LSB anything at 16 bit. Most music is compressed to 20 dB DR and I don't know about the pieces I used but even if they are an extreme 50 dB it's still 40 dB above anything.  If you scroll back many many pages people assumed I was hearing noise and again to confirm I am hearing none whatsoever on any clip at any volume. It's got to be phase or I don't know what.


 
  
 I'm not sure that people were assuming you heard noise or anything different in your ABX test logs.


----------



## RRod

sonitus mirus said:


> I'm not sure that people were assuming you heard noise or anything different in your ABX test logs.


 
  
 More what I was trying to say to him was that all he *should* hear is noise, so that if he was hearing anything else then settings needed to be looked at.


----------



## stv014

> I hear you on all the amplitude analysis and 0.5 LSB.  But are you sure there are no subtle frequency shifts? The reason I am hung up on frequency/phase is that I am pretty convinced I can't hear 0.5 LSB anything at 16 bit. Most music is compressed to 20 dB DR and I don't know about the pieces I used but even if they are an extreme 50 dB it's still 40 dB above anything.  If you scroll back many many pages people assumed I was hearing noise and again to confirm I am hearing none whatsoever on any clip at any volume. It's got to be phase or I don't know what.


 
  
 Undithered quantization adds non-linear distortion, which can be subjectively perceived as different effects depending on what the input is. If the input is complex enough and has a high entropy, then the error can still end up being noise even without dithering. But with a low level and/or pure tonal signal, the distortion products (which are similar to those of crossover distortion in that they can be high order and not decrease with lower input level, so it is subjectively a "bad" type of distortion) can become audible at loud enough listening volume.
  
 Even if the music has overall not very high dynamic range, the dynamic range can be higher in individual critical bands with instruments that have a sufficiently "pure" spectrum with a low noise floor. So, if there is originally not much content around 3-4 kHz (the most sensitive band of hearing), for example, and the quantization adds a narrow peak there, it might not be masked, and it becomes audible. If you posted some short samples of the test tracks from the parts where there is apparently an audible difference, it could be analyzed better, and other people could try ABX testing the samples, too.
  
 As already suggested by others, it would also be interesting to see what happens if you use dither, which ensures that the quantization error is really just uncorrelated noise, and even though its overall RMS level is higher, it is always spread evenly over the spectrum without any narrow band distortion products.


----------



## limpidglitch

stv014 said:


> Undithered quantization adds non-linear distortion, which can be subjectively perceived as different effects depending on what the input is. If the input is complex enough and has a high entropy, then the error can still end up being noise even without dithering. But with a low level and/or pure tonal signal, the distortion products (which are similar to those of crossover distortion in that they can be high order and not decrease with lower input level, so it is subjectively a "bad" type of distortion) can become audible at loud enough listening volume.


 
  
 That effect could be heard (and seen) in the examples Rod posted a page back.
 A distinct unpleasant sawtooth like sound near the end of the 't' track, audible after 40dB or so of amplification.


----------



## RRod

limpidglitch said:


> That effect could be heard (and seen) in the examples Rod posted a page back.
> A distinct unpleasant sawtooth like sound near the end of the 't' track, audible after 40dB or so of amplification.


 
  
 Woo someone looked at them


----------



## Greenears

Thanks all for the interesting posts. It seems that some sort of consensus emerges that there might be some audible artifacts on a specific narrow section on undithered if it is unmasked by that passage. When I can I will try abx dithered against both 16 and 24 undithered and see where it leads. 

Sorry about the phase sidetrack but I was absolutely convinced by many posters I could not hear 0.5 lsb amplitude. But now I'm reunconvinced. Sic.


----------



## bigshot

It's pretty safe to assume that if you hear a difference between 16 and 24 using music at normal listenng volumes, something is wrong somewhere with either your files or your equipment.


----------



## RRod

greenears said:


> Thanks all for the interesting posts. It seems that some sort of consensus emerges that there might be some audible artifacts on a specific narrow section on undithered if it is unmasked by that passage. When I can I will try abx dithered against both 16 and 24 undithered and see where it leads.
> 
> Sorry about the phase sidetrack but I was absolutely convinced by many posters I could not hear 0.5 lsb amplitude. But now I'm reunconvinced. Sic.


 
  
 You can hear any signal that is actually there if you can crank the volume loud enough and if the signal is of a type that can stand out above the noise.
  
Here's a 10s, 10 octave sine sweep from 20Hz at -101dBFS in 24 bit.
Here's the same file at 16bit with triangular dither.
  
 If you look at the spectrograms you will see the sweep clearly on both, but the 16bit file has noise much closer in loudness to the sweep. So of course if you try to actually hear the sweep, you'll detect noise in the 16bit file you don't hear in the 24bit file. I could do the same thing with 32 vs 24bit as long as I'm making theoretical files on the computer. In either case, at -101dB, you'd have to be listening in an environment at really low SPL to be able to hear anything at full scale without pain.


----------



## Stillhart

Sorry if this has been asked already:  I can't dig through 2000+ posts to find the info.  Is HDTracks.com basically a scam then?  They're selling remastered tracks in 24 bit and that remastering is the difference we're hearing?
  
 Also, how does this theory relate to DSD?


----------



## castleofargh

stillhart said:


> Sorry if this has been asked already:  I can't dig through 2000+ posts to find the info.  Is HDTracks.com basically a scam then?  They're selling remastered tracks in 24 bit and that remastering is the difference we're hearing?
> 
> Also, how does this theory relate to DSD?


 

 you should think you're paying for the remastering. and in some cases it's well worth paying as they can sound much better than the CD version. but it's a case by case situation. some remarstering are absolute crap. also some hires albums are just the CD version upsampled so nothing gained there.
 that's why it's hard to give a straight definitive answer.
  
 DSD is worst, as almost all DSD albums come from PCM masters. so if they touched nothing, you're actually losing "quality" in the conversion(not to mention that a lot of DSD players will convert the DSD back to PCM, so useless double conversion). but just like HDtracks and 24bit albums, some masters will only be available in DSD. so if you want that particular master, you need to get the DSD and a DSD player.
  
 but yeah for us, the audio consumer, I would avoid thinking that high res equals to better sound.


----------



## Stillhart

Thanks, that's helpful.  This thread has been eye-opening.  As someone who doesn't particularly believe in the power of $300 cables and pointy things on the feet of my equipment, I'm happy to read scientific proof that some of the questionable stuff is effectively bunk.


----------



## headwhacker

stillhart said:


> Thanks, that's helpful.  This thread has been eye-opening.  As someone who doesn't particularly believe in the power of $300 cables and pointy things on the feet of my equipment, I'm happy to read scientific proof that some of the questionable stuff is effectively bunk.


 
  
 And for peace of mind you should do your own blind ABX test when you get the time. It will take some effort but it's worth it.


----------



## Greenears

@Stillhart
  
 Please note I did not test the HDtracks 16 bit versions, although I have them.  I downsampled the 24 bit versions.  That's because I was not trying to answer the question you are asking.  I was asking whether the 16 bit format itself was a limitation.  If they only make a certain master available in 24 bit that's a different issue but I don't know if that is the case.


----------



## kraken2109

Be careful what you say about HDTracks. They're a forum sponsor and defended by the mods. Last time I called them a scam I got moderated, deleted posts, warnings etc


----------



## Stillhart

kraken2109 said:


> Be careful what you say about HDTracks. They're a forum sponsor and defended by the mods. Last time I called them a scam I got moderated, deleted posts, warnings etc


 
  
 Nobody called them a scam.  I asked if they were a scam and I got a reasonable response.  I wish they'd make it more clear that you're basically paying for a remastering job, not any HD benefits, but caveat emptor.  *shrug*
  
 To be fair, I compared some of the stuff I've purchased from them to the CD version and the remastering on some tracks sounds pretty good.  I'm just not thrilled that I feel slightly misled as to the source of the sonic improvements.


----------



## analogsurviver

castleofargh said:


> you should think you're paying for the remastering. and in some cases it's well worth paying as they can sound much better than the CD version. but it's a case by case situation. some remarstering are absolute crap. also some hires albums are just the CD version upsampled so nothing gained there.
> that's why it's hard to give a straight definitive answer.
> 
> DSD is worst, as almost all DSD albums come from PCM masters. so if they touched nothing, you're actually losing "quality" in the conversion(not to mention that a lot of DSD players will convert the DSD back to PCM, so useless double conversion). but just like HDtracks and 24bit albums, some masters will only be available in DSD. so if you want that particular master, you need to get the DSD and a DSD player.
> ...


 
 I agree it is hard to give a definitive answer - due to the things mentioned.
  
 But I could not disagree more about DSD being the worst. NOT if it started life as DSD and was edited with tools that go to LOW digit PCM only around the intended edit and do not touch the original DSD otherwise - and if played on DAC capable of native DSD playback. That is >>90% pure DSD - available from the consortium 
 https://www.nativedsd.com/
  
 I agree ALL downloads should be documented as well as recent remasters of Jazz at the Pawnshop - ADC this and this for PCM, ADC this and this for DXD, ADC this and this for DSD.   I agree that merely "upsampling" CD content available to everyone to whatever "hirez" format and charging for it is a fraud. Given the progress in computer audio, today's SOTA pro converters might well get eclipsed by better consumer converters in few years - and the same can be said about the software.
 So, unless the download provider is willing to disclose the method by which the digital download has been created, I suggest caution with parting from your $. 
  
 If applied/executed correctly/honestly, hirez DOES equate better sound. But there is much fishing in murky waters going on at the moment. And no, one can not expect The Dark Side Of The Moon by Pink Floyd to remain the test disc forever - there IS such a thing beyond which ever better digital formats can no longer bring out of (ageing...) analog master tape anything "better" ( provided that the downloader did obtain the access to the original masters in the first place ) - so ALWAYS approach the remasters of remasters of remasters ........-......of remasters with a grain of salt.
 The last totally wrong one was (stereo) Beatles box of vinyl made from - DIGITAL remasters. Oh dear ... - paying that much for the WORST of both worlds...
  
 This is an appropriate tale of the wisdom of Nasrudin 
 http://en.wikipedia.org/wiki/Nasreddin
 : (from memory, read some 30 years ago...)
  
 There was a custom that upon a visit by a friend, the host was required to order his wife to kill a chicken and prepare a nice chicken soup for the guest. So, a good friend comes to see Nassredin > wife>chicken>soup>happy friend. But - the same custom applied in case somebody comes to visit you and presents him/herself as a friend of that Original #1 friend. So....>wife>chicken>soup>happy friend #2 of a friend #1. AND... it applied to whatever friend # XY who presented him/herself as friend of the friend # (XY-1). Upon an umpteenth visit, the wife of Nasreddin went quietly, without even having been told to do so, with head down, to the ever more depleted chickenhouse, thinking of yet another chicken going down the drain - but this time Nasreddin caught up with her, pulled her to side and said :
  
 "Now listen: go to the kitchen, chop some vegetables in the pot, WASH SOME FEET OF OUR CHICKEN in that pot, DO NOT kill any chicken - and boil/simmer  the soup as usual..." 
 " ?????"
 "JUST DO AS TOLD..."
  
 So, this friend of a friend of a.......- friend waits patiently for the renowned superb chicken soup of Nasreddin's wife - and is in the end of course surprised by the few vegatables floating in that boiled water. 
  
 Well - since you are a friend
	

	
	
		
		

		
			





 of a friend of a...............  -  .................- friend,
  
 THIS is
  
 a soup from the soup from the ......... - of the soup of .............   ..the soup 
	

	
	
		
		

		
		
	


	




.


----------



## cjl

Analogsurvivor: I've kind of given up a point by point rebuttal of your posts, since it doesn't seem to matter. You don't have any evidence for any of your claims though, and they're completely unsubstantiated. 16/44 PCM is audibly perfect, and if you want to claim otherwise, you should really show some actual evidence (double blind listening tests or measurements of some kind of distortion which is actually known to be audible).


----------



## analogsurviver

cjl said:


> Analogsurvivor: I've kind of given up a point by point rebuttal of your posts, since it doesn't seem to matter. You don't have any evidence for any of your claims though, and they're completely unsubstantiated. 16/44 PCM is audibly perfect, and if you want to claim otherwise, you should really show some actual evidence (double blind listening tests or measurements of some kind of distortion which is actually known to be audible).


 
 Cjl: I have only stated in my post that  it does matter whether any hirez is actual hirez or it is upsampled CD PCM redbook 44.1/16.
  
 From your position, ANYTHING above the CD is an overkill, not required and in the end regarded as ripoff.
  
 In one of the links "a couple of posts back" (in this or Testing Myths thread) , there was a reference to the list of papers presented at the AES convention in October 2014.
 By Meridian, possibly most likely by
  http://www.aes.org/events/137/presenters/?ID=2425
 - and one of the papers CLEARLY stated that for certain AUDIBLE signals CD is NOT transparent. Those who are members of the AES should be able to source this paper - there is any number of  discussions about it in the forums and my search for it is as good or flawed as any.  My pick is this: 
  
 http://www.whatsbestforum.com/showthread.php?15255-Conclusive-quot-Proof-quot-that-higher-resolution-audio-sounds-different
  
 It is a 155 page long thread - and I only went trough page one.  And no, although I will eventually go trough it all, I do NOT want to spend the time and resources merely to prove that 44.16 redbook CD is not enough. I know it - since I heard the Philips then prototype CD player in 1979 at our electronics show. Or was it 1980? - but that year it was alone. Next year, I remember besides production Philips model at least Hitachi - or maybe one or two more. The third year came the flood...
 Of course, CD DID improve over decades - but basic limitations that put me off initially ( when I was in my late teens/early 20s with undoubtedly better hearing than today ) remain - they are inherent.
  
 It boils down to this: if and when somebody or some organization has interest (or lack thereof ) in something, that person/organization will promote what is in his/its best (usually commercial) interest. Meridian used to be (mostly) CD oriented - but most probably they have been developing "beyond CD" behind the scenes for ages.
 Once there are tangible results (backed with some proof), they decided to go public.
  
 Another example is Chesky Records. I remember Chesky being one of the most outspoken opponents to binaural recording, less than 10 years ago. They did list a myriad of real, certainly plausible reservations against binaural. Look at their catalog
 http://www.chesky.com/ 
 now - AFTER they must have clearly "clandestinely" developed a successful binaural recording rig that sounds acceptable also on speakers - that is commercially far more viable than ordinary stereo recording. A decent pair of headphones and a decent DAP/DAC combo at say $500 and up level fed with a decent binaural recording will run rings around same music recorded in stereo and played on same value stereo speakers - and has incomparably more potential customers...- and no, they do not record redbook either. DSD - for a reason.
  
 But to put it bluntly; whoever had ever had the chance to listen to (in order of falling preference) live music/live mike feed/ analog reel to reel tape / analog CASSETTE tape / analog record / vs CD will know what I am talking about. Except for live music, everything else omits "something" in ever greater measure - and although on paper CD may well "look" the best of them all, it never did sound as real as other on paper far more flawed above mentioned sources. Hirez, regardless if PCM or DSD based, is simply trying to allow for the sound quality of analog (with(out) all of its shortcomings like channel separation etc - I AM familiar with them and no, I am not going to present them as virtues, because they are not ) with the convenience of redbook - so what are you so much against ?
  
 During the WW II, British had problem(s) with high flying German reconnaisance planes. While the progress on the official high altitude Spitfire version back in England was progressing at snail pace, real world pressing needs of warfare in Africa required action - immediately. So the 
 http://en.wikipedia.org/wiki/RAF_Aboukir 
 produced in practically field conditions a few Spitfire MK Vs that were stripped of EVERYTHING except one single gun in order to reach the altitude at which the Junkers JU 86 P/R were flying - with impunity so far. No radio /mast/antenna, no rear mirror, no paint (polished to the max to reduce weight/drag - I have read about it, although known photos show camouflage painted planes ), no rear seat armor - no NOTHING that was not absolutely required for the plane to still be able to fly. No pressurized cabin - with pilots willingly exposing themselves to the known and unknown dangers of flying at that high altitude. Such high flying Spit was meant to be used in pairs in order to bring the intruder low enough where more normally armed Spit could finish the job.
  
 It is not entirely clear whether or not these Spits have actually achieved any recorded air kills (but they did at least inflict damage )  - but their mere presence stopped the Germans from  flying. Which is de facto as good as shooting them down - without this additional air reconaissance, the Desert Fox no longer could place his numerically inferiour forces so effectively as before. It is a small, but important part of the mosaic in the African campaign.
  
 Official high altitude Spit MK VI
 http://en.wikipedia.org/wiki/Supermarine_Spitfire
 did not become available for another year or so after these Aboukir field conversions/strip-downs proved successful - and you can  count on the fact that NOBODY on the British ( or Geman...) side gave a damn whether these planes were "officially approved" or "botched in the field" - they fulfilled the NEED - and that was all that mattered.
  
 Similarly to the number of high flying planes ( <<<<< 1 % of total sorties by Luftwaffe ), what can not be captured by CD redbook 44.1kHz/16bit may be rare - but, like those few spy planes, can not be neglected/ignored. Were it not for that handful of ( less than five )  field modified Spits, the war would at the very least be prolonged - if not worse.


----------



## RRod

stillhart said:


> Nobody called them a scam.  I asked if they were a scam and I got a reasonable response.  I wish they'd make it more clear that you're basically paying for a remastering job, not any HD benefits, but caveat emptor.  *shrug*
> 
> To be fair, I compared some of the stuff I've purchased from them to the CD version and the remastering on some tracks sounds pretty good.  I'm just not thrilled that I feel slightly misled as to the source of the sonic improvements.


 
  
 I find lack of details… disturbing. It would seem easy enough for companies to put all kinds of information on hi-res masters on their websites. And not just "hey, this is 24/192", but actual differences to listen for in masters that go beyond "we made it louder so it will sound better." And of course they are usually low on comparison tracks, because if there aren't any mastering changes, then there really isn't anything to offer audibly from the higher rates/bit depth.
  
 Take the Pono store right now. Admittedly it's still a work in progress, but there's no information for a prospective buyer. OK, $25 for a new recording of Sounds of Silence… why? Because 24/192? That's it? No blurb about any new mastering or anything comparing it to the 16/44.1 version available for regular CD price? So yeah, I guess it's just the arbitrarily spaced graph of album resolution that supposed to make me want to pay double. L'sigh.


----------



## Greenears

So I'm back into testing again. I bought HD Tracks Daft Punk Random Access Memories Hi rez, 88/24. Instant Crush track 05, I'm about 16 trials in and getting nothing.  My standard system of Sox 24 to 16 conversion, no dither (I know everyone gets on me for that). I thought I might have something to latch onto on a number of marks but none of them have panned I'm I think 7/16 and clearly guessing.   Anyone have any other tracks/marks they can suggest?
  
 I have listened to this album a bunch over the holidays but now that I convert to 16 and actually try to ABX features I thought were due to 24 bit are apparently in the 16 bit downconvert.  But ... I've seen this before some tracks I can never get past guessing, then I'll find one and I can do reasonably well.  I really want to get a positive on ABX.  Please help me!


----------



## Stillhart

greenears said:


> So I'm back into testing again. I bought HD Tracks Daft Punk Random Access Memories Hi rez, 88/24. Instant Crush track 05, I'm about 16 trials in and getting nothing.  My standard system of Sox 24 to 16 conversion, no dither (I know everyone gets on me for that). I thought I might have something to latch onto on a number of marks but none of them have panned I'm I think 7/16 and clearly guessing.   Anyone have any other tracks/marks they can suggest?
> 
> I have listened to this album a bunch over the holidays but now that I convert to 16 and actually try to ABX features I thought were due to 24 bit are apparently in the 16 bit downconvert.  But ... I've seen this before some tracks I can never get past guessing, then I'll find one and I can do reasonably well.  I really want to get a positive on ABX.  Please help me!


 
  
 Can you please clarify?  You're looking to hear a difference between HD and CD?  Isn't this thread about how there's no difference?  Or am I missing something?


----------



## RazorJack

kraken2109 said:


> Be careful what you say about HDTracks. They're a forum sponsor and defended by the mods. Last time I called them a scam I got moderated, deleted posts, warnings etc


 
  
 I know how this feels man, been in a similar situation concerning a cable manufacturer.
  
 Don't worry, at least freedom of speech applies to one section of Head-fi: the Science forum!


----------



## bigshot

I'm the king of getting in trouble for that.


----------



## Greenears

stillhart said:


> Can you please clarify?  You're looking to hear a difference between HD and CD?  Isn't this thread about how there's no difference?  Or am I missing something?


 
 To 16 or to 24 that is the Question!  Or what was the question again? 
  
 You need to scroll back the last 10-20 pages or so for my other posts.  My summary:
  
 The OP made a lengthy statement advocating that 16 bit is all that is needed and then invited rebuttals.  I attempted to ABX test 24 bit vs 16 bit tracks to prove it one way or the other.  Unfortunately my tests have not been as conclusive as I thought they would be.  I'm still searching for that one track where I can get 9/10 and really definitely show there is an audible difference in 24.  I took a few week hiatus, RAM is my newest attempt.  I've gone through the HD Tracks sampler, the Linn sampler, and a few other suggestions.  RAM album was suggested by someone way back so I thought I'd have a crack at it.  In short, so far I've got tantalizingly close on a couple of tracks, but on many others it's clearly guessing.


----------



## Stillhart

greenears said:


> To 16 or to 24 that is the Question!  Or what was the question again?
> 
> You need to scroll back the last 10-20 pages or so for my other posts.  My summary:
> 
> The OP made a lengthy statement advocating that 16 bit is all that is needed and then invited rebuttals.  I attempted to ABX test 24 bit vs 16 bit tracks to prove it one way or the other.  Unfortunately my tests have not been as conclusive as I thought they would be.  I'm still searching for that one track where I can get 9/10 and really definitely show there is an audible difference in 24.  I took a few week hiatus, RAM is my newest attempt.  I've gone through the HD Tracks sampler, the Linn sampler, and a few other suggestions.  RAM album was suggested by someone way back so I thought I'd have a crack at it.  In short, so far I've got tantalizingly close on a couple of tracks, but on many others it's clearly guessing.


 
  
 Well it would seem that you've failed to prove your hypothesis.  Which was the expected result according to the first post.  Correct?


----------



## castleofargh

greenears said:


> So I'm back into testing again. I bought HD Tracks Daft Punk Random Access Memories Hi rez, 88/24. Instant Crush track 05, I'm about 16 trials in and getting nothing.  My standard system of Sox 24 to 16 conversion, no dither (I know everyone gets on me for that). I thought I might have something to latch onto on a number of marks but none of them have panned I'm I think 7/16 and clearly guessing.   Anyone have any other tracks/marks they can suggest?
> 
> I have listened to this album a bunch over the holidays but now that I convert to 16 and actually try to ABX features I thought were due to 24 bit are apparently in the 16 bit downconvert.  But ... I've seen this before some tracks I can never get past guessing, then I'll find one and I can do reasonably well.  I really want to get a positive on ABX.  Please help me!


 

 if you want to have fun try to find the CD version ^_^. I also went with that for some tests because I also read somewhere(who the hell wrote that??????) that it was a meaningful album to test high res. well I haven't been disappointed, on the hires version my brains are physically vibrating with the bass and sub bass boost. it sure is a meaningful album to see how they're selling us a mastering under the label "high res". 
  
 I wonder if the guys suggesting this album are so bad at noticing differences that they actually mistake a remaster and several db variations for being the result of high res recording?
  
 it conforted me into thinking that classical and some well recorded jazz stuff are the way to go when looking for details and dynamic.


----------



## Krutsch

greenears said:


> To 16 or to 24 that is the Question!  Or what was the question again?
> 
> You need to scroll back the last 10-20 pages or so for my other posts.  My summary:
> 
> The OP made a lengthy statement advocating that 16 bit is all that is needed and then invited rebuttals.  I attempted to ABX test 24 bit vs 16 bit tracks to prove it one way or the other.  Unfortunately my tests have not been as conclusive as I thought they would be.  I'm still searching for that one track where I can get 9/10 and really definitely show there is an audible difference in 24.  I took a few week hiatus, RAM is my newest attempt.  *I've gone through the HD Tracks sampler, the Linn sampler, and a few other suggestions.*  *RAM album was suggested by someone way back so I thought I'd have a crack at it.*  In short, so far I've got tantalizingly close on a couple of tracks, but on many others it's clearly guessing.


 
  
 So, I'm not a sound scientist, but like to think I am a good active listener. RAM is a great album (in a Grammy winning sort of way), but it's "busy" in that there is a lot going on, sound-wise, and it's kind of loud (IMO, although well recorded).
  
 May I make a suggestion? Try something that was originally recorded with/for DSD to start with that will have a mix of loud and quiet passages; it might be easier to find recognizable artifacts to get yourself to the goal of passing the test, so to speak. It's fascinating following your journey here.
  
 Anyway, maybe find a recording from the Pentatone catalog (assuming you like Classical) and look for something with enough variation to stretch the dynamic range (like, say, Mahler). I am a huge fan of Pentatone's recordings and they have a great reputation for sound engineering.


----------



## bigshot

I did a listening test with a sound engineer friend of mine using the Pentatone Parvo Jarvi Stravinsky disk. Couldn't tell any difference between the CD layer and the SACD layer. So I ripped it to AAC 256 VBR. Still no difference.
  
 Why waste time doing tests that have already been established (except to audiophools)


----------



## RRod

bigshot said:


> I did a listening test with a sound engineer friend of mine using the Pentatone Parvo Jarvi Stravinsky disk. Couldn't tell any difference between the CD layer and the SACD layer. So I ripped it to AAC 256 VBR. Still no difference.
> 
> Why waste time doing tests that have already been established (except to audiophools)


 
  
 I just got that disc. Audiophilia sure is a small world of a small world ^_^


----------



## RRod

krutsch said:


> So, I'm not a sound scientist, but like to think I am a good active listener. RAM is a great album (in a Grammy winning sort of way), but it's "busy" in that there is a lot going on, sound-wise, and it's kind of loud (IMO, although well recorded).
> 
> May I make a suggestion? Try something that was originally recorded with/for DSD to start with that will have a mix of loud and quiet passages; it might be easier to find recognizable artifacts to get yourself to the goal of passing the test, so to speak. It's fascinating following your journey here.
> 
> Anyway, maybe find a recording from the Pentatone catalog (assuming you like Classical) and look for something with enough variation to stretch the dynamic range (like, say, Mahler). I am a huge fan of Pentatone's recordings and they have a great reputation for sound engineering.


 
  
 The thing is that even something like Mahler done up by Telarc isn't even using all the dynamic range even of the CD, and already it's about all I can stand before needing to adjust the pot mid-symphony.


----------



## Stillhart

I'm really confused.  It looks like he's sampled more than enough to prove that there's no difference.  Is he being disingenuous?  In other words, am I missing out on the obvious sarcasm?


----------



## Krutsch

bigshot said:


> I did a listening test with a sound engineer friend of mine using the Pentatone Parvo Jarvi Stravinsky disk. Couldn't tell any difference between the CD layer and the SACD layer. So I ripped it to AAC 256 VBR. Still no difference.
> 
> *Why waste time doing tests *that have already been established (except to audiophools)


 
  
 Simply put: @Greenears wants to prove the assertion on its own merit, and I think some credit is deserved for the effort. I've always wanted to do a similar exercise, but I just don't have the time now. So, I thought I'd add my $0.02 to the discussion and recommend music that may or may not help.


----------



## RRod

stillhart said:


> I'm really confused.  It looks like he's sampled more than enough to prove that there's no difference.  Is he being disingenuous?  In other words, am I missing out on the obvious sarcasm?


 
  
 He's first operating under the hypothesis that there is a difference, and trying to hear what a positive test would sound like. There are in fact audible differences between 16 and 24-bits IF you jack up the volume to insane enough levels to hear near the noise floors, but no one listens to music this loudly. Hence the impossibility in finding an actual music track where one can detect a difference using a properly setup ABX test.


----------



## ascension278

Used Hugo + Roxannes to sample the tracks.

 Blind testing results:
  
 16/44.1
 Songs play at a higher volume (need to be cranked up to have equal levels)
 Songs have a less darker background. Somewhat grainier and more congested

 24/96
 Volume has to be cranked up
 Songs have darker backgrounds
 Song is more effortless
  
 17 times out of 20 I could identify the 24bit  songs
  
 * Using a variety of songs. But the 16/44 and 24/96 songs are different.


----------



## RRod

ascension278 said:


> Used Hugo + Roxannes to sample the tracks.
> 
> Blind testing results:
> 
> ...


 
  
 A proper testing framework would compare the same track at the two specifications (with the lower spec upsampled to prevent DAC hints), volume matched, ideally made from the same source as to avoid difference in mastering (say from a CD vs. an HD download).


----------



## Head Injury

ascension278 said:


> Used Hugo + Roxannes to sample the tracks.
> 
> Blind testing results:
> 
> ...


 

 If you're getting large volume differences you're comparing different masterings or your 16/44.1 encode is borked. If you're changing the volume manually you're very likely screwing up the level matching, and they'll sound different as a result.
  
 If you're not comparing with the same songs, from the same mastering, and level matched, the results aren't going to be conclusive.


----------



## bigshot

Actually it's most likely that your equipment is borking the high sample rate files.


----------



## headwhacker

bigshot said:


> I'm the king of getting in trouble for that.




I doubt that coz you are still gere


----------



## limpidglitch

Bork, bork, bork.


----------



## Greenears

ascension278 said:


> Used Hugo + Roxannes to sample the tracks.
> 
> Blind testing results:
> 
> ...


 
 As has been the practice on this thread, please post your ABX logs and file information so that others can reproduce your results.  If you refuse to post logs, your results will be discounted.
  
 To clarify the methodology a number of us agreed to, you need to downsample the 24 bit file to 16 bit and ABX that.  It doesn't mean you can't also ABX a published 16 bit version and provide two logs.  The reason for the downsample is to control for any possibility that the published 16 bit file has different mastering than the 24 bit.  By simply chopping the last 8 bits you eliminate mastering as a possible difference (and also in all of our experience the levels match perfectly).


----------



## Greenears

krutsch said:


> So, I'm not a sound scientist, but like to think I am a good active listener. RAM is a great album (in a Grammy winning sort of way), but it's "busy" in that there is a lot going on, sound-wise, and it's kind of loud (IMO, although well recorded).
> 
> May I make a suggestion? Try something that was originally recorded with/for DSD to start with that will have a mix of loud and quiet passages; it might be easier to find recognizable artifacts to get yourself to the goal of passing the test, so to speak. It's fascinating following your journey here.
> 
> Anyway, maybe find a recording from the Pentatone catalog (assuming you like Classical) and look for something with enough variation to stretch the dynamic range (like, say, Mahler). I am a huge fan of Pentatone's recordings and they have a great reputation for sound engineering.


 
 I'm willing to try, even though not a big Mahler fan.  Where can I get this Pentatone recording legitimately in the US? (We can't access Qobuz yet).
  
 Note I've already gone through 10 tracks of the HD Tracks sampler, 2 of the Linn sampler, and I have the full RAM album in 24 bit but only really tried one track so far.  I haven't had any success with louder passages so far, it tends to be on more moderate relative volme or single instrument sections.  Depsite the fact that initially I thought I could hear differences on some loud sections.


----------



## RRod

greenears said:


> I'm willing to try, even though not a big Mahler fan.  Where can I get this Pentatone recording legitimately in the US? (We can't access Qobuz yet).
> 
> Note I've already gone through 10 tracks of the HD Tracks sampler, 2 of the Linn sampler, and I have the full RAM album in 24 bit but only really tried one track so far.  I haven't had any success with louder passages so far, it tends to be on more moderate relative volme or single instrument sections.  Depsite the fact that initially I thought I could hear differences on some loud sections.


 
  
 You'll have issues testing Pentatone as they use SACD, so the ABXing is already more complicated since you can't use a typical foobar-like paradigm.


----------



## Greenears

rrod said:


> He's first operating under the hypothesis that there is a difference, and trying to hear what a positive test would sound like. There are in fact audible differences between 16 and 24-bits IF you jack up the volume to insane enough levels to hear near the noise floors, but no one listens to music this loudly. Hence the impossibility in finding an actual music track where one can detect a difference using a properly setup ABX test.


 
  
 To be imminently clear: I am testing the hypothesis that there is an audible difference between 24 and 16 bit, as a first step.  To test a hypothesis, you assume it is true and construct a test to prove that. But you need to be very careful in the test design that if you do get a positive result it can't be attacked on the grounds that the difference you heard was due to another uncontrolled factor.  So the simplest way to control for many issues (mastering, volume differences, downsampling filters) is to simply chop the last 8 bits of a 24 bit file, and leave everything else (including sample rate) untouched.  I am doing that with the incantation sox -G file24.flac -16 file16.flac and then I check the metadata and file size to see that it's appropriate.


----------



## Greenears

rrod said:


> You'll have issues testing Pentatone as they use SACD, so the ABXing is already more complicated since you can't use a typical foobar-like paradigm.


 
 I don't have SACD equipment, and I gather from forums that ripping SACD is hard (impossible?).  If there is a legitimately downloadable DSD file version of this mastering and if sox can convert it to FLAC 16 bit I'm happy to have a go at it.


----------



## Greenears

stillhart said:


> I'm really confused.  It looks like he's sampled more than enough to prove that there's no difference.  Is he being disingenuous?  In other words, am I missing out on the obvious sarcasm?


 
 I am not being sarcastic, I am really running the tests. I've posted a number of ABX logs above, and always respond to any requests for a repost.
  
 I've learned I'm a rare animal on these forums that I came into this without a firmly set opinion.  I learned this the hard way when any results I got leaning towards hearing a difference one crowd told me the test was flawed.  The moment my results leaned towards no difference, another crowd told me my tests were flawed, try different tracks etc etc.
  
 >It looks like he's sampled more than enough to prove that there's no difference. 
  
 Well, herein lies the 64 bit question. I would summarize my results so far (you can scroll up to see all the logs) as getting tantalizingly close to detecting a difference on a couple of tracks, but also unquestionably completely guessing on many other tracks.  The results are just right on the fence, not weak enough for me to dismiss but not the 1% on a large sample run that I'd like to see to prove the difference.  Adding fuel to the fire are a couple other posts I found that suggested a stronger difference (with valid methodology) but I've not been able to reproduce their setup for various reasons.  I would really like to see 9/10 or 17-18/20 to feel there is a clear difference.
  
 Another note, despite being repeatedly suggested on this thread, is that NO I am not cranking the volume and listening for noise.  I am listening at low to moderate volume and I've NEVER heard audible noise on any tracks 24 bit or 16 bit.  Where I've had limited success it's been on some type of qualitative timbre in the sound of an instrument.  
  
 A further wrinkle in this twisting journey is that recently I've started to look into the effects of Dynamic Rang Compression (DRC) on quantization noise.  But I've not been able to find definitive technical papers on that. Papers on QN yes, DRC yes, but not the combination thereof.  I've tried to think it through myself, but I'm unsure whether DRC will magnify QN on the soft passages, or will actually mask QN because it is a much bigger source of nonlinear distortion.  So I'm left uncertain whether to go for tracks that have high or low DR, or have been put through strong DRC or none.
  
 The quest continues....


----------



## Krutsch

greenears said:


> I'm willing to try, even though not a big Mahler fan.  Where can I get this Pentatone recording legitimately in the US? (We can't access Qobuz yet).
> 
> Note I've already gone through 10 tracks of the HD Tracks sampler, 2 of the Linn sampler, and I have the full RAM album in 24 bit but only really tried one track so far.  I haven't had any success with louder passages so far, it tends to be on more moderate relative volme or single instrument sections.  Depsite the fact that initially I thought I could hear differences on some loud sections.


 

 I only suggested Mahler because there might be a passage that ranges from very loud to very quiet; but as you mention above, you might try a violin/piano duo piece. HDTracks has a large selection of Pentatone classics for purchase.
  
 Speaking personally, I have spent the last couple of days re-sampling my high-res collection down to 48/16 or 44.1/16, using iZotope, so I can play my complete library on my Sonos boxes. What a headache... I'm sorry I ever ripped that first DVD Audio disc, which is what started me on the path for high-res.


----------



## Lespectraal

Audiophilia and religion are similar in the sense that both of them are somewhat faith based. Both seem to call upon the idea that there is "something", without the means of proving that. It is as if the idea in question is made up, totally within the realm of imagination and therefore has no place in reality.

 In fact, in philosophy, there is a topic called the mind body problem that explores and questions the relation between the mind and the body. One proposed answer to this problem is immaterial consciousness. It is said that our brains are mere "receivers" and that our consciousness is incorporeal, meaning that it has absolutely no physical foundation, therefore it cannot be measured, tested or likewise verified.

 Perplexing.


----------



## Greenears

krutsch said:


> I only suggested Mahler because there might be a passage that ranges from very loud to very quiet; but as you mention above, you might try a violin/piano duo piece. HDTracks has a large selection of Pentatone classics for purchase.
> 
> Speaking personally, I have spent the last couple of days re-sampling my high-res collection down to 48/16 or 44.1/16, using iZotope, so I can play my complete library on my Sonos boxes. What a headache... I'm sorry I ever ripped that first DVD Audio disc, which is what started me on the path for high-res.


 
 My condolences.  I already learned that lesson (with help from a friend) that whatever you decide on as your "archival" format needs to be carefully chosen.  Then, plan on simultaneously converting the golden archive version to several easily played formats so that your "xDevices" all work without constant reconversion.
  
 The nice thing about FLAC that I'm finding is that increasing number of devices just play it natively, even up to 192/24.


----------



## Greenears

lespectraal said:


> Audiophilia and religion are similar in the sense that both of them are somewhat faith based. Both seem to call upon the idea that there is "something", without the means of proving that. It is as if the idea in question is made up, totally within the realm of imagination and therefore has no place in reality.
> 
> In fact, in philosophy, there is a topic called the mind body problem that explores and questions the relation between the mind and the body. One proposed answer to this problem is immaterial consciousness. It is said that our brains are mere "receivers" and that our consciousness is incorporeal, meaning that it has absolutely no physical foundation, therefore it cannot be measured, tested or likewise verified.
> 
> Perplexing.


 
 Clever segue into esoteric philosophy 
  
 You may be right .... but Alan Turing proposed the Turing Test which seems to suggest a way to test if consciousness is present, sort of.  In a way, it is an ABX test of consciousness!  Yes there is hope


----------



## RRod

greenears said:


> Well, herein lies the 64 bit question. I would summarize my results so far (you can scroll up to see all the logs) as getting tantalizingly close to detecting a difference on a couple of tracks, but also unquestionably completely guessing on many other tracks.  The results are just right on the fence, not weak enough for me to dismiss but not the 1% on a large sample run that I'd like to see to prove the difference.  Adding fuel to the fire are a couple other posts I found that suggested a stronger difference (with valid methodology) but I've not been able to reproduce their setup for various reasons.  I would really like to see 9/10 or 17-18/20 to feel there is a clear difference.


 
  
 How likely would you want a false positive to be? If you want 1% (1 false positive out of every 100 tests), then you need 16/20 or 10/10. If you want 1 out of 1000 false positives, then you'd need 18/20 or 10/10. These are the numbers you should decide on: the false positive rate and how many trials you can stand to do.


----------



## Krutsch

greenears said:


> My condolences.  I already learned that lesson (with help from a friend) that *whatever you decide on as your "archival" format needs to be carefully chosen*.  Then, plan on simultaneously converting the golden archive version to several easily played formats so that your "xDevices" all work without constant reconversion.
> 
> The nice thing about FLAC that I'm finding is that increasing number of devices just play it natively, even up to 192/24.


 

 Yes. I actually have four complete copies of my library: ALAC (for use in iTunes and iDevices), FLAC (for in-home streaming), FLAC-2 (which has 44.1/48 down-sampled versions of high-res files for Sonos, Android devices and over-the-Internet streaming) and MP3/320 (Google Play).
  
 Lately, I've been using MinimServer/Streamer on a second Mac Mini which will transcode from anything to WAV, on-the-fly with FFMPEG, so I could probably get rid of the FLAC collections, but as you say, outside of Apple devices, FLAC seems to be standard for lossless.
  
 Obviously, I didn't *plan* on making this so complicated, but I am a gadget fiend and I keep all of these versions for testing out various devices.
  
 It's actually gotten to the point where my daughter and girlfriend have expressed frustration in how complicated the 2-channel and surround systems are (i.e. "...why can't we just listen to music without a tutorial?") - I am constantly moving gear around and buying new stuff. So, I solved that problem with a Sonos - now, they can stream the house library (FLAC-2 on a WD MyCloud), their own Google Play and Spotify accounts and leave the rest of the system (and me) alone.


----------



## RRod

krutsch said:


> Yes. I actually have four complete copies of my library: ALAC (for use in iTunes and iDevices), FLAC (for in-home streaming), FLAC-2 (which has 44.1/48 down-sampled versions of high-res files for Sonos, Android devices and over-the-Internet streaming) and MP3/320 (Google Play).
> 
> Lately, I've been using MinimServer/Streamer on a second Mac Mini which will transcode from anything to WAV, on-the-fly with FFMPEG, so I could probably get rid of the FLAC collections, but as you say, outside of Apple devices, FLAC seems to be standard for lossless.
> 
> ...


 
  
 And it always seems very logical to us: "Just set the TV to this, the receiver to this, and sing this chant 3 times"


----------



## bigshot

I've got a Logitech Harmony remote that automatically sets everything with one button on my iPhone. I hate unnecessary complication.


----------



## Greenears

rrod said:


> And it always seems very logical to us: "Just set the TV to this, the receiver to this, and sing this chant 3 times"


 
 Double double toil and trouble....


----------



## The Walrus

'High-Definition' Music Explained: Can You Really Tell the Difference?  
 http://www.billboard.com/articles/business/6429580/what-is-high-definition-music-debate-sony-walkman-pono 
 ​


----------



## limpidglitch

Brave journalist, there aren't many of them out there taking that line.
 And <3 Ethan Winer.


----------



## Krutsch

bigshot said:


> I've got a Logitech Harmony remote that automatically sets everything with one button on my iPhone. I hate unnecessary complication.


 

 I have a Harmony remote for the surround system, as well. It has one touch commands like: Watch TV, Watch Apple TV, Listen to Music, Watch DVD (everything is a DVD to them, Blu or otherwise).
  
 And, yet, they still complain that it's too complicated, because there are a lot of boxes and I have additional soft-key commands to do things like change the room correction settings, et al. I keep telling them to ignore the extra stuff, but it's too much. So, I go downstairs and turn it on for them: I hold the remote, press one key, wait, and then hand them the remote - seriously.


----------



## bigshot

Time to buy them a boom box!


----------



## Krutsch

bigshot said:


> Time to buy them a boom box!


 

 Ha ha... that's basically what the Sonos is.


----------



## Greenears

Well this is what success looks like.
  
 oo_abx 1.3.4 report
 foobar2000 v1.3.6
 2015/01/08 20:04:52
 File A: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\test192.flac
 File B: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\test192_1644.flac
 20:04:52 : Test started.
 20:06:37 : 01/01  50.0%
 20:06:46 : 02/02  25.0%
 20:06:57 : 03/03  12.5%
 20:07:05 : 04/04  6.3%
 20:07:12 : 05/05  3.1%
 20:07:19 : 06/06  1.6%
 20:07:26 : 07/07  0.8%
 20:08:42 : 08/08  0.4%
 20:08:54 : 09/09  0.2%
 20:09:44 : 10/10  0.1%
 20:09:49 : Test finished.
  ---------- 
 Total: 10/10 (0.1%)
  
 Before you get excited ... what I did just for grins was to change my Sox incantation to include a "rate 44100".  Up until today I was not changing sample rates, just chopping bits from 24 to 16.  So this is an ABX of 192/24 vs 44/16.  However .... turns out that in the original download from Linn there is a high pitched whine or whistle that was present equally in the 192/24 and 192/16.  But it went away in the 44.1 downsample.  Actually the downsample sounds better.  The lack of whine was obvious to me and I could get A or B within seconds.
  
 So this is not a valid test, but at least it proves I'm not asleep at the switch.  I redid vivaldi this way and got nothing (9/20).  I also did some regular un-resampled tests.  I downloaded some Ayre 24 bit needle drops of 3 vinyl tracks, chopped them to 16 and also got nothing (on the CSN track).  Does that prove 16 is enough to encode vinyl?  I also found a 1984 paper where BAS did a 16 bit codec loop test on vinyl and blind ABXd it and seem to show 16 can encode vinyl inaudibly.
  
 So where does that leave me? I guess still on the fence.  The problem with ABX is that negatives don't prove there isn't a positive out there, only the more you do the confidence interval improves until there is low likelihood there is a positive out there.  But I still have the record of a few good runs ... so I don't know.


----------



## castleofargh

I am the dither ghost and I come to haunt you!!!


----------



## bigshot

I don't want high pitched whistles or whines in my music


----------



## stv014

greenears said:


> Before you get excited ... what I did just for grins was to change my Sox incantation to include a "rate 44100".  Up until today I was not changing sample rates, just chopping bits from 24 to 16.  So this is an ABX of 192/24 vs 44/16.  However .... turns out that in the original download from Linn there is a high pitched whine or whistle that was present equally in the 192/24 and 192/16.  But it went away in the 44.1 downsample.  Actually the downsample sounds better.  The lack of whine was obvious to me and I could get A or B within seconds.


 
   
It sounds like there is some kind of ultrasonic interference in your 192 kHz sample, and due to playback issues, audible aliasing and/or IMD products appear in the audio band. One possible explanation is that the OS or audio driver resamples the 192 kHz input with a low quality converter.


----------



## Greenears

stv014 said:


> It sounds like there is some kind of ultrasonic interference in your 192 kHz sample, and due to playback issues, audible aliasing and/or IMD products appear in the audio band. One possible explanation is that the OS or audio driver resamples the 192 kHz input with a low quality converter.


 
 There is clearly some anomaly on that one track that exposes a bug somewhere in the end-to-end chain, that is sample rate dependent.  It's not related to the music content, it is there from start to finish even during quiet pauses.  So it is not aliasing or IMD of the music content - it may be bleeding in from a higher frequency but it sounds bug-like to me not any kind of regular distortion.  The other possibility is that it is actually an ultrasonic artifact from the recording process that is actually caught in the 192/24 file, and when I downsample it the FIR filter for the downsample knocks it out.  If by IMD you mean that, that it is an ultrasonic town that is somehow mixing in and causing a product in the audible range then that is possible.  I consider that a minor bug though - the whole output process should be immune up to 192/2 kHz and chop that off in the output LPF.
  
 But anyway, it only happened on one of my now 30 or so "hires" tracks and a short sample at that, and it's not anything to get me bent out of shape.   It's relatively subtle, only because I listened to that track a lot before could I pick it up.


----------



## RRod

greenears said:


> There is clearly some anomaly on that one track that exposes a bug somewhere in the end-to-end chain, that is sample rate dependent.  It's not related to the music content, it is there from start to finish even during quiet pauses.  So it is not aliasing or IMD of the music content - it may be bleeding in from a higher frequency but it sounds bug-like to me not any kind of regular distortion.  The other possibility is that it is actually an artifact from the recording process that is actually caught in the 192/24 file, and when I downsample it the FIR filter for the downsample knocks it out.  If by IMD you mean that, that it is an ultrasonic town that is somehow mixing in and causing a product in the audible range then that is possible.  I consider that a minor bug though - the whole output process should be immune up to 192/2 kHz and chop that off in the output LPF.
> 
> But anyway, it only happened on one of my now 30 or so "hires" tracks and a short sample at that, and it's not anything to get me bent out of shape.   It's relatively subtle, only because I listened to that track a lot before could I pick it up.


 
  
 If the filtering for 44.1 really got the sound out, then it's a tone above 20kHz, so either you have astoundingly good hearing or it's equipment related like IMD. What does the spectrogram look like?


----------



## castleofargh

greenears said:


> stv014 said:
> 
> 
> > It sounds like there is some kind of ultrasonic interference in your 192 kHz sample, and due to playback issues, audible aliasing and/or IMD products appear in the audio band. One possible explanation is that the OS or audio driver resamples the 192 kHz input with a low quality converter.
> ...


 

 could simply be that the other high res files you have don't possess much energy in ultrasounds. maybe have a look at that one song in audacity or other software?
 there was some keychain samples from arny krueger that seemed to help some people notice some "problems" like that.


----------



## Greenears

krutsch said:


> I only suggested Mahler because there might be a passage that ranges from very loud to very quiet; but as you mention above, you might try a violin/piano duo piece.* HDTracks has a large selection of Pentatone classics for purchase*.
> 
> Speaking personally, I have spent the last couple of days re-sampling my high-res collection down to 48/16 or 44.1/16, using iZotope, so I can play my complete library on my Sonos boxes. What a headache... I'm sorry I ever ripped that first DVD Audio disc, which is what started me on the path for high-res.


 
  
 OK, so I trolled around the HDtracks store a bit looking for Pentatone.  I also ran into a Deutsche Grammophon remaster of Von Karajan's classic 1963 recording of the 9 Beethoven symphonies now available in 24 bit.
  
 I have to say the experience left me wanting.  There is scant (read zero) information or liner notes on how and when these were mastered and recorded.  I went onto Pentatone and DG sites for more information, and came away with less. In addition some of the pricing is eye opening.  Particularly galling was one release was only available as a DSD.iso due to "recording quality".  Some others were not available for download at all with the legend "download not available due to the recording quality".  What nonsense is that?  And then charging more for DSD than PCM 192/24 and calling it "Studio Master" quality?  I don't mind if they charge a few extra $ for 24 bit or DSD but this implication that DSD is better than PCM192 is stretching things to say the least.
  
 If they want to say "not available for download due to contract issues" I have no problem with that - that is how the business works we know that.  But to claim that the disc is different quality than the download when it's exactly the same bits is shall we say a tad disingenuous. 
  
 As for DG/HVK 9 symphonies, that was even stranger.  HDtracks is offering 24 bit FLAC for $40, but DG site doesn't reference them only itunes for download which of course is not going to be in 24 bit.  The other option is BluRay.  Why woudn't DG reference HDtracks if it is an authorized true 24 bit download?  Again when I can buy a used box set of the umpteenth remastered CDs probably for $5, give me something that informs me what has been done in 2014 - if anything.
  
 Sigh. I'm done with the Iliad, moving onto the Odyssey. 
  
his file type is not available for download due to the recording quality, this file type is not available for download due to the recording quality​


----------



## Stillhart

greenears said:


> I am not being sarcastic, I am really running the tests. I've posted a number of ABX logs above, and always respond to any requests for a repost.
> 
> I've learned I'm a rare animal on these forums that I came into this without a firmly set opinion.  I learned this the hard way when any results I got leaning towards hearing a difference one crowd told me the test was flawed.  The moment my results leaned towards no difference, another crowd told me my tests were flawed, try different tracks etc etc.
> 
> ...


 
  
 I'm sorry but "tantalizingly close to detecting a difference" is another way of saying "detected no difference whatsoever".  You've found no difference ever.  You say that you have no firmly set opinion but it looks like you're doing everything you can to find a difference even though your empirical testing backs up the scientific theory.
  
 I have no opinion on this either other than reading and understanding the science.  Your testing seems to back up the science.  I appreciate that you're being super thorough.  I truly think you're doing a fantastic job of proving that there's no difference.  Which is why I don't understand why you seem so dead set on finding something that science agrees isn't there.  I feel like you're doing this Steven Colbert "mock conservatives by pretending to be one" kind of thing.


----------



## bigshot

Eventually random chance will come up with all sixes and you can yell Yatzhi! and claim victory.


----------



## castleofargh

stillhart said:


> I'm sorry but "tantalizingly close to detecting a difference" is another way of saying "detected no difference whatsoever".  You've found no difference ever.  You say that you have no firmly set opinion but it looks like you're doing everything you can to find a difference even though your empirical testing backs up the scientific theory.
> 
> I have no opinion on this either other than reading and understanding the science.  Your testing seems to back up the science.  I appreciate that you're being super thorough.  I truly think you're doing a fantastic job of proving that there's no difference.  Which is why I don't understand why you seem so dead set on finding something that science agrees isn't there.  I feel like you're doing this Steven Colbert "mock conservatives by pretending to be one" kind of thing.


 
 R.I.Port
	

	
	
		
		

		
			




 50% of my news source is gone, if they ever stop the daily show how will I learn about what's outside my window without being bored to death?


----------



## Krutsch

greenears said:


> OK, so I trolled around the HDtracks store a bit looking for Pentatone.  I also ran into a Deutsche Grammophon remaster of Von Karajan's classic 1963 recording of the 9 Beethoven symphonies now available in 24 bit.
> 
> I have to say the experience left me wanting.  There is scant (read zero) information or liner notes on how and when these were mastered and recorded.  I went onto Pentatone and DG sites for more information, and came away with less. In addition some of the pricing is eye opening.  Particularly galling was one release was only available as a DSD.iso due to "recording quality".  Some others were not available for download at all with the legend "download not available due to the recording quality".  What nonsense is that?  And then charging more for DSD than PCM 192/24 and calling it "Studio Master" quality?  I don't mind if they charge a few extra $ for 24 bit or DSD but this implication that DSD is better than PCM192 is stretching things to say the least.
> 
> ...


 

 Sorry you had a bad search experience. I have the Karajan Beethoven cycle on SACD; one of my favorites. I also have a number of recordings from Pentatone featuring Julia Fischer and Martin Helmchen, as well as her more recent Sarasate recording with Milana Chernyavska (Decca?). I have some of these on optical plastic, but the Sarasate I downloaded from HDTracks. I've seen most/all of the above that I have on disc also available for download at HDTracks.
  
 So, all of this was just a suggestion, but on to the Odyssey you shall go.
  
 I realize this is the sound science thread; but, at some point, you should consider just sitting back and enjoying the music.


----------



## Greenears

stillhart said:


> I'm sorry but "tantalizingly close to detecting a difference" is another way of saying "detected no difference whatsoever".  You've found no difference ever.  You say that you have no firmly set opinion but it looks like you're doing everything you can to find a difference even though your empirical testing backs up the scientific theory.
> 
> I have no opinion on this either other than reading and understanding the science.  Your testing seems to back up the science.  I appreciate that you're being super thorough.  I truly think you're doing a fantastic job of proving that there's no difference.  Which is why I don't understand why you seem so dead set on finding something that science agrees isn't there.  I feel like you're doing this Steven Colbert "mock conservatives by pretending to be one" kind of thing.


 
 All of these things are possible in theory.  Or, you could apply Occam's razor, and conclude that I was actually just tantalizingly close to detecting a difference.  Which is not a great place to be but that's where I am.
  
 All the details and log postings are a number of pages back.  They speak for themselves but if you have a question on a specific one let me know.


----------



## Stillhart

greenears said:


> All of these things are possible in theory.  Or, you could apply Occam's razor, and conclude that I was actually just tantalizingly close to detecting a difference.  Which is not a great place to be but that's where I am.
> 
> All the details and log postings are a number of pages back.  They speak for themselves but if you have a question on a specific one let me know.


 
  
 Sorry, it's not my intention to be insulting or put you on the defensive.  Like I said, I truly think you're doing good work.  
  
 But you can't be "tantalizingly close" to something that doesn't exist.


----------



## The Walrus

...


----------



## bigshot

I suppose you could think about it as an experiment that proves you can't hear a difference, or an experiment that you haven't been able to hear a difference yet. But I think it's easier to just conclude that if the difference is this hard to hear, it doesn't mean diddley squat when it comes to enjoying music played on your stereo.


----------



## lamode

gregorio said:


> It seems to me that there is a lot of misunderstanding regarding what bit depth is and how it works in digital audio.


 
  
 I just read this artilce. There are several small errors in it but the conclusion is of course correct.
  
 The biggest error was to say that "no one has a speaker system capable of 144dB dynamic range". Actually, we all do. They are passive analogue components so they have almost infinite DR (limited only by discrete electrons at the end of the day). This doesn't matter of course, because any DR beyond 80dB or so is wasted in real life (thanks to background noise) and no amplifier has this wide dynamic range.
  
 It would also be more clear to say that every bit you add reduces the noise level by 6dB, and if you already can't hear the noise in 16 bit (and you can't) then any further reduction is a meaningless waste of bits. It's that simple.
  
 You didn't touch on the sampling rate, but that's also very simple. Most microphones can't pick up 30kHz, most loudspeakers can't reproduce this kind of frequency and at the end of the chain, no humans can hear it, so the higher sampling rates are equally useless.
  
 I'm not going to read the whole 146 pages but I assume people have pointed out these and other mistakes/omissions.
  
 At the end of the day, you are still correct - high-res audio is a complete waste of time. If you do hear a difference it is because of some remixing or remastering. In which case you could downsample the high res to 16/44 and still enjoy the improvements. So buy the DAC which SOUNDS the best and forget this whole 24/32 bit 192/384/768kHz nonsense.


----------



## Greenears

stillhart said:


> Sorry, it's not my intention to be insulting or put you on the defensive.  Like I said, I truly think you're doing good work.
> 
> But you can't be "tantalizingly close" to something that doesn't exist.


 
 As I have posted already on this thread, others have posted results which indicate it exists. But I would like to reproduce it for myself.
  
 Nobody is insulting or defending.  It is patently obvious you think that anything above 16 bit is total bunkum, and will discount any test results ... as half the respondents on this thread have already shown.  And, there is another half that think that 24 bit or DSD is a slice of audio heaven, and we could have 1000 published test results and they would be ignored. 
  
 As already discussed at length above, maybe not everything in life is a conspiracy or Kobayashi Maru test.  It could be ... or .... maybe I jut want to know the answer.


----------



## RRod

greenears said:


> As I have posted already on this thread, others have posted results which indicate it exists. But I would like to reproduce it for myself.
> 
> Nobody is insulting or defending.  It is patently obvious you think that anything above 16 bit is total bunkum, and will discount any test results ... as half the respondents on this thread have already shown.  And, there is another half that think that 24 bit or DSD is a slice of audio heaven, and we could have 1000 published test results and they would be ignored.
> 
> As already discussed at length above, maybe not everything in life is a conspiracy or Kobayashi Maru test.  It could be ... or .... maybe I jut want to know the answer.


 
  
 See bigshot's comment, though. If the differences amounted to "audio heaven", then would you not agree that the effects should be just a bit more audible than you are finding? Many of us have already done our ABX tests on a few tracks and made our decision. If hearing *any* difference between 16 and 24bits requires ABXing hundreds of tracks, then one could perhaps say there is "some" difference, but night-and-day?


----------



## bigshot

greenears said:


> It is patently obvious you think that anything above 16 bit is total bunkum, and will discount any test results ... as half the respondents on this thread have already shown.  And, there is another half that think that 24 bit or DSD is a slice of audio heaven, and we could have 1000 published test results and they would be ignored.


 
  
 Not all opinions are created equal. Some people base their opinions on their own testing, and some people make stuff up in their head. The thing is, you aren't the first person on earth who has tried to discern a difference between HD and redbook. Around Sound Science, you're probably the only one who hasn't done all the testing they need to come to a conclusion. I did it ten years ago when I was working with a ProTools workstation used to record for the TV show I was working on at the time. I'm sure other people here go back further than I do. But the one thing I can say for all us is that our testing all pointed to the same thing. Yours points there too, but you don't want to admit it.


----------



## Greenears

rrod said:


> See bigshot's comment, though. If the differences amounted to "audio heaven", then would you not agree that the effects should be just a bit more audible than you are finding? Many of us have already done our ABX tests on a few tracks and made our decision. If hearing *any* difference between 16 and 24bits requires ABXing hundreds of tracks, then one could perhaps say there is "some" difference, but night-and-day?


 
 I've already stated many times above that if there is any difference, it's extremely close.  The several credible posts that said they did find a difference also said the same.  One said they could only pass the ABX test with one headphone and not the other, and at that very close.
  
 But even then there is doubt.  What if it is so close just because I don't have the best equipment or material?  Frankly the material is a bigger question mark - I don't know for sure it was actually recorded at 24 bit and the effects and compression applied.  For me if I'm archiving something forever, let's say in 10 years I get better equipment I don't want to have to go and re-buy the same material yet again.  For me to be really satisfied that 16 bit is the be all and end all I would want to record my own test material (so I could know the provenance) and then try maybe 3 best-in class DACs and 5 headphones and maybe a couple of speakers.  Then I would be satisfied at least for my ears.


----------



## Greenears

bigshot said:


> Not all opinions are created equal. Some people base their opinions on their own testing, and some people make stuff up in their head. The thing is, you aren't the first person on earth who has tried to discern a difference between HD and redbook. Around Sound Science, you're probably the only one who hasn't done all the testing they need to come to a conclusion. I did it ten years ago when I was working with a ProTools workstation used to record for the TV show I was working on at the time. I'm sure other people here go back further than I do. But the one thing I can say for all us is that our testing all pointed to the same thing. Yours points there too, but you don't want to admit it.


 
 If you have test results please publish them.  I did.  I used easily obtainable files, anyone here can try to reproduce my testing.
  
 Maybe I'm not the first person on earth, but nobody has found any published test results on 24 vs 16 in the last 7 years, with enough detail that you could try to reproduce.  So the earthlings seem pretty secretive.  This was also discussed at length above.  It's an easy test, and it needs to be done on updated modern 24-bit ADC/DAC.


----------



## RRod

greenears said:


> I've already stated many times above that if there is any difference, it's extremely close.  The several credible posts that said they did find a difference also said the same.  One said they could only pass the ABX test with one headphone and not the other, and at that very close.
> 
> But even then there is doubt.  What if it is so close just because I don't have the best equipment or material?  Frankly the material is a bigger question mark - I don't know for sure it was actually recorded at 24 bit and the effects and compression applied.  For me if I'm archiving something forever, let's say in 10 years I get better equipment I don't want to have to go and re-buy the same material yet again.  For me to be really satisfied that 16 bit is the be all and end all I would want to record my own test material (so I could know the provenance) and then try maybe 3 best-in class DACs and 5 headphones and maybe a couple of speakers.  Then I would be satisfied at least for my ears.


 


greenears said:


> If you have test results please publish them.  I did.  I used easily obtainable files, anyone here can try to reproduce my testing.
> 
> Maybe I'm not the first person on earth, but nobody has found any published test results on 24 vs 16 in the last 7 years, with enough detail that you could try to reproduce.  So the earthlings seem pretty secretive.  This was also discussed at length above.  It's an easy test, and it needs to be done on updated modern 24-bit ADC/DAC.


 
  
 So if you're not going to be satisfied with the tests you are posting here, what are any of us talking about? If you must have 3 DACs and 5 headphones, then go buy them, set up a proper statistical framework for making your final decision, and do the test in earnest. Then you can have peace of mind and we here on the forum can at least have a non-moving goalpost for our arguments for the "can't hear it" side.
  
 I think people tend to discard their ABX tests because the burden of proof really isn't on the Redbookers side. "Oh look, I couldn't tell a difference... how surprising" <- that's about how I feel, so I'm not too compelled to log the results somewhere. And even if I did, the counter would always be "well you need a different track" or "well you need a different DAC" or "well you need better headphones." Which is funny, because someone listening to recently-recorded classical on HD800s with a DAC that way out-specs the headphones would, I think, be exactly the kind of person who's ABX results could be considered sound. I mean, how good does my gear need to be? If someone wants to send me some Stax for free I'd happily do lots of ABX tests :3


----------



## Stereodude

greenears said:


> Or, you could apply Occam's razor, and conclude that I was actually just tantalizingly close to detecting a difference.


 
  
 Except the data doesn't suggest that.  If you were "close" the data wouldn't show you were guessing, but might not be to the level of conclusive.


----------



## castleofargh

greenears said:


> rrod said:
> 
> 
> > See bigshot's comment, though. If the differences amounted to "audio heaven", then would you not agree that the effects should be just a bit more audible than you are finding? Many of us have already done our ABX tests on a few tracks and made our decision. If hearing *any* difference between 16 and 24bits requires ABXing hundreds of tracks, then one could perhaps say there is "some" difference, but night-and-day?
> ...


 
  here obviously the devil's in the details and how much we decide to care about those incredibly small and most of the time inaudible details.
 if we take 2 TV or monitor displays of the very same model, they will have very slight differences, even on great expensive monitors made for image editing, pre calibrated by the manufacturer... you will still actually have small differences on the image, and with time more changes will occur(but they're not silly enough to call it burn in). does it matter? most people don't have a clue, and those who care will just get some colorimeter. and even then, most will just use the cheap stuff that changes the profile, not something advanced to really deal with the details. and that's pros we're talking about, few consumers bother with it. and that's vision, a sense that dominates hearing massively for a humans. and something we can look at for hours if we chose to, making comparisons and personal observation something much more important and accurate than with music changing every instant, and replacing what we're trying to hear with something else. confusing us even more.
  
 how many of the people so openly fond of all the high res audio improvements, used a calibration tool on their system and got a room treatment for their speakers? outside of pros or people close to the audio business, I would bet that it must amount to very very few speaker owners. maybe a handful of guys did some calibration on headphones because they happened to have the right tools. but I wouldn't expect more. so is high-res really about sound fidelity or about egos? look at all the guys who would rather have big frequency response errors instead of using EQ, those changes matter all the way up to 0db, but they have many excusesreasons not to do it.
 but all the noise from the studio needs to be perfect up to -144db! because I need to be able to hear what the guys were talking about in the room next to the recording studio when I listen to the song. a banana would find that reasoning dumb.
  
 and for archiving, sadly in 10years we'll all have lost a lot more hearing than any file format progress can ever hope to compensate for(and if we need to convert to some new format it will most likely be because of convenience like it happened with mp3 and flac). you'd tell me this for speakers, we could hope for some new technology one day to really improve them. now trying to know if our file will be perfect up to -96db or -144db or -20000db... who cares? no real change will ever happen in the -96 first db where the music is.
  
 I'm not ditching your efforts, you're curious, I also was at some point and I too tried what you tried, it's a very sane process to question things, and if anything those who never tried and talk about resolution are the ones I would criticize. but it doesn't change the conclusion about the need for high res, not one bit^_^.
 I'll start wondering if I want 24/96 files for all my music the day my sound system will actually have 96db resolution. and that's not for tomorrow morning, that much I can predict. until then, high res is just pissing in the wind for me.


----------



## sonitus mirus

It is almost as if some are searching for a 0.5% improvement with headphones where there is potentially 5% or more to be had with speakers and room treatment.  The resources necessary to achieve that 0.5% improvement could be better spent, if any improvement exists in the first place.


----------



## bigshot

greenears said:


> If you have test results please publish them.  I did.  I used easily obtainable files, anyone here can try to reproduce my testing.


 
  
 Why would anyone need to reproduce a test that resulted in no difference? Go ahead and do a fair test with any files and you'll come up with the same result.
  
 I'm interested in improving perceived sound quality. Testing is a necessary evil if I want to accomplish that. I've compared every component I've ever bought to make sure it's audibly transparent. If I *do* hear a difference, it gets packed back up in the box and returned. So far, I haven't found a player or amp that fails the test. Since everything was audibly transparent, I could have done no testing at all and ended up with the same result.
  
 There ARE things, however that DID make a significant difference. My selection of speakers made a huge difference. Room acoustics and equalization made the speakers perform at their best. 5.1 is a massive improvement over two channel. DSPs help a lot too. These are all the things that improve sound, but few people in audiophile forums discuss these topics. Instead they either discuss things like fancy cables and 24 bit high sampling rate that make no audible improvement whatsoever, or they talk endlessly about testing procedures to determine that cables and 24 bit make no audible improvement. At the end of the day, they haven't done anything to make their system sound better. How backwards is that?!


----------



## lamode

Here's a classic example of a product claiming 24 bit capability: The FiiO E10 http://www.head-fi.org/t/575084/impression-fiio-e10
  
 However the SNR is only 90dB, which is equivalent to 15 bits. The other 9 bits are just lost in noise! A quiet 16 bit DAC would actually be an improvement.
  
 But the point is moot, as real-world listening environments limit the dynamic range even more. Don't fall for the hype.


----------



## FFBookman

"sound science" ?!?!  please. myopic at best. trying to find math and laboratory examples of what real people hear all the time.
  
 you focus on the main program and the main instrument, and play on substandard players, the 24bit doesn't sound much better, could perhaps even sound worse. i agree there.
  
 you focus on the room and the shape and placement of the instruments, the air in the recording, you hear 24bit improvement. musicians and audio engineers hear it. everyone i've played 24bit to that has had more than 3 seconds to focus on it, hears it.
  
 to try to capture it and have your ears prove themselves over and over in a convoluted and clumsy "test" misreads how our ears work. they don't like to be trapped and tested. they won't give consistent results when asked to perform like this. this is the fatal flaw behind all of this so-called science.
  
 proving it's hard to hear in your lab doesn't mean it doesn't exist. you all reverse the placebo effect - you are accepting the placebo as the best and ignoring a higher quality. it's your loss.


----------



## FFBookman

gregorio said:


> [snip]
> 
> So, 24bit does add more 'resolution' compared to 16bit but this added resolution doesn't mean higher quality, it just means we can encode a larger dynamic range. This is the misunderstanding made by many. There are no extra magical properties, nothing which the science does not understand or cannot measure. The only difference between 16bit and 24bit is 48dB of dynamic range (8bits x 6dB = 48dB) and *nothing else*. This is not a question for interpretation or opinion, it is the provable, undisputed logical mathematics which underpins the very existence of digital audio.
> 
> So, can you actually hear any benefits of the larger (48dB) dynamic range offered by 24bit? Unfortunately, no you can't. The entire dynamic range of some types of music is sometimes less than 12dB. The recordings with the largest dynamic range tend to be symphony orchestra recordings but even these virtually never have a dynamic range greater than about 60dB. All of these are well inside the 96dB range of the humble CD. What is more, modern dithering techniques (see 3 below), perceptually enhance the dynamic range of CD by moving the quantisation noise out of the frequency band where our hearing is most sensitive. This gives a percievable dynamic range for CD up to 120dB (150dB in certain frequency bands)    [snip]


 
  
 Bull. Resolution is the number of steps on the Y axis that the ADC can encode, and 16,000,000 is a lot more slots than 65,000. It's like a car stereo that only has 13 volume settings, you just can't set it to exactly where you want it. That volume control suffers from low resolution, it cannot resolve at a precise point you prefer.
  
 Focusing on total dynamic range is missing the point of it all. As usual. The difference in 24bit is the accuracy of the soundstage, the accuracy and blending of the EQ, the blending of the parts, and the overall room air and shape that you can pick up in the recording. It gives all delays and reverbs a more accurate, easy to understand sound. Fatigue is lessened when listening loudly, because the ears are comfortable with the natural program presented.There are no formulas for this part of audio, no machine that will measure such things to be translated into digital, so it is roundly dismissed by 16/44 disciples. But it is the easiest tell with 24bit audio, because it's the thing they start removing when needing to get rid of data and shorten the file.
  
 Nyquist worked for the phone company and died of old age before CD's were even started. He wasn't around to defend himself. 16/44 is more corporate science than real science.


----------



## Greenears

ffbookman said:


> you focus on the room and the shape and placement of the instruments, the air in the recording, you hear 24bit improvement. musicians and audio engineers hear it. everyone i've played 24bit to that has had more than 3 seconds to focus on it, hears it.


 
  
 The first time I heard 24 bit on a 24 bit system I thought "Wow, this sounds great."  All kinds of detail, air etc.  Only when I chopped off 8 bits and did A-B comparison did I realize all of that was in the 16 bit as well.
  
 Maybe there is a clear example file out there, but I haven't found it yet.  Scroll back about 10 pages to see longer posts on my personal garden path journey.   You should try an A-B test - it's fun and you have nothing to lose.


----------



## castleofargh

ffbookman said:


> "sound science" ?!?!  please. myopic at best. trying to find math and laboratory examples of what real people hear all the time.
> 
> you focus on the main program and the main instrument, and play on substandard players, the 24bit doesn't sound much better, could perhaps even sound worse. i agree there.
> 
> ...


 
  
 you can criticize the audio tests for no being perfect, they're not. but then bring a solution that works better, not something that is not reliable and even less consistent.
 what you're offering is pretty much to have students at school not passing any test because they are stressful and not a good image of how they really are in real life. and somehow that's also true, some will have trouble when confronted with stress.
 but your present solution is to ask the student if he believes he's passing, and give him a diploma if he answers yes.
 let me tell you, I wouldn't like to ride a plane with a pilot who graduated that way.
  
 blind tests and abx are used everywhere, even for drugs. so they are good enough for important stuff related to health, but not good enough for you in audio. I'm sorry but I still think the placebo is on your side. if you can't show to anybody that you can discriminate cd from high res, why should we trust you? in fact how can you trust yourself?


----------



## castleofargh

ffbookman said:


> Bull. Resolution is the number of steps on the Y axis that the ADC can encode, and 16,000,000 is a lot more slots than 65,000. It's like a car stereo that only has 13 volume settings, you just can't set it to exactly where you want it. That volume control suffers from low resolution, it cannot resolve at a precise point you prefer.


 
 not true at all.
 increasing bit depth to 24bit really just offers values that are even quieter than -96db. they do not increase resolution in the 0 to -96db! you may not believe it but it's a fact. each bit added has half the voltage of the bit before it so all the added bits can never equate to the value of the 16th bit. meaning that they can only be used to describe sounds quieter than said bit above. it's a simple case of n/2+n/4+n/8...  http://en.wikipedia.org/wiki/1/2_%2B_1/4_%2B_1/8_%2B_1/16_%2B_%E2%8B%AF
 if the last bit in 16/44 is 1 then you can add as many bits as you like, you will never be able to make a sound louder than 1.


----------



## bigshot

It's not a good idea to go into the "Sound Science! This isn't SCIENCE!" When you don't know what science is yourself.


----------



## FFBookman

castleofargh said:


> you can criticize the audio tests for no being perfect, they're not. but then bring a solution that works better, not something that is not reliable and even less consistent.
> what you're offering is pretty much to have students at school not passing any test because they are stressful and not a good image of how they really are in real life. and somehow that's also true, some will have trouble when confronted with stress.
> but your present solution is to ask the student if he believes he's passing, and give him a diploma if he answers yes.
> let me tell you, I wouldn't like to ride a plane with a pilot who graduated that way.
> ...


 

 Interesting take. I hadn't thought of this in regards to grading DSP students. My debate was more philosophical about the foundations of hearing science which informs DSP science.
  
 I wouldn't care if you teach the students something like "under most use cases, a quality of 16/44 is sufficient. But under certain circumstances, like master-quality music and ultra high resolution location sound, both natural and artificial, a higher sample rate is preferable. 24/44 can exhibit a much larger soundstage and realistic timbre, with additional hard to measure qualities said to improve when played samples rates up to 192k optimally."
  
 That's more or les what I learned in the early days of DSP college curriculum, and that's also my experience in the practical side of things. I've heard 24bit, 16bit, 8bit, 4bit, etc. and I've listened on multiple occasions to sample rates up to 96. I don't have a ton of experience with 192k so I usually specify that I feel the real upgrade is 24bit anything. And the Beatles/AppleRecords/BBC agreed with me by releasing the entire Beatles catalog digitally at 24/44 and no higher.


----------



## RRod

ffbookman said:


> Interesting take. I hadn't thought of this in regards to grading DSP students. My debate was more philosophical about the foundations of hearing science which informs DSP science.
> 
> I wouldn't care if you teach the students something like "under most use cases, a quality of 16/44 is sufficient. But under certain circumstances, like master-quality music and ultra high resolution location sound, both natural and artificial, a higher sample rate is preferable. 24/44 can exhibit a much larger soundstage and realistic timbre, with additional hard to measure qualities said to improve when played samples rates up to 192k optimally."
> 
> That's more or les what I learned in the early days of DSP college curriculum, and that's also my experience in the practical side of things. I've heard 24bit, 16bit, 8bit, 4bit, etc. and I've listened on multiple occasions to sample rates up to 96. I don't have a ton of experience with 192k so I usually specify that I feel the real upgrade is 24bit anything. And the Beatles/AppleRecords/BBC agreed with me by releasing the entire Beatles catalog digitally at 24/44 and no higher.


 
  
 If I can go full snob, I have plenty of 16bit classical recordings with larger soundstage and more realistic timbre than any Beatles track I've heard. Soundstage and timbre are much more affected by venue, mic choice & placement, and mixing than 16 or 24 bits. Regardless, I know of no Beatles song that has the dynamic range to begin to tax 16 bits.


----------



## FFBookman

greenears said:


> The first time I heard 24 bit on a 24 bit system I thought "Wow, this sounds great."  All kinds of detail, air etc.  Only when I chopped off 8 bits and did A-B comparison did I realize all of that was in the 16 bit as well.
> 
> Maybe there is a clear example file out there, but I haven't found it yet.  Scroll back about 10 pages to see longer posts on my personal garden path journey.   You should try an A-B test - it's fun and you have nothing to lose.


 

 I can accept that, that's a compelling argument. I know I have been fooled even by 8bit files, and often times a 16bit file really gets to me emotionally. It's not a perfect science, our ears are not math.
  
 The crux of my argument supersedes the math because the math is based on false assumptions about human abilities to sense vibration and musicality. None of what I speak of shows up on a picture of a waveform or in math formulas. There's something about the space and room that it imparted in recordings that gets sucked out as you move down the resolution. It's not immediately identifiable as missing when you don't have it.
  
 Our ears adjust to what they are given, and they are never ABX tested in real life. It is an unnatural test and our auditory system, driven strongly by our emotions, hates it. It runs and hides and gets confused. It rebels. It is confused by volume and acts counterintuitively. Smaller and nearer sounds better, like a sweet sugar, dither is like honey, the modern production tricks hide or even exploit this limited room to their advantage. In summary, that test style might be "good enough for drugs" but it's not good enough for music. Music is ultimately emotionally stronger than any drug, by far.
  
 So a new test is needed. I will pay the _science of sound_ more respect when it does the work to actually test people in a natural and organic way, a way that respects the importance music plays in their emotional state, largely in private time, and does not treat the ears like a video game or a switch. Our ears don't want to play whack a mole but can hear the differences in detail and soundstage when they are locked in and enjoying the song over the course of days, if not weeks. You have to have someone who knows and loves a CD version of something then just give them a 24bit version from a capable playback system. Tell them or not, that's for you scientists to count.
  
 I know from watching it that they all hear it. People online like yourself are telling me they don't hear it even after meeting my various criteria, and I just feel sorry for them. I just hope they can hear it sometime, I don't want to be crazy and marginalized because I can hear different qualities of audio.
  
 I think most people can hear it but this effective DSP math should not convince them they can't. It's destructive and self-deprecating to a fault. The human hearing system along with our vibration and pressure changing sensors make us a *very dialed in* microphone. Throw in the stereo imaging and the huge brain full of memories and we are quite the hearing and mixing machine.


----------



## bigshot

Just do a fair line level matched direct switchable A/B blind comparison test. I can't tell you how many people came in here convinced they KNEW but only had subjective impressions to back it up. THAT isn't science. Or they try to argue that our ears or equipment aren't good enough. My ears and equipment are fine. Come by and visit me in Los Angeles and I will give you a demonstration to show you what REALLY matters when it comes to sound quality.


----------



## Stillhart

ffbookman said:


> I can accept that, that's a compelling argument. I know I have been fooled even by 8bit files, and often times a 16bit file really gets to me emotionally. It's not a perfect science, our ears are not math.
> 
> The crux of my argument supersedes the math because the math is based on false assumptions about human abilities to sense vibration and musicality. *None of what I speak of shows up on a picture of a waveform or in math formulas.* There's something about the space and room that it imparted in recordings that gets sucked out as you move down the resolution. It's not immediately identifiable as missing when you don't have it.
> 
> ...


 
  
 The red bolded part of your argument up there?  That sounds like religion, not science.  Your whole post reads like the exact opposite of science.  It's the same kinds of arguments that you might see people using to "prove" the existence of god.  
  
 "I know from watching it that they all hear it. People online like yourself are telling me they don't hear it even after meeting my various criteria, and I just feel sorry for them. I just hope they can hear it sometime, I don't want to be crazy and marginalized because I can hear different qualities of audio."
  
 Replace all the bits about audio with bits about god and reread what you just said.  /SMH


----------



## FFBookman

rrod said:


> If I can go full snob, I have plenty of 16bit classical recordings with larger soundstage and more realistic timbre than any Beatles track I've heard. Soundstage and timbre are much more affected by venue, mic choice & placement, and mixing than 16 or 24 bits. Regardless, I know of no Beatles song that has the dynamic range to begin to tax 16 bits.


 

 If that's full snob i'm right there with you. I can't argue with any of that. Which is why the consumer should get to choose. If mp3 is Price A, and CD is Price B, then 24bit can be Price C, I don't mind that much, I don't plan on buying any of that again for the rest of my life. It sounds like sitting in the control room of the mixing session, I don't need anything more.
  
 I also claim that it's not about total dynamic range, its what the ear uses those extra 15,935,000 possible values for. The spatial arrangement of the LR stereo mix, the so- called soundstage, is about very minimal interplay of Left/Right pan and mic placement. Different mics into different reverbs are things I can hear again at 24bit. I thought only vinyl delivered such visceral clues but I've been blown away by 24bit.
  
 I don't doubt you can test my ears and trick them. I've done some self-testing and if the song delivers a real punch at that moment, not the play before or after, it sounds amazing. I don't doubt that 16bit audio can sound great, even outstanding at times, but 24bit gives the bigger pallette, and we can hear it , or at least feel it, regardless of what we think.


----------



## RRod

ffbookman said:


> If that's full snob i'm right there with you. I can't argue with any of that. Which is why the consumer should get to choose. If mp3 is Price A, and CD is Price B, then 24bit can be Price C, I don't mind that much, I don't plan on buying any of that again for the rest of my life. It sounds like sitting in the control room of the mixing session, I don't need anything more.
> 
> I also claim that it's not about total dynamic range, its what the ear uses those extra 15,935,000 possible values for. The spatial arrangement of the LR stereo mix, the so- called soundstage, is about very minimal interplay of Left/Right pan and mic placement. Different mics into different reverbs are things I can hear again at 24bit. I thought only vinyl delivered such visceral clues but I've been blown away by 24bit.
> 
> I don't doubt you can test my ears and trick them. I've done some self-testing and if the song delivers a real punch at that moment, not the play before or after, it sounds amazing. I don't doubt that 16bit audio can sound great, even outstanding at times, but 24bit gives the bigger pallet, and we can hear it , or at least feel it, regardless of what we think.


 
  
 You have to allow for the possibility that our ear isn't doing anything with those extra 16million values. I'd argue you get more from soundstage by multiplying the data rate by 3 rather than by 256; that is, go to 5.1 x 16 x 44.1 rather than 2 x 24 x 44.1. If you don't think so, then that's fine. But if we can actually "sense" these extra values, then we should be able to demonstrate so in *some* kind of controlled environment. People can pass ABX of 8 vs 16bit with plenty of material and people will go "yes, the test is good". Suddenly when they can't pass 16 vs 24 people go "the test is bad; everyone knows you can sense the extra information." That's mighty convenient for companies trying to sell hi-res albums at twice price.


----------



## FFBookman

bigshot said:


> Just do a fair line level matched direct switchable A/B blind comparison test. I can't tell you how many people came in here convinced they KNEW but only had subjective impressions to back it up. THAT isn't science. Or they try to argue that our ears or equipment aren't good enough. My ears and equipment are fine. Come by and visit me in Los Angeles and I will give you a demonstration to show you what REALLY matters when it comes to sound quality.


 

 I'd take you up on if if I could but I don't plan on being in LA anytime soon. I hear the difference every time I make 16 bit dithered files from 24bit masters. I hear it every time I send 24bit mixes to my mastering engineer and get back 16bit and 24bit masters. I hear it almost every time a 24bit file comes up on my ponoplayer. 
  
 I'm sure you can play 16bit files that sound great. Me too. But if I was smart enough to do them all at 24bit 10-20 years ago I would be more part of the solution instead of a hypocrite that has been downsampling for years.
  
 I don't care for ear tests using the existing methods. They do not deliver solid data. Confusion reigns due to the intrusive testing environment.


----------



## Krutsch

ffbookman said:


> I hear it almost every time a 24bit file comes up on my ponoplayer.


 
  
 Ah... finally, we get to the bottom-line.


----------



## w00dman

Pardon for interrupting gentlemen, but it's kinda baffling that 24bit vs 16bit is even open for debate.
Isn't this one of easily disproved myths? Wasn't this settled like long time ago?

Furthermore its hard to argue, if someone is unwilling to accept physics and math, and is instead proposing his sixth sense,


----------



## castleofargh

w00dman said:


> Pardon for interrupting gentlemen, but it's kinda baffling that 24bit vs 16bit is even open for debate.
> Isn't this one of easily disproved myths? Wasn't this settled like long time ago?
> 
> Furthermore its hard to argue, if someone is unwilling to accept physics and math, and is instead proposing his sixth sense,


 

 welcome to headfi


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## w00dman

castleofargh said:


> welcome to headfi




well ty very much,
regardless if sarcasm or not


----------



## Head Injury

ffbookman said:


> I hear it almost every time a 24bit file comes up on my ponoplayer.


 
 Neil, we told you to stop coming back!


----------



## FFBookman

rrod said:


> You have to allow for the possibility that our ear isn't doing anything with those extra 16million values. I'd argue you get more from soundstage by multiplying the data rate by 3 rather than by 256; that is, go to 5.1 x 16 x 44.1 rather than 2 x 24 x 44.1. If you don't think so, then that's fine. But if we can actually "sense" these extra values, then we should be able to demonstrate so in *some* kind of controlled environment. People can pass ABX of 8 vs 16bit with plenty of material and people will go "yes, the test is good". Suddenly when they can't pass 16 vs 24 people go "the test is bad; everyone knows you can sense the extra information." That's mighty convenient for companies trying to sell hi-res albums at twice price.


 

 I agree with all that, and i think you will get the closest thing to a testable environment when you just give the consumers a choice. Whether it's apple/google offering 24bit files along with these HD-DAPS taking off, or some totally evil business decision to survive , If they go back to those analog masters and put them out at 24/192 i'm very interested in purchasing several of them to perform the test on my colleagues, friends and family. I guess I'm the religious one on this, a side I'm not usually on. I leave you this evening with a quote from someone that has done more science than all of us combined:
  
_“The whole point of science is that most of it is uncertain. That’s why science is exciting–because we don’t know. Science is all about things we don’t understand. The public, of course, imagines science is just a set of facts. But it’s not. Science is a process of exploring, which is always partial. We explore, and we find out things that we understand. We find out things we thought we understood were wrong. That’s how it makes progress.” – Freeman Dyson, 90, Mathematical Physicist_


----------



## Stillhart

I'm glad you see the irony in your own rhetoric such that you can post a quote that goes against everything you've just said.  
  
 EDIT - I just read this:  http://archimago.blogspot.de/2014/06/24-bit-vs-16-bit-audio-test-part-ii.html
  
 Apparently these folks didn't have enough faith.


----------



## lamode

> Originally Posted by *FFBookman* /img/forum/go_quote.gif
> 
> everyone i've played 24bit to that has had more than 3 seconds to focus on it, hears it.


 
  
 No-one can even hear the difference when a live signal is fed through a 16/44 digital loop. See the AES study, for example: http://www.aes.org/e-lib/browse.cfm?elib=14195
  
 So obviously no-one will hear the difference with higher-resolution formats either. You claim that people "hear it", but what do they hear? Did you perform a proper ABX test? No, of course not.


----------



## lamode

ffbookman said:


> Bull. Resolution is the number of steps on the Y axis that the ADC can encode, and 16,000,000 is a lot more slots than 65,000.


 
  
 And 1GHz is a much higher frequency than 20kHz, but so what? Your ears can't hear it and your ears also can't hear sounds at -144dB, which are represented by the 24th bit in digital audio. Your ears can't even hear the 18th bit. This is all wishful thinking, and proven as such in controlled tests.


----------



## EddieE

w00dman said:


> Pardon for interrupting gentlemen, but it's kinda baffling that 24bit vs 16bit is even open for debate.
> Isn't this one of easily disproved myths? Wasn't this settled like long time ago?
> 
> Furthermore its hard to argue, if someone is unwilling to accept physics and math, and is instead proposing his sixth sense,


 
  
 The problem is that you have a large section of the audio industry and media trying their hardest to convince people otherwise, selling products based on the exact opposite of those physics and maths. 
  
 The audio industry needs people to keep buying new gear, despite their old gear being good to go for decades if not a life time. The music industry needs people to start buying off them again. Establishing 24bit as an improvement meets both those objectives.
  
 Once a consumer spends a lot of money on such products and media, you then get an 'emperors new clothes situation' whereby they have to justify their position and hence become evangelists themselves.


----------



## lamode

eddiee said:


> The problem is that you have a large section of the audio industry and media trying their hardest to convince people otherwise, selling products based on the exact opposite of those physics and maths.
> 
> The audio industry needs people to keep buying new gear, despite their old gear being good to go for decades if not a life time. The music industry needs people to start buying off them again. Establishing 24bit as an improvement meets both those objectives.
> 
> Once a consumer spends a lot of money on such products and media, you then get an 'emperors new clothes situation' whereby they have to justify their position and hence become evangelists themselves.


 
  
 +1
  
 I would add that there is room for improvement in our playback systems, but the weak point is not the 16/44 format. So all this effort and marketing is being directed in the wrong places, which is the tragedy behind all this.


----------



## Lespectraal

Omne ignotum pro magnifico


----------



## Avi

stillhart said:


> EDIT - I just read this:  http://archimago.blogspot.de/2014/06/24-bit-vs-16-bit-audio-test-part-ii.html



 


Thanks for the link; fascinating!


----------



## PENTATONE

greenears said:


> OK, so I trolled around the HDtracks store a bit looking for Pentatone.  I also ran into a Deutsche Grammophon remaster of Von Karajan's classic 1963 recording of the 9 Beethoven symphonies now available in 24 bit.
> 
> I have to say the experience left me wanting.  There is scant (read zero) information or liner notes on how and when these were mastered and recorded.  I went onto Pentatone and DG sites for more information, and came away with less. In addition some of the pricing is eye opening.  Particularly galling was one release was only available as a DSD.iso due to "recording quality".  Some others were not available for download at all with the legend "download not available due to the recording quality".  What nonsense is that?  And then charging more for DSD than PCM 192/24 and calling it "Studio Master" quality?  I don't mind if they charge a few extra $ for 24 bit or DSD but this implication that DSD is better than PCM192 is stretching things to say the least.
> 
> ...


 

 Hi Greenears,
  
 Thank you for your feedback. We would like to give some explanations in response to your concerns.
  
 In every CD's booklet of the Remastered Classics Series, we provide information about the location and the date of the recording as well as the remastering.
 Moreover, we have received several questions on our social media (Twitter and Facebook) regarding the downloads availability of this series. We clearly stated there that the unavailability is indeed due to legal aspects (as you guessed), but we are working hard in order to find soon as solution for it. It would be very helpful to know where did you see the statement you mention above "the download unavailibility is due to the recording quality", so we will be able to correct this if it appears in any of our channels or address the issue to the appropriate party.
  
 If you need more information about the remastering process, you can always watch the video on our youtube channel:
 1. PENTATONE Remastered Classics: An Introduction: https://www.youtube.com/watch?v=nV8VGjfqIDU
 2. Zooming in into PENTATONE Remastered Classics: https://www.youtube.com/watch?v=GBctVE8qGEs
  
 Regards,
  
 PENTATONE


----------



## Greenears

pentatone said:


> Hi Greenears,
> 
> Thank you for your feedback. We would like to give some explanations in response to your concerns.
> 
> ...


 
 >It would be very helpful to know where did you see the statement
  
 I said on HDtracks right above.  As far as I know they are the only place offering Pentatone 24 bit for download in the US.
  
 CD booklets don't help the download generation.  We don't want physical media or physical books.  Further it doesn't help the CD customer either since they can't read that information until they unwrap the CD after which time it is not returnable.  ?! I want this information before I purchase it.  Why not just put some information about the recording date and method on the info next to the track on your or partner website?
  
 In any event now that you are responding - please let me know if there are any recordings made in the last 2 years that were 100% 24 bit digital (no tape involved) with state-of-the art 24 bit capture and processing and that are output in 24 bit.  I would be interested to know if any such track can be found in the US for download.


----------



## bigshot

I think most Pentatone releases are recorded DSD.


----------



## Krutsch

bigshot said:


> I think most Pentatone releases are recorded DSD.


 

 I think the Pentatone Remaster Classics referred to above are remastered to DSD from the original analog tapes. The founders purchased the rights and the tapes from Philips, which did the original recordings.


----------



## bigshot

Yes, but everything Pentatone does themselves is DSD based. (not that it makes a lick of difference.)


----------



## Lespectraal

In my "audiophile" experience, the only thing that really made a sonic difference is the mastering of the music.

I've come to realize that I am at the mercy of the sound engineers/producers for the actual quality of the music that reaches my ears.


----------



## Danz03

I totally agreed, but it also depends on the original mix master as well, if it was very badly produced or damaged, nothing in this world could make it better.
  
 Quote:


lespectraal said:


> In my "audiophile" experience, the only thing that really made a sonic difference is the mastering of the music.
> 
> I've come to realize that I am at the mercy of the sound engineers/producers for the actual quality of the music that reaches my ears.


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## Lespectraal

I meant to include that as well.


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## JHern

Great thread, not at all surprised to hear that the folks who made the CD standard knew what they were doing. Of course, this requires some care in the ADC and post-processing that are the final steps in digital audio mastering.
  
 I know it is a somewhat different topic...but curious to hear more thoughts: how does bit-depth affect the linearity of a digital sound reproduction system? Of course, the sound wave (speaker to ear) is itself linear, but what happens between the digital file and the movement of the speaker cone is not necessarily linear.


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## James-uk

jhern said:


> Great thread, not at all surprised to hear that the folks who made the CD standard knew what they were doing. Of course, this requires some care in the ADC and post-processing that are the final steps in digital audio mastering.
> 
> I know it is a somewhat different topic...but curious to hear more thoughts: how does bit-depth affect the linearity of a digital sound reproduction system? Of course, the sound wave (speaker to ear) is itself linear, but what happens between the digital file and the movement of the speaker cone is not necessarily linear.



http://youtu.be/nLEhfieoMq8
If I understand your question correctly then this video answers it.


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## Krutsch

james-uk said:


> http://youtu.be/nLEhfieoMq8
> If I understand your question correctly then this video answers it.


 

 That was a great video; more clear and concise than other examples I've seen and is one I will share when asked about 24-bit audio.


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## castleofargh

krutsch said:


> james-uk said:
> 
> 
> > http://youtu.be/nLEhfieoMq8
> ...


 

 the video is ok and the conclusions are obviously right. but it takes some shortcuts. a few people already find that Monty was taking too many shortcuts in his vids, so they would certainly not be happy about this one.
 also there are a few stuff I'm not sure I agree with. the most obvious is when he uses a DR value from the DR database, and takes it as the actual dynamic range of that track at the end. it's easy to find songs that have about 50db of dynamic range, but they won't get 50 as DR database value because they don't measure the same thing. for the video it's ok because the actual dynamic of those tracks are still well under 16bit, so the conclusion still works just fine.


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## Head Injury

james-uk said:


> http://youtu.be/nLEhfieoMq8
> If I understand your question correctly then this video answers it.


 

 Yes, nice simple video that clears up the basics very well!
  
 Avoid the comments though, trust me. I wonder if we know this "ezrazski" guy? His arguments sound so familiar.


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## Krutsch

castleofargh said:


> the video is ok and the conclusions are obviously right. but it takes some shortcuts. a few people already find that Monty was taking too many shortcuts in his vids, so they would certainly not be happy about this one.
> also there are a few stuff I'm not sure I agree with. *the most obvious is when he uses a DR value from the DR database, and takes it as the actual dynamic range of that track at the end.* it's easy to find songs that have about 50db of dynamic range, but they won't get 50 as DR database value because they don't measure the same thing. for the video it's ok because the actual dynamic of those tracks are still well under 16bit, so the conclusion still works just fine.


 

 Sure... but that is getting hung-up on details that are meaningless to a target consumer audience (e.g. my dad or a non-audiophool friend). Believe me, I am getting an increasing number of questions/comments from people about audio stuff that never before cared. I was in a work meeting last Friday with a storage engineer that I really respect and, out of nowhere, he starts telling me about his new John Coltrane SACD and how great it sounds (but he's puzzled about the whole copy protection / down-sampling thing with SACD and digital out).
  
 For someone that wants to argue with you about DR values, and starts with a knowledge base and a set of expectations, send them right to the AES convention papers of your choosing.
  
 IMO, what makes this video compelling is the lack of a condescending tone (e.g. like just about everything else I've read/seen). I really appreciated the illustrated "lollipop" diagram (which quickly covers the folly of stair-stepping, that we all learned in calculus, as an abstract representation of audio resolution), along with the highly demonstrative examples of quantization noise. That latter piece was nicely done and really drives home the point that those extra bits are just reducing noise.


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## JHern

james-uk said:


> http://youtu.be/nLEhfieoMq8
> If I understand your question correctly then this video answers it.


 

 A nice video, but no, it doesn't explain my question.
  
 The 16-bit vs 24-bit discussion thus far focuses entirely on the recording and digital mastering product. I completely agree with the main argument that 16-bit is more than sufficient as the mastering product. I have an advanced mathematical education so I have no issues understanding this part.
  
 My question has to do with the practical effect on electronics going in the opposite direction, what happens from the digital audio file to the speaker cone, and whether using a higher bit depth (perhaps in combination with variable/adaptive gain) at any point in this process is useless, or not? For example...my favorite-sounding audio player, Audirvana, uses 64-bit (variable gain) for internal processing (they say it is to avoid round-off errors)...is this just a waste of my computer's CPU time? Another example, is buying anything more than a 16-bit depth DAC also a complete waste of money? Could it be that all the digital processing in this stage is just nonsense, that the sound improvements we hear in players like Audirvana are just a matter of coloring/styling the output to sound more pleasing to the ear? Should we focus more attention on the pre-amp stages in the DAC chip rather than worrying about all of the digital machinations of the chip?
  
 I hope everyone can see that these are different questions. But I fear that the distinction might be lost on some, and if there is value to the A=>D=>A process (at any point) in using higher bit-depths then let's be careful to not throw the baby out with the bathwater.


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## lamode

jhern said:


> Audirvana, uses 64-bit (variable gain) for internal processing (they say it is to avoid round-off errors)...is this just a waste of my computer's CPU time? Another example, is buying anything more than a 16-bit depth DAC also a complete waste of money?


 
 If you are passing the data through multiple filters, then each stage will add up to half a LSB of error. So working in a larger bit space will allow you to apply filters without compounding the noise & distortion. Plus, some DSP algorithms require far more than 24 bits for intermediate computational products just to get final results accurate to near 16 or 24 bits.
  
 Why you would need 64-bit I have no idea, but I don't expect there is any penalty if you are using a 64-bit processor anyway (as most of us are these days). The data will be down sampled to 16 or 24 bit in the end, and will have up to half an LSB of error in the end, so if your DAC can handle 24 bit then better to output your data as 24 bit, though you probably won't hear the difference anyway.
  
 Good question, btw!


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## Greenears

DSP / Digital Filter algorithms use a basic element called a multiply-accumulate (MAC).  The idea is to multiply two numbers then store the result and add it to the next result (accumulate).  When you multiply two 16-bit numbers, you need 32 bits to store the full result if the values are large enough. Each time you add, in theory if the values are large enough you can need an extra bit for accumulate (since adding the two largest values is like doubling which is 1 bit in binary arithmetic).  For this reason standard DSPs feature a 16x16 into 40 bit MAC.  For audio you can do 24x24 into 48 or 60 bits, or in your case 64.  Often times they round up the accumulate word length since it is easier for computers to work on powers of 2 like 32 and 64 than other word lengths.


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## castleofargh

jhern said:


> james-uk said:
> 
> 
> > http://youtu.be/nLEhfieoMq8
> ...


 
 you question comes down to a very simple question, can you abx a 16bit vs a 24bit of the same song? the result gives you an idea of the importance of using heavy hires files. if you're looking for audible changes that's all you need.
 if you're looking for measurements, then we already have distortions and noise floor of gears showing that the math works pretty well with 16bit.


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## limpidglitch

castleofargh said:


> you question comes down to a very simple question, can you abx a 16bit vs a 24bit of the same song? the result gives you an idea of the importance of using heavy hires files. if you're looking for audible changes that's all you need.
> if you're looking for measurements, then we already have distortions and noise floor of gears showing that the math works pretty well with 16bit.


 
  
 He's not really asking if there's an advantage to 24bit files over 16bit ones, as I understand it.
 He's asking whether there's an advantage to using 64bit processing when adjusting volume, or using a 24bit DAC when playing back a 16bit file. To the latter question I'd say yes, nwavguy touched upon this while discussing the design of the ODAC. As to the former question I don't really know, but I suspect you're well past adding any benefits from higher precision processing.


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## castleofargh

I thought those were examples of a more general question. 
	

	
	
		
		

		
			




   
 

 I do use my odac with foobar's output in 24bit, but I really can't say I hear a difference(unless I go full replaygain and -70db on volume setting on the computer or something stupid like that). to me it's more of a safeguard so that I can use a little of foobar's volume without fear of crushing the track. that's my own level of homeopathy, I can't hear that it really saves anything, but I believe it does. 
	

	
	
		
		

		
		
	


	




 
 and on paper, I guess it can also serve to keep the D to A noise below, so that the actual noise floor still is the dithered noise from the CD(or the noise floor from the amp section?).
  
 anyway, it's not like 16bit DACs are still a thing nowadays. a few guys from Sparta resist with NOS DACs and painted abs, but that's pretty much it in audio.


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## FFBookman

High end math degrees can't hear better than my ears, or your own ears. If you only have measurements of 1 type of hearing (the frequency range) then you only have math for that 1 domain.
  
 Timing and spatial recognition in the simulated stereo field is where the major change is, and it's where all the 16bit holdouts like to avoid. The MP3 people outright dismiss all of it. How convenient to throw it away since you don't have math for it.
  
 Vinyl people and recording engineers and musicians and music lovers hear it because it's "presence".
  
 16bit has really impressed me through the ponoplayer, it is a decent format. But the 24bit files just have extra room, extra goodness, extra presense that you will never get a measurement for. 
  
 The "myth" is believing we have math equivalent to the human auditory system. Not even close.


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## lamode

ffbookman said:


> High end math degrees can't hear better than my ears, or your own ears. If you only have measurements of 1 type of hearing (the frequency range) then you only have math for that 1 domain.
> 
> Timing and spatial recognition in the simulated stereo field is where the major change is, and it's where all the 16bit holdouts like to avoid. The MP3 people outright dismiss all of it. How convenient to throw it away since you don't have math for it.
> 
> ...


 
  
 ... which is all nonsense, as no-one can tell the difference between 16 and 24 bit in an double blind test. I don't want to stop you from wasting your own money but once you write something like this, I feel I need to write something to prevent other trusting souls from wasting theirs.


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## FFBookman

castleofargh said:


> you question comes down to a very simple question, can you abx a 16bit vs a 24bit of the same song? the result gives you an idea of the importance of using heavy hires files. if you're looking for audible changes that's all you need.
> if you're looking for measurements, then we already have distortions and noise floor of gears showing that the math works pretty well with 16bit.


 

 I think ABX listening tests, along with confirmation bias, makes most of these results garbage. If you don't hear an advantage right away and feel it, you are pushing some sort of agenda.
  
 It isn't like the 16bit sounds like garbage, it sounds amazing if you love the song. The 24bit version, assuming from the same analog masters, sounds less digital. It's bigger, wider, more natural sounding. The presence and air and accuracy is there, whereas 16bit has a bit of that tinny small-box sound. But so much of it as about the performance, the mix, and the master.
  
 Then there's the format, something the marketplace actually determines. I say MP3 needs to be retired, any bitrate. FLAC 16/44 should be the baseline format for digital audio. Their lossy tricks worked for over a decade but we deserve better for ourselves. We don't have dialup anymore.
  
 Of course MP3 cheats and splices up it's bass frequency, and most modern players and headphones pump it back up to confuse the hell out of everyone. The modern production sucks all room out of the mix at every stage, with sometimes hundreds levels of compression at work in just the basic mix, not counting what the 10 other plugs in put on the signal.


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## FFBookman

lamode said:


> ... which is all nonsense, as no-one can tell the difference between 16 and 24 bit in an double blind test. I don't want to stop you from wasting your own money but once you write something like this. I feel I need to write something to prevent other trusting souls from wasting theirs.


 

 your tests lie to you. give you bad data. you base your entire argument on that bad data. people can hear better than an ABX test will ever show because the test itself assaults our natural hearing processes.


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## bigshot

My intelligence is being assaulted too! (joke)
  
 High end audio is built upon the foundation of expectation bias. It's the secret ingredient that makes 24 sound better than 16, one wire sound better than another wire, and expensive amps and DACs sound better than ordinary ones. Numbers and specs are the incantation that invokes the sonic improvement. The high end audio shaman uses imagery like "sound being chopped up into little bits" and "stair steps" to plant the seed in the mind that grows and grows as long as it is watered by imprecise testing and fertilized by even more subjective self justification. Self delusion is actually the single most effective way to make your stereo sound better to you. It really works! Thousands of high end audiophiles are proof of that.
  
 Unfortunately, you happen to be posting in the Sound Science forum. This is the one place where expectation bias is going to get a pin stuck in it. We don't have anything to sell you, so we won't feed your placebo needs. You'll only lose your expectation bias and lose all the value you have invested in it all these years. That would be a terrible waste of all that hard work.


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## FFBookman

just because it's good enough for medicine and visual domains doesn't mean it's good enough for our ears, or our music. 
  
 unless you get a better test than an ABX i will disregard your findings. you don't own math and physics, just the myopic kind that applies false limits based on ignorance. 
  
 there's more to hearing than you have the math, or instruments, to properly measure.


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## bfreedma

ffbookman said:


> just because it's good enough for medicine and visual domains doesn't mean it's good enough for our ears, or our music.
> 
> unless you get a better test than an ABX i will disregard your findings. you don't own math and physics, just the myopic kind that applies false limits based on ignorance.
> 
> there's more to hearing than you have the math, or instruments, to properly measure.


 
  
 You're trolling Sound Science, right?


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## limpidglitch

ffbookman said:


> High end math degrees can't hear better than my ears, or your own ears. If you only have measurements of 1 type of hearing (the frequency range) then you only have math for that 1 domain.
> 
> Timing and spatial recognition in the simulated stereo field is where the major change is, and it's where all the 16bit holdouts like to avoid. The MP3 people outright dismiss all of it. How convenient to throw it away since you don't have math for it.
> 
> ...


 
  
 Does listening to a 16bit file in a 24bit container sound like a fart in an auditorium?
 Does listening to down sampled 24bit file feel like riding in a convertible with a buzz-cut?
 Does listening to an mp3 file feel like sitting in a tattoo salon with a pair of didgeridoos stuck to the sides of your head?
  
  


ffbookman said:


> just because it's good enough for medicine and visual domains doesn't mean it's good enough for our ears, or our music.
> 
> unless you get a better test than an ABX i will disregard your findings. you don't own math and physics, just the myopic kind that applies false limits based on ignorance.
> 
> there's more to hearing than you have the math, or instruments, to properly measure.


 
  
 Thankfully maths and physics is completely open source, unlike audiophile hoodoo. If you wan't to critique it, have at it!
 If you want to wax poetically like some third rate Henry D. Thoreau, please go somewhere else.


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## FFBookman

haha i guess i am trolling if you guys have scared out all other thought. i was trying to follow your weird examples until i got your sarcasm. can you give me a measurement of how annoyed i am by your thought process?  do you have numbers for that?
  
 what's the best playback format for mathematicians? what's the best DAP under $500 for DSP programmers? do you compromise your music down for convenience but won't acknowledge other compromises?


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## bigshot

You're the one arguing that numbers matter, not us. Perhaps you could tell us which system is the best for mathematicians? I'm going to guess it's whatever system has the biggest numbers, regardless of whether it sounds any better or not.


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## castleofargh

ffbookman said:


> just because it's good enough for medicine and visual domains doesn't mean it's good enough for our ears, or our music.
> 
> unless you get a better test than an ABX i will disregard your findings. you don't own math and physics, just the myopic kind that applies false limits based on ignorance.
> 
> there's more to hearing than you have the math, or instruments, to properly measure.


 
 so subjective method isn't good enough, measurement isn't good enough, but you alone with a pono, and all is clear. how convenient. somehow I'm reminded of the dunning kruger link somebody posted some times ago.

 abx is one of the most effective subjective test to search for audible differences. if you know of a more efficient subjective test that we all can take at home without our own gears, I'm all ears. or are you just criticizing for the pleasure of it?
  
 now about "there's more to hearing than you have the math, or instruments, to properly measure." well sure, there is bias^_^. mood, expectations, pricetag, misinformation... they're all well passed measurements and math.
 but else, if it's on a CD, then it has been turned into volts, measured and turned into digits. as long as you're satisfied with the album you're listening to, your not making any sense.


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## limpidglitch

bigshot said:


> You're the one arguing that numbers matter, not us. Perhaps you could tell us which system is the best for mathematicians? I'm going to guess it's whatever system has the biggest numbers, regardless of whether it sounds any better or not.


 
  
 My impression is that mathematicians like small and concise numbers, engineers like round numbers and astrophysicists like huge numbers.
  
 By that reasoning a mathematicians system would be the bare minimum, implemented really elegantly, though sadly completely impractically. An engineers system would be the the required minimum, rounded up, so maybe 20bit/50kHz? And a an astrophysicist wouldn't really care, as sound can't travel in a vacuum anyway.
  
 Following from this I think it is reasonable to claim that the current redbook standard was devised by a mix of engineers and mathematicians, and not astrophysicists.


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## FFBookman

i'm not alone with pono, almost all musicians and recording engineers can hear it. they might not think it's needed for the average consumer, but they can hear it. most have been tracking and mixing at 24bit for over a decade now. and when played through a proper signal chain, oh looky there, sounds better. who cares about math. you should work on sorting out mathematically why it sounds better instead of telling people who hear it in the simplest terms that they are crazy. you are flat earth types here, relying on common sense to disguise the flaws in your argument.
  
 the quickness that you people dismiss people who make, record, and mix music for a living is amazing to me. they are the experts in music and emotional sound, not you, yet because they don't can't translate it to high-end physics and formulas you disregard them. do you think the people who provide enjoyable sound for you have math degrees? very few of them.
  
 but we can all hear and can understand quality if it's provided to us. we all get ear fatigue and can detect digital distortions, and we can all detect spatial differences. maybe you don't care, maybe you haven't figured out how to hear it yet. but you can and whether or not you've figured out how to get your antiquated formulas rewritten in enough time for anyone to care, we shall see.


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## FFBookman

I can get on board with that. Corporate-driven science often has to find "good enough".  Why not rely on voice and hearing loss studies to determine our hearing limits, and ignore all music and biology experts that hear deficiencies at 16/44?
  
 Because in the late 1970's they knew that a 24bit DAC was going to cost over $1k per chip. No one would buy a CD player if it cost over $2500 retail, no matter how cool the new technology was, or how great it sounded.
  
 Gotta get that cost down. Gotta make compromises. First there is cost of DAC, 16bit far cheaper than 24bit in 1980's.
  
 Also there's storage -- Redbook could hold about 760mb of data per disc, which would have translated to about 5 songs @ 24/96. Shorter run-time than cassette or vinyl. Nope.
  
 All great arguments for 16/44. All tied to ancient tech and ancient practices. All expired in 2015.


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## RazorJack

I knew I wouldn't regret subscribing to this thread


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## Tuco1965




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## RRod

He's just mad because he knows if we down-sampled some hi-res Ponostore material to 16/44.1, put it on the speakers, had him walk in the room, and told him "hey listen to this hi-res Pono track, isn't is so good!", he'd go "oh yeah, way better than Redbook!"


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## limpidglitch

ffbookman said:


> i'm not alone with pono, almost all musicians and recording engineers can hear it. they might not think it's needed for the average consumer, but they can hear it. most have been tracking and mixing at 24bit for over a decade now. and when played through a proper signal chain, oh looky there, sounds better. who cares about math. you should work on sorting out mathematically why it sounds better instead of telling people who hear it in the simplest terms that they are crazy. you are flat earth types here, relying on common sense to disguise the flaws in your argument.
> 
> the quickness that you people dismiss people who make, record, and mix music for a living is amazing to me. they are the experts in music and emotional sound, not you, yet because they don't can't translate it to high-end physics and formulas you disregard them. do you think the people who provide enjoyable sound for you have math degrees? very few of them.
> 
> but we can all hear and can understand quality if it's provided to us. we all get ear fatigue and can detect digital distortions, and we can all detect spatial differences. maybe you don't care, maybe you haven't figured out how to hear it yet. but you can and whether or not you've figured out how to get your antiquated formulas rewritten in enough time for anyone to care, we shall see.


 
  
 But the earth _is_ flat. As θ approaches 0, sinθ approaches θ.
 Simple, even Archimedes knew that.
  


ffbookman said:


> I can get on board with that. Corporate-driven science often has to find "good enough".  Why not rely on voice and hearing loss studies to determine our hearing limits, and ignore all music and biology experts that hear deficiencies at 16/44?
> 
> Because in the late 1970's they knew that a 24bit DAC was going to cost over $1k per chip. No one would buy a CD player if it cost over $2500 retail, no matter how cool the new technology was, or how great it sounded.
> 
> ...


 
  
 Audiophiles have been singing this song for as long as CDs have existed, and yet no-one has managed to demonstrate the purported shortcomings.


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## bigshot

ffbookman said:


> i'm not alone with pono, almost all musicians and recording engineers can hear it.


 
  
 Did you know that you were speaking to a person who has worked professionally as a sound engineer right now?


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## castleofargh

my mom payed me 5 francs(old frog money) when I was a kid so that I would stop playing the recorder at home. being payed made me a professional by definition.
fear the cursed tool of evil!!!!!!!


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## FFBookman

bigshot said:


> Did you know that you were speaking to a person who has worked professionally as a sound engineer right now?


 

 Congratulations, friend. I too have made and published music, I don't doubt the size of your multitrack. Can you tell me what dither is?
  
 I'm sure you use it, please explain to me what it is doing and why I need it when going from 24bit audio to 16bit audio?


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## FFBookman

Do you doubt my redbook breakdown?  
  
 Do you really believe that it was scientifically proven by "pure science" that there was a digital sampling rate that represented all that humans could ever detect
  
 or do you believe there were real limitation to both storage on the small optical disc as well as DAC cost that drove 16bit as the standard.  Phillips wanted 14bit and smaller discs but Sony cockblocked and moved it to 16bit with larger discs. Phillips had already retooled their factory for their size 14bit discs, Sony scored some points. 
  
 All cool and logical in a business sense, but this farce about there is nothing more than that in the DSP realm is crazy. We all know it's there, that's why sound cards and every mixing gear in 15 years has done 24bit. Are you calling every person that works in audio anywhere where 24bit is used guilt of snake oil? 
  
 Crazy. It's 2015. We were hearing the 24bit difference in the studio by the mid,late 90's.


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## James-uk

ffbookman said:


> Do you doubt my redbook breakdown?
> 
> Do you really believe that it was scientifically proven by "pure science" that there was a digital sampling rate that represented all that humans could ever detect
> 
> ...



Can you provide any evidence to back up your claims?


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## FFBookman

james-uk said:


> Can you provide any evidence to back up your claims?


 

 Google it yourself. Read up on redbook creation, read up on laserdisc and other optical disc development in the 70's. I don't keep links handy because I'm not paid to do this.
  
 Also read up Nyquest. He worked for the phone company most of his life developing early digital telephony concepts and sampling rates needed to transmit mono voice. Great work but not the foundation of hearing science or having anything to do with simulated stereo music production that we all listen to.
  
 He is always pulled out of context because his theorem is inflated to infallible when it knows nothing of stereoscopic listening and vibration sensing. It knows nothing of the room and position of the ears. It has none of that contextual depth.
  
 The evidence is when I hear 24bit on my pono player, played through almost any set of speakers. They are alive and full of depth and do this magical thing - they start to disappear.


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## RRod

ffbookman said:


> Do you doubt my redbook breakdown?
> 
> Do you really believe that it was scientifically proven by "pure science" that there was a digital sampling rate that represented all that humans could ever detect
> 
> ...


 
  
 So why are you ignoring the fact that there's a difference between mastering at 24bit and delivering content at 24bits? No one begrudges engineers for doing the former, but the latter is unnecessary, especially if they're going to charge the prices they do for HD content.
  
 Actually, nm, I know the answer already.


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## limpidglitch

ffbookman said:


> Also read up Nyquest. He worked for the phone company most of his life developing early digital telephony concepts and sampling rates needed to transmit mono voice. Great work but not the foundation of hearing science or having anything to do with simulated stereo music production that we all listen to.
> 
> He is always pulled out of context because his theorem is inflated to infallible when it knows nothing of stereoscopic listening and vibration sensing. It knows nothing of the room and position of the ears. It has none of that contextual depth.


 
  
 Could you explain why going from mono to stereo (and presumably to surround) would make a difference to this, and why bit depth is of such importance to spacial cues?
  


castleofargh said:


> my mom payed me 5 francs(old frog money) when I was a kid so that I would stop playing the recorder at home. being payed made me a professional by definition.
> fear the cursed tool of evil!!!!!!!


 
  
 Ah, memories 
	

	
	
		
		

		
		
	


	



 Are these infernal devices as prevalent in US primary schools as they seem to be in European?


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## lamode

ffbookman said:


> your tests lie to you. give you bad data. you base your entire argument on that bad data. people can hear better than an ABX test will ever show because the test itself assaults our natural hearing processes.


 
  
 Music assaults our natural hearing processes?


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## bigshot

ffbookman said:


> Can you tell me what dither is?


 

 To learn about dither and how important it is or isn't, you can refer to the videos in my sig file. I'm not paid to follow your tangential questions. I'm talking about whether redbook audio is capable of reproducing everything humans can hear in recorded music or not.


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## lamode

ffbookman said:


> i'm not alone with pono, almost all musicians and recording engineers can hear it. they might not think it's needed for the average consumer, but they can hear it.


 
  
 When can these mythical people hear it? As they clearly can't hear a difference when you play 16 and 24 bit music side by side.


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## lamode

ffbookman said:


> I can get on board with that. Corporate-driven science often has to find "good enough".  Why not rely on voice and hearing loss studies to determine our hearing limits, and ignore all music and biology experts that hear deficiencies at 16/44?
> 
> Because in the late 1970's they knew that a 24bit DAC was going to cost over $1k per chip. No one would buy a CD player if it cost over $2500 retail, no matter how cool the new technology was, or how great it sounded.
> 
> ...


 
  
 Nonsense. You could very easily make a cheaper CD player with a 16 bit DAC even if the data is in 24 bit format. In fact some early CD players used 14 bit DACs to save money, which proves my point.
  
 No need to add more bits as you will NEVER be listening in a room with low enough noise to hear the extra bits anyway.


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## James-uk

ffbookman said:


> Google it yourself. Read up on redbook creation, read up on laserdisc and other optical disc development in the 70's. I don't keep links handy because I'm not paid to do this.
> 
> Also read up Nyquest. He worked for the phone company most of his life developing early digital telephony concepts and sampling rates needed to transmit mono voice. Great work but not the foundation of hearing science or having anything to do with simulated stereo music production that we all listen to.
> 
> ...




have you any evidence to show that any human can hear a difference between 24 and 16 bit audio? You seem to have time on your hands so please humour me with this information. I don't want anecdotes I want evidence. If you cannot provide this then your claims are empty and therefore void.


----------



## lamode

ffbookman said:


> The evidence is when I hear 24bit on my pono player, played through almost any set of speakers. They are alive and full of depth and do this magical thing - they start to disappear.


 
  
 Along with your credibility.


----------



## James-uk

http://archimago.blogspot.ca/2015/01/last-words-on-pono-mastering-analysis.html?m=1


----------



## ericr

james-uk said:


> http://archimago.blogspot.ca/2015/01/last-words-on-pono-mastering-analysis.html?m=1




So does that mean those who bought into Neil Young's hype have been p(ono)wned?


----------



## Happy Hamlet

james-uk said:


> http://archimago.blogspot.ca/2015/01/last-words-on-pono-mastering-analysis.html?m=1


 
  
 This blog claims to be "a 'more objective' take on audiophile topics(...)". Well, I guess everything is relative


----------



## cjl

happy hamlet said:


> This blog claims to be "a 'more objective' take on audiophile topics(...)". Well, I guess everything is relative


 

 Would you care to point out which part you think is wrong?


----------



## RRod

cjl said:


> Would you care to point out which part you think is wrong?


 
  
 Or that is any less objective than the slew of Pono tongue-bathings that are so easy to find these days?


----------



## castleofargh

this is what decided me. 

  
  
 mp3 is better you can scuba dive deeper. next time the dude refuses to get me below 20meters because of my level, I show him that I use mp3.
 I wouldn't like to go flying, I tend to be sick in altitude, and that's a sure way to end up dissected in a lab in area 51.
 QED
  
 I feel like this graph closes up the debate on high res once and for all. thank you pono guys!


----------



## bigshot

That sure tells you who their target audience is! Sesame Street has better science!


----------



## bfreedma

"Pono gives you wings"?.... (at least according to their graphic)
 Crank up the Red Bull copyright infringement team


----------



## Krutsch

castleofargh said:


> this is what decided me.
> 
> 
> 
> ...


 

 I looked for that "graph" on the pono site; where did you find it?


----------



## RRod

Note how Icarus stays away from the sun, lest we find out how fake his wings are.


----------



## bigshot

That image was part of a video they made to promote the Pono.


----------



## sonitus mirus

krutsch said:


> I looked for that "graph" on the pono site; where did you find it?


 
  
 It was originally on their Kick Starter site, but it has been replaced with a different graph.
  
 http://www.innerfidelity.com/content/its-masters-damit


----------



## castleofargh

krutsch said:


> I looked for that "graph" on the pono site; where did you find it?


 

 as bigshot says. I remembered it from one of the silly videos, and just googled pono+underwater in google image hoping somebody would have done a screenshot already ^_^.


----------



## Krutsch

sonitus mirus said:


> It was originally on their Kick Starter site, but it has been replaced with a different graph.
> 
> http://www.innerfidelity.com/content/its-masters-damit


 

 So, I watched that video which was interesting. Thanks for sharing.
 Maybe Sound Science needs to consider replacing the ABX comparator with the head-bop test. 
	

	
	
		
		

		
			




  
 Then, at the 5:00 minute mark, after being passionately lectured about the craft and the artistry, we have the singer from Linkin Park shrieking into a mic. And then, we get to watch Snoop Dogg wax-on all nostalgic about CDs... right, so when you're reading the lyrics, you can really understand the context of what Snoop means when rants about b-i-t-c-h-e-s and hoes.
  
 I hear what my 12 year-old daughter listens to, and it's no better. I try... I've exposed her to a lot of music and I encourage her to use the 2-channel system, so it doesn't sound like crap. I bought her a nice set of Grados to replace the ear buds and sometimes she wants a copy of my graybeard stuff she's just heard for her iTunes library. But as soon as I take my hand off that steering wheel, she's back to Bruno Mars, using the speakers on her laptop.
  
 Really, who on earth is the target for that marketing message?


----------



## castleofargh

krutsch said:


> Maybe Sound Science needs to consider replacing the ABX comparator with the head-bop test.


 
 here is what busta rhymes had to say after listening to the pono in a car(because that's where you can judge real music for a portable DAP... in a car).


----------



## mindbomb

The question has come back to focus because of pono's crappy marketing, but tbh, I kinda don't mind the 24 bit. I have 3 points:
  
 1. The increased dynamic range is nice since 96db on CD is arguably not enough, and you no longer have to use dithering which increases noise.
 2. The size increase is only 50%. With advancements in storage, this is gonna seem like less and less of a big deal. Sizes of albums are still really small compared to good high res video or a game with high res textures.
 3. Why 16 bit lossless in the first place? If you are really a miser with bits, you can get a high quality lossy file that you can't distinguish from the lossless at fraction of the bitrate. And if you encode directly from the 24 bit master, you get better dynamic range too afaik.


----------



## Lespectraal

mindbomb said:


> The question has come back to focus because of pono's crappy marketing, but tbh, I kinda don't mind the 24 bit. I have 3 points:
> 
> 1. The increased dynamic range is nice since 96db on CD is arguably not enough, and you no longer have to use dithering which increases noise.
> 2. The size increase is only 50%. With advancements in storage, this is gonna seem like less and less of a big deal. Sizes of albums are still really small compared to good high res video or a game with high res textures.
> 3. Why 16 bit lossless in the first place? If you are really a miser with bits, you can get a high quality lossy file that you can't distinguish from the lossless at fraction of the bitrate. And if you encode directly from the 24 bit master, you get better dynamic range too afaik.


 
  
 I am not an expert in any of the fields pertaining to audio, so there's that.
  
 Now I'm going to use my knowledge I have gained so far from reading what I have researched, and

 1. Music doesn't go above 60db anyways. 96db is plenty for playback.
 2. The size increase is only 50%, with the sound quality increase of 0%.
 3. 16bit/44.1kHz is the sweet spot for capturing what that matters for playback with transparency and fidelity.
  
 Those points matters for studio use, but has no bearing at all for playback(Digital volume control does benefit from the added bits though).


----------



## castleofargh

Quote:


mindbomb said:


> The question has come back to focus because of pono's crappy marketing, but tbh, I kinda don't mind the 24 bit. I have 3 points:
> 
> 1. The increased dynamic range is nice since 96db on CD is arguably not enough, and you no longer have to use dithering which increases noise.
> 2. The size increase is only 50%. With advancements in storage, this is gonna seem like less and less of a big deal. Sizes of albums are still really small compared to good high res video or a game with high res textures.
> 3. Why 16 bit lossless in the first place? If you are really a miser with bits, you can get a high quality lossy file that you can't distinguish from the lossless at fraction of the bitrate. And if you encode directly from the 24 bit master, you get better dynamic range too afaik.


 
  
 1. as said more dynamic is absolutely not for the song itself, no song uses 90db, it's at best for the noise floor.
 before we had even less than 16bit, like k7 tapes and vinyls. and both were cool enough for sound quality, a little too much distortion, but the most audible and obvious problem with those was the noise. so a way to push that noise down was clearly a good decision. now do we hear the dithered noise floor of a 16bit CD when we listen to our songs? no, else people wouldn't fail ABx so much. I mean if all I had to do was listen for some noise, I clearly shouldn't struggle so much to tell 16/44 apart from 24/96 in an abx. did you try that? to actually judge for yourself how low a noise you can really hear when music is playing at the same time?
 so if it's not for the dynamic of the song and not for the noise, what's the point of adding bits?
  
 2. I agree with you that storage isn't the reason to stay at 16bit. if it was the same price, with the same products available, I might very well have my tracks in 24bit flac on the computer. at least for archive purpose. but we clearly don't live in that world. we live in a world where there is no transparency, and you end up paying more for something you usually know nothing about.
  
 3. why would encoding to mp3@320 from 24bit improve dynamic range?
  
  
  
 most of us here do have some hires files, some audio DVDs some SACDs. we're not anti-hires because our religion told us to, we're really just anti BS. we know that a highres file will measure slightly better than 16/44, at least for the noise floor. we're all just saying that people should concentrate on what we actually hear, instead of buying into marketing.


----------



## lamode

mindbomb said:


> The question has come back to focus because of pono's crappy marketing, but tbh, I kinda don't mind the 24 bit. I have 3 points:
> 
> 1. The increased dynamic range is nice since 96db on CD is arguably not enough, and you no longer have to use dithering which increases noise.
> 2. The size increase is only 50%. With advancements in storage, this is gonna seem like less and less of a big deal. Sizes of albums are still really small compared to good high res video or a game with high res textures.
> 3. Why 16 bit lossless in the first place? If you are really a miser with bits, you can get a high quality lossy file that you can't distinguish from the lossless at fraction of the bitrate. And if you encode directly from the 24 bit master, you get better dynamic range too afaik.


 
  
 1. Really? Are you actually hearing quantization noise? You'd be lucky to have a listening room as quiet as 30dB, which means that if you were listening to music with peaks at 116dB (painful) your quantization noise would still be as quiet as the room background noise.
  
 If you:
 a) have exceptional hearing (very few do) AND
 b) you live in an anechoic chamber (even fewer do) AND
 c) you listen to music with really really quiet passages (only a few classical recordings would qualify) AND
 d) you have your system turned up so loud that the peaks are painful AND
 e) the master itself has a noise floor lower than 96dB
  
 THEN you might hear a little quantization noise, which is still a benign hiss. (and I'm being generous here as there are techniques to achieve over 100dB DR with a CD)
  
 2. True, the 50% extra data of 24 bit is not a BIG deal, but it's still wasted space and space still costs money even if it's not a lot.
  
 3. You won't get an MP3 file with higher DR just because you converted from a 24 bit file.
 16 bit is a good format because it is enough for any domestic listening situation (as I showed above). No-one is choosing it because they are a "miser with bits"


----------



## mindbomb

To the issue of background noise, there are IEMs that provide isolation on that order. But also, I think room noise is usually constant and can be tuned out due to habituation. And also, just because the noise is below the room noise wouldn't mean it is inaudible. For the lossy file with high dynamic range, I'm not sure about mp3, but vorbis and aac are floating point, so they could have more dynamic range than 16 bits, right?


----------



## RRod

mindbomb said:


> To the issue of background noise, there are IEMs that provide isolation on that order. But also, I think room noise is usually constant and can be tuned out due to habituation. And also, just because the noise is below the room noise wouldn't mean it is inaudible. For the lossy file with high dynamic range, I'm not sure about mp3, but vorbis and aac are floating point, so they could have more dynamic range than 16 bits, right?


 
  
 Vorbis and AAC use floating point for their internal calculations, but the PCM signal that is recreated upon decoding will be 16bit if that was the format of the signal that went into it.


----------



## mindbomb

I'd like to add that I compared the size of some lossless 16 bit files to lossless 24 bit files, and it seemed the size difference was about 100%, rather than the 50% it is for uncompressed. I guess it is easier to compress 16 bit files. This raises the stakes beyond what I was expecting. 
  
 The main thing I was trying to say by bringing up lossy is that 16 bit is an arbitrary standard. It seems to me that people in this thread treat it like an ideal value, where everything above 16 bit lossless is a pointless waste of space. What is to stop someone who listens from high quality lossy files from saying that it is the sweet spot and 16 bit lossless is a pointless waste of space? Or that 24 bit is the sweet spot, and DSD is a waste of space? Is there anything particularly special about 16 bit lossless?


----------



## bigshot

Anything beyond the line of audible transparency then.


----------



## castleofargh

mindbomb said:


> I'd like to add that I compared the size of some lossless 16 bit files to lossless 24 bit files, and it seemed the size difference was about 100%, rather than the 50% it is for uncompressed. I guess it is easier to compress 16 bit files. This raises the stakes beyond what I was expecting.
> 
> The main thing I was trying to say by bringing up lossy is that 16 bit is an arbitrary standard. It seems to me that people in this thread treat it like an ideal value, where everything above 16 bit lossless is a pointless waste of space. What is to stop someone who listens from high quality lossy files from saying that it is the sweet spot and 16 bit lossless is a pointless waste of space? Or that 24 bit is the sweet spot, and DSD is a waste of space? Is there anything particularly special about 16 bit lossless?


 

 abx!
 instead of pondering about who has the most complete understanding of digital audio, just listen for yourself outside of biases like already knowing what is playing. and decide for yourself what is or isn't important with only your ears.
  
 if the differences are obvious, then you may ask yourself if they are bad changes or just differences, and decide to pick one format or another depending on space, price, compatibility or just how they sound.
 it's a perfectly rational decision and there is no wrong answer to it.
  
 now when I fail to hear a difference in 99% of the passages I try, even on hand picked critical tracks (high dynamic, calm passages...), then I feel perfectly fine saying that more than 16 bit is a waste of space. I don't need to be reassured about my choices, because I put them to the test, heard no difference and moved on.
  
 I archive in flac 16bit, because it seems logical to have a lossless archive format. it just offers more options for the future, when a new codec will come up or when I will want to hardcode some EQ and replay gain on some tracks for a particular DAP with particular IEMs. reencoding from mp3 may not be really audibly bad, but it's not ideal as time passes. so I keep those in lossless, for the same reason I don't archive my pictures in jpg.
 but all my DAPs are with mp3 320 or vbrmax. because when I'm on the move or in a train, more than mp3 is a waste of space for me ^_^.
 so to answer you question, nothing can stop me! nothing!!!!!! muhahahahaha


----------



## RRod

mindbomb said:


> I'd like to add that I compared the size of some lossless 16 bit files to lossless 24 bit files, and it seemed the size difference was about 100%, rather than the 50% it is for uncompressed. I guess it is easier to compress 16 bit files. This raises the stakes beyond what I was expecting.
> 
> The main thing I was trying to say by bringing up lossy is that 16 bit is an arbitrary standard. It seems to me that people in this thread treat it like an ideal value, where everything above 16 bit lossless is a pointless waste of space. What is to stop someone who listens from high quality lossy files from saying that it is the sweet spot and 16 bit lossless is a pointless waste of space? Or that 24 bit is the sweet spot, and DSD is a waste of space? Is there anything particularly special about 16 bit lossless?


 
  
 There's nothing special about 16bit; it just happens to be the bit-depth that gets the necessary dynamic range for the broad range of music out there. Sometimes for fun (just to show you how weird some of us are), I find the minimal PCM bit-depth and sample rate at which I cannot pass an ABX test of a track vs. the original Redbook version. I have plenty of classical tracks that work fine at 14bit, lots of non-classical tracks that can go down to 12bits, and the occasional brick-wall special that is indistinguishable at 8bits. Since I'm older than some of the folks on here (not bigshot obviously ), I can usually also down-sample to 36kHz or so without affecting quality either.
  
 You can see then, perhaps, why some of us get miffed at the suggestion that people need MOAR bits and samples. If 12/36 in indistinguishable to me from 16/44.1, why on earth would I want 24/192? And if someone else says their ears are much more super than mine, then I am naturally skeptical if they haven't done the same type of experimentation I have with their own hearing. People are all happy to put in the work of buying gear and hi-res material and writing subjective reviews, but they scoff at those of us who go a step further and put in the work of *testing* these things against the limits of our own organism.


----------



## Krutsch

rrod said:


> *There's nothing special about 16bit*; it just happens to be the bit-depth that gets the necessary dynamic range for the broad range of music out there.


 
  
 I have to believe that the adoption of 16-bit was driven by the processors of the era; and 8 bit was obviously too small and 24-bit would take way too much space. Everything else, sampling frequency, playing time, and disc diameter were choices.


----------



## RRod

krutsch said:


> I have to believe that the adoption of 16-bit was driven by the processors of the era; and 8 bit was obviously too small and 24-bit would take way too much space. Everything else, sampling frequency, playing time, and disc diameter were choices.


 
  
 Probably the way to put it is that the ability of processors to handle 16bits meant digital was finally viable, as the requisite dynamic range was finally possible (though I know 14-bits was favored by one company giving input into the mix).


----------



## limpidglitch

rrod said:


> There's nothing special about 16bit; it just happens to be the bit-depth that gets the necessary dynamic range for the broad range of music out there. Sometimes for fun (just to show you how weird some of us are), I find the minimal PCM bit-depth and sample rate at which I cannot pass an ABX test of a track vs. the original Redbook version. I have plenty of classical tracks that work fine at 14bit, lots of non-classical tracks that can go down to 12bits, and the occasional brick-wall special that is indistinguishable at 8bits. Since I'm older than some of the folks on here (not bigshot obviously ), I can usually also down-sample to 36kHz or so without affecting quality either.
> 
> You can see then, perhaps, why some of us get miffed at the suggestion that people need MOAR bits and samples. If 12/36 in indistinguishable to me from 16/44.1, why on earth would I want 24/192? And if someone else says their ears are much more super than mine, then I am naturally skeptical if they haven't done the same type of experimentation I have with their own hearing. People are all happy to put in the work of buying gear and hi-res material and writing subjective reviews, but they scoff at those of us who go a step further and put in the work of *testing* these things against the limits of our own organism.


 
  
 I've been playing around with lower bit-rates as well, more specifically 8/44.1, to see for myself the effects of dither and noise shaping. (linky)
 I must say the results impress me. Although claiming transparency would be to take it a bit too far, seeing how low you can get the noise when having only 256 levels available makes it pretty obvious how when having 65 536 levels available we can make the noise completely disappear.


----------



## lamode

mindbomb said:


> I'd like to add that I compared the size of some lossless 16 bit files to lossless 24 bit files, and it seemed the size difference was about 100%, rather than the 50% it is for uncompressed. I guess it is easier to compress 16 bit files. This raises the stakes beyond what I was expecting.
> 
> The main thing I was trying to say by bringing up lossy is that 16 bit is an arbitrary standard. It seems to me that people in this thread treat it like an ideal value, where everything above 16 bit lossless is a pointless waste of space. What is to stop someone who listens from high quality lossy files from saying that it is the sweet spot and 16 bit lossless is a pointless waste of space? Or that 24 bit is the sweet spot, and DSD is a waste of space? Is there anything particularly special about 16 bit lossless?


 
  
 I already gave you an example which demonstrated quite clearly, I thought, why 16 bit is enough for real life situations. I'm almost certain that you have never heard 16-bit quantization noise in your life and therefore it's just not an issue.


----------



## RRod

limpidglitch said:


> I've been playing around with lower bit-rates as well, more specifically 8/44.1, to see for myself the effects of dither and noise shaping. (linky)
> I must say the results impress me. Although claiming transparency would be to take it a bit too far, seeing how low you can get the noise when having only 256 levels available makes it pretty obvious how when having 65 536 levels available we can make the noise completely disappear.


 
  
 Fixed format PCM is a bit hammer and nail in this regard. There are certainly sections of many tracks that can be successfully captured with fewer than 16 bits or with frequencies lower than 22kHz, but with a fixed format you just go with the flow.


----------



## stv014

mindbomb said:


> But also, I think room noise is usually constant and can be tuned out due to habituation.


 
  
 Dithered quantization noise is also constant, and uncorrelated to the input signal. It is possible to hear a tone at less than -100 dBFS level even after 16-bit quantization noise is added (about -95.8 dBFS A-weighted for simple white noise TPDF dither).
  


mindbomb said:


> I'd like to add that I compared the size of some lossless 16 bit files to lossless 24 bit files, and it seemed the size difference was about 100%, rather than the 50% it is for uncompressed. I guess it is easier to compress 16 bit files.


 
   
Of course it is, because the extra 8 bits added by increasing the sample size to 24 bits usually contain mostly noise that cannot be compressed. It is not uncommon for recordings to have too high analog noise floor even for 16-bit PCM.


----------



## Lespectraal

Glad to see that this person I've been following for years has some sense in him.


----------



## sonitus mirus

lespectraal said:


> Glad to see that this person I've been following for years has some sense in him.


 
  
 This reviewer has a great presentation style to go along with a high quality video production.  I subscribed.  Thanks for sharing.


----------



## Krutsch

sonitus mirus said:


> This reviewer has a great presentation style to go along with a high quality video production.  I subscribed.  Thanks for sharing.


 

 Hmm... this one I won't be sharing with others. Fun to listen to, but isn't in the same league as the earlier example we were discussing, which endeavored to explain *why* you don't need HD audio to a layperson audience.


----------



## limpidglitch

krutsch said:


> Hmm... this one I won't be sharing with others. Fun to listen to, but isn't in the same league as the earlier example we were discussing, which endeavored to explain *why* you don't need HD audio to a layperson audience.


 
  
 Yeah, a bit condescending, without really bringing any information to the discussion. Not much better than his peers on 'the other side'.
 (And stop it with the arms!)


----------



## prot

ffbookman said:


> High end math degrees can't hear better than my ears, or your own ears. If you only have measurements of 1 type of hearing (the frequency range) then you only have math for that 1 domain.
> 
> Timing and spatial recognition in the simulated stereo field is where the major change is, and it's where all the 16bit holdouts like to avoid. The MP3 people outright dismiss all of it. How convenient to throw it away since you don't have math for it.
> 
> ...



I am prolly quite late to this 'party' but I wanna comment on this thing. It is simply *just another one of those "there is voodoo in the audio and only us the initiates can hear it"*
No there isnt. Not since about 1890 when things like waves and electromagnetism were fully explained by solid math and physics. The same math and physics that are behind any audio device ever built. If there was anything missing or wrong with those, it will be missing or wrong in your playback device too. And you wont hear it even if that audio-vodoo existed. So no, there is no audible voodoo anywhere. And btw, that's called logic. Much better than ears . 

Prolly John Dunlavy put it even better when he said 20+ years ago that if anyone discovers and unmeasured effect in audio, he'll get a nobel prize. Didnt happen afaik. 
 The only myth here is that you have golden ears. Is our math perfect? Nope, not even close. Just better than ears. Mine, yours, anyones.
And it's posts like this that actually keep me from buying 24bit music. Normally I don mind buying and using 24bit devices and music. Just dont put that 'it sounds better' bull into it. Noone ever heard that. Anywhere. And many people did test. And test. And nothing.


----------



## prot

And many thanks to user @gregorio for starting this controversial thread and for a very informative first post. As controversial as they are, those things need to be said and discussed and tested. Preferably without preconceptions and without golden ears.


----------



## Greenears

rrod said:


> There's nothing special about 16bit; it just happens to be the bit-depth that gets the necessary dynamic range for the broad range of music out there. Sometimes for fun (just to show you how weird some of us are), I find the minimal PCM bit-depth and sample rate at which I cannot pass an ABX test of a track vs. the original Redbook version. I have plenty of classical tracks that work fine at 14bit, lots of non-classical tracks that can go down to 12bits, and the occasional brick-wall special that is indistinguishable at 8bits. Since I'm older than some of the folks on here (not bigshot obviously ), I can usually also down-sample to 36kHz or so without affecting quality either.
> 
> You can see then, perhaps, why some of us get miffed at the suggestion that people need MOAR bits and samples. If 12/36 in indistinguishable to me from 16/44.1, why on earth would I want 24/192? And if someone else says their ears are much more super than mine, then I am naturally skeptical if they haven't done the same type of experimentation I have with their own hearing. People are all happy to put in the work of buying gear and hi-res material and writing subjective reviews, but they scoff at those of us who go a step further and put in the work of *testing* these things against the limits of our own organism.



What tool do you use to make 14 bit files and what volume level did you listen to when passing the abx? I'm interested to try this test.


----------



## RRod

greenears said:


> What tool do you use to make 14 bit files and what volume level did you listen to when passing the abx? I'm interested to try this test.


 
  
 Doing a non-standard bit-depth is a pain (meaning it takes more than one script, so far), since sox (the tool I normally use) only handles the common values (8, 16, 24, 32). I have to go into something like Matlab or R to do the dither + truncation.
  
 p.s. If anyone knows a gnu or posix command line way of doing this, that would be great!


----------



## limpidglitch

It's easy to do in Audition, but then Audition isn't strictly free.
  
 When you truncate, do you just use a script to remove the last few bits of each word? How do you make sure the file is still playable afterwards?


----------



## RRod

limpidglitch said:


> It's easy to do in Audition, but then Audition isn't strictly free.
> 
> When you truncate, do you just use a script to remove the last few bits of each word? How do you make sure the file is still playable afterwards?


 
  
 Yeah I take the integer wav values, convert to binary character, change the last two bits to 0s (for 16 -> 14 -> 16), then convert back to integers and save the wav. Much more work than I'd like. But for the purposes of illustration here, one can do the 8bit case easily enough with, say, a metal track that lacks a fade-out.
  
 I didn't know Audition had a free version; cool! Still, I'd like a pure command line version at some point. I can probably use some kind of hex editing to do the trick.


----------



## stv014

This simple C program quantizes a 16-bit stereo WAV file to a lower resolution (2 to 15 bits). The dither parameter can be 0 for no dithering, 1 for +/-0.5 LSB uniform distribution dither, 2 for +/-1 LSB white noise TPDF, or 3 for simple "shaped" TPDF that sounds slightly better at 44.1 kHz.
  
 It only works correctly with WAV files that have a header length of 44 bytes, and the sample format must be signed 16-bit PCM, which is also the output format. Only stereo files are supported. The validity of the input format is not verified.
  
 Sample values near the maximum may be clipped, especially at very low resolutions (for example, at 2 bits, the only possible output values are -2, -1, 0, and 1, and positive samples are clipped at -6 dBFS).


----------



## RRod

stv014 said:


> This simple C program quantizes a 16-bit stereo WAV file to a lower resolution (2 to 15 bits). The dither parameter can be 0 for no dithering, 1 for +/-0.5 LSB uniform distribution dither, 2 for +/-1 LSB white noise TPDF, or 3 for simple "shaped" TPDF that sounds slightly better at 44.1 kHz.
> 
> It only works correctly with WAV files that have a header length of 44 bytes, and the sample format must be signed 16-bit PCM, which is also the output format. Only stereo files are supported. The validity of the input format is not verified.
> 
> Sample values near the maximum may be clipped, especially at very low resolutions (for example, at 2 bits, the only possible output values are -2, -1, 0, and 1, and positive samples are clipped at -6 dBFS).


 





 I'll give it a shot, thanks much!
  
 Works like a champ!


----------



## stv014

To round down instead of towards the nearest integer, change line 76

```
*x = (int) (xf + (xf < 0.0 ? -0.5 : 0.5));
```
 to

```
*x = (int) floor(xf);
```
 and add this at line 4:

```
#include <math.h>
```
 This change fixes the clipping issue (or at least improves it in the case of TPDF dither), at the cost of adding a small negative DC offset of -0.5 LSB at the reduced resolution. Non-dithered quantization will simply set the lower bits to zero.


----------



## RRod

OK, I'll add the option for that as an argument for testing. Does the resample executable in your dsputils collection handle non-standard bit-depths as well (or is it only for sample rate conversion)?


----------



## stv014

rrod said:


> OK, I'll add the option for that as an argument for testing. Does the resample executable in your dsputils collection handle non-standard bit-depths as well (or is it only for sample rate conversion)?


 
  
 It is limited to standard formats (8/16/24/32-bit integer, 32/64-bit float), and defaults to float output. The same also applies to the convolve utility.


----------



## mindbomb

One more thing I'd like to bring up: blu-ray movies. It's most common for them to have 24 bit/48khz 5.1 audio, usually with proprietary lossless compression and avg bitrates of around 3 megabit/s. Now, would you guys say that this bit depth is also too high for the same reason as in music? Or are the qualities of movie audio such that more dynamic range is better appreciated?


----------



## lamode

mindbomb said:


> One more thing I'd like to bring up: blu-ray movies. It's most common for them to have 24 bit/48khz 5.1 audio, usually with proprietary lossless compression and avg bitrates of around 3 megabit/s. Now, would you guys say that this bit depth is also too high for the same reason as in music? Or are the qualities of movie audio such that more dynamic range is better appreciated?


 
  
 Yes, for the same reason as music, it is pointless to have a DR so much greater than the S/N ratio once the ambient noise is taken into account. It's around 60dB overkill. Even in a theoretical perfectly quiet environment, the least significant bits are inaudible and even if they were audible, they would be drowned out by the sound of your own heartbeat.


----------



## bigshot

The only reason that movies need bigger files than music is because it is six channels instead of two. But the "bigger files are better" theory holds true even more there, because the image is compressed too. A lot of blu-ray reviewers base their review on the size of the disk. Even if it looks perfect and has no visible artifacting, they still complain that the image could be better because it's a 25 gig blu-ray disk instead of a 50.


----------



## lamode

bigshot said:


> The only reason that movies need bigger files than music is because it is six channels instead of two.


 
  
 That's not the only reason. The audio spec includes up to 24/192 audio.


----------



## Tautologi

I doubt a movie have a dynamic range above 30 dB which really makes me wonder why they have a 24 bit depth.


----------



## castleofargh

bigshot said:


> The only reason that movies need bigger files than music is because it is six channels instead of two. But the "bigger files are better" theory holds true even more there, because the image is compressed too. A lot of blu-ray reviewers base their review on the size of the disk. Even if it looks perfect and has no visible artifacting, they still complain that the image could be better because it's a 25 gig blu-ray disk instead of a 50.


 

 I remember reading an anecdote about the size of the screen making the sound "bigger" on dr olive's blog. (the post where he kind of explained how ignorant we were for still using sighted tests in audio
	

	
	
		
		

		
		
	


	




).


----------



## RRod

tautologi said:


> I doubt a movie have a dynamic range above 30 dB which really makes me wonder why they have a 24 bit depth.


 
  
 I dunno, a 2-channel downmix of my copy of LOTR:ROTK Part II gets a DR rating of 17, which judging by experience would mean a bit more than 30dB of actual range. Still doesn't need 24bits, though.


----------



## lamode

tautologi said:


> I doubt a movie have a dynamic range above 30 dB which really makes me wonder why they have a 24 bit depth.


 
  
 The spec allows for 24/192 but it doesn't require it. There's no reason why most films wouldn't use 16/44 or 16/48 for all channels.


----------



## bigshot

lamode said:


> That's not the only reason. The audio spec includes up to 24/192 audio.


 

 I know that blu-rays support high rate sound. But there is no reason to do that.


----------



## ferday

rrod said:


> I dunno, a 2-channel downmix of my copy of LOTR:ROTK Part II gets a DR rating of 17, which judging by experience would mean a bit more than 30dB of actual range. Still doesn't need 24bits, though.




Isn't the Dolby standard 0 to -31dB? That's going off memory but I think so....


----------



## RRod

ferday said:


> Isn't the Dolby standard 0 to -31dB? That's going off memory but I think so....


 
  
 This is a DTS-HD MA track; dunno what standard it adheres to, but I can't see it being just 31dB… that's like 5bits


----------



## bigshot

Dolby has a million different standards.


----------



## ferday

rrod said:


> This is a DTS-HD MA track; dunno what standard it adheres to, but I can't see it being just 31dB… that's like 5bits :eek:




Can't remember where I read it but I really do recall a maximum dynamic range of 31dB for a Dolby spec...

Edit: here's a mention but really esoteric
http://en.m.wikipedia.org/wiki/Dolby_Digital_Plus


----------



## cspirou

So I read the first post of this really long thread and I think its very convincing argument to stick with 16 bits over 24 bits.

However I believe this is assuming typical playback to passive speakers. What about active speakers using DSP? Mixing and mastering is mentioned as an exception to 24 bit audio. It seems reasonable to me that if you are chopping up the signal for 2 to 4 drivers (or more) that you would want the extra 8 bits of information to be sure you aren't losing anything at the DSP stage. This is a consumer application I can see where 24 bit audio could make a difference.

Maybe someone with a Linkwitz LX521 and a nanoDIGI from miniDSP can test the difference between 24 bit and 16 but audio.

Meanwhile I'll still buy 24 bit audio for the same reason I buy vinyl, because in many cases the better version of the recording is available on 24 bit or vinyl while the CD version gets shafted.


----------



## OddE

bigshot said:


> Dolby has a million different standards.


 
  
 -Which makes perfect sense when your business idea is to license standards...


----------



## analogsurviver

tautologi said:


> I doubt a movie have a dynamic range above 30 dB which really makes me wonder why they have a 24 bit depth.


 
 It depends on the movie. One that definitely is way above 30 dB dynamic range is http://en.wikipedia.org/wiki/Das_Boot The latest digitally remastered (or re-recorded ? )
 soundtrack is guaranteed to have your neighbours banging on the door if you per some miracle possess speaker system capable of giving it justice. It goes from as quiet as possible within a submarine, whispering clearly audible, to the final onslaught of the RAF Mosquitos - only the real thing could be even more frightening. I used headphones - this is no tame Top Gun sound !
  
 There are few Russian films with war theme that come close in realistic sound - but are most definitely not for the faint at heart. 
  
 Peace time movies with great dynamic range not describing some catastrophe could be about some sort or another of motor racing. Of very recent(ish) production.


----------



## RRod

ferday said:


> Can't remember where I read it but I really do recall a maximum dynamic range of 31dB for a Dolby spec...
> 
> Edit: here's a mention but really esoteric
> http://en.m.wikipedia.org/wiki/Dolby_Digital_Plus


 
  
 I get about 60dB or so for the the LOTR track (taking the 99th percentile of 2s RMS values and the max peak). I don't know how LKFS loudness works (I assume it's meant to work in concert with standards for delivery systems in theaters), but it seems like it means more like "you can't have a louder perceived sound than -31dB full scale" rather than a dynamic range measurement.


----------



## limpidglitch

I just watched this film regarding LUFS/LKFS.
 The LKFS mentioned in that Dolby article is probably the integrated perceived loudness, similar to how it's used in radio and TV broadcast.
  
 
  
 If you want the nitty gritty, you can read the BS.1770 ITU recommendations. It's pretty neat.


----------



## mindbomb

ferday said:


> Isn't the Dolby standard 0 to -31dB? That's going off memory but I think so....


 
 iirc, dolby has optional drc info, ala replaygain, so I think that's what you are referring to.


----------



## ferday

Probably ha ha

Memory only goes so far these days


----------



## Greenears

stv014 said:


> This simple C program quantizes a 16-bit stereo WAV file to a lower resolution (2 to 15 bits). The dither parameter can be 0 for no dithering, 1 for +/-0.5 LSB uniform distribution dither, 2 for +/-1 LSB white noise TPDF, or 3 for simple "shaped" TPDF that sounds slightly better at 44.1 kHz.
> 
> It only works correctly with WAV files that have a header length of 44 bytes, and the sample format must be signed 16-bit PCM, which is also the output format. Only stereo files are supported. The validity of the input format is not verified.
> 
> Sample values near the maximum may be clipped, especially at very low resolutions (for example, at 2 bits, the only possible output values are -2, -1, 0, and 1, and positive samples are clipped at -6 dBFS).


 

 Thank you so much for the file - I downloaded it fine. Unfortunately on this network right now I (a) don't have a Unix prompt and (b) don't have gcc installed.  I guess I could download it for Windows, or borrow a friend's system to try.  Not a big barrier I'll try it at some point in the next couple of weeks.
  
 I assume the 44 byte .wav format above is the standard one that comes out of SOX.
  
 I want to experiment by truncating the files from 24 bit down to the N bits where I can get 9/10 on ABX.  Discovering where this "knee of the curve" is on my system will be very instructive.  Then I can try a number of different tracks and see if the DR or other features of the track affect the where the "knee" is for that particular track relative to the reference knee.  It shouldn't be too hard based on the testing I've already done. Once you have this data, trying a new headphone or DAC or other is easier, you can just see if it raises the "knee" rather than the likely "no difference".  This approach actually controls out a lot of variables that cause problems for ABX - the most important of which is that the track itself may not have enough "resolution" to pass ABX no matter what you do.
  
 So although there is no shortage of noise and hyper-opinionism on this forum .... I have found a few nuggets that help move along the actual science of sound.  
  
 This is one!   Thanks again.


----------



## RRod

greenears said:


> This approach actually controls out a lot of variables that cause problems for ABX - the most important of which is that the track itself may not have enough "resolution" to pass ABX no matter what you do.


 
  
 How is this a shortcoming of ABX? You mean that only tracks with a certain actual dynamic range can be successfully ABXed between a certain pair of bit depths?


----------



## Greenears

rrod said:


> I get about 60dB or so for the the LOTR track (taking the 99th percentile of 2s RMS values and the max peak). I don't know how LKFS loudness works (I assume it's meant to work in concert with standards for delivery systems in theaters), but it seems like it means more like "you can't have a louder perceived sound than -31dB full scale" rather than a dynamic range measurement.


 

 I think everyone needs to look at this post and be reminded of something key:  All DR measurements require a window of time and a method of averaging (in this case time window is 2s and the method is 99 percentile etc)  The instantaneous DR (1 sample time) can always be much higher as has been pointed out about 20 pages ago.  I'm told that most modern tracks are normalized to peak around -6 or -3 dBFS you basically could have 90 dB instantaneous DR with redbook. 
  
 The real question I and others are asking is (a) how much of these instantaneous peaks are there (b) are there enough to become audible (c) can you tell the difference if you extend them to 120dB DR (20 bits equivalent).  Another reference pointed out that a short peak of 120 dB is not noticeable, even though longer exposure (seconds?) will cause pain.


----------



## RRod

greenears said:


> I think everyone needs to look at this post and be reminded of something key:  All DR measurements require a window of time and a method of averaging (in this case time window is 2s and the method is 99 percentile etc)  The instantaneous DR (1 sample time) can always be much higher as has been pointed out about 20 pages ago.  I'm told that most modern tracks are normalized to peak around -6 or -3 dBFS you basically could have 90 dB instantaneous DR with redbook.
> 
> The real question I and others are asking is (a) how much of these instantaneous peaks are there (b) are there enough to become audible (c) can you tell the difference if you extend them to 120dB DR (20 bits equivalent).  Another reference pointed out that a short peak of 120 dB is not noticeable, even though longer exposure (seconds?) will cause pain.


 
  
  
 I'd be careful testing that one. What was the definition of instantaneous DR at a sample?


----------



## Greenears

rrod said:


> I'd be careful testing that one. What was the definition of instantaneous DR at a sample?


 






  Both hands firmly over ears ....


----------



## stv014

greenears said:


> Thank you so much for the file - I downloaded it fine. Unfortunately on this network right now I (a) don't have a Unix prompt and (b) don't have gcc installed.  I guess I could download it for Windows, or borrow a friend's system to try.  Not a big barrier I'll try it at some point in the next couple of weeks.


 
  
 It can be built with basically anything that can compile a simple "hello.c" program. A basic installation of MinGW should work. I could also create and upload an .exe version.
  


> I assume the 44 byte .wav format above is the standard one that comes out of SOX.


 
  
 44 bytes is the minimum length of the WAV header. This is a very simple program with no library dependencies, so it does not actually parse the header of the input file, it just copies the first 44 bytes unchanged, and assumes that the sample format is stereo 16-bit PCM. For converting a 24-bit input file, it will need to be modified.


----------



## castleofargh

how do I go about to try that bit extermination when I'm a noob and don't have linux? I did something like that with adobe audition(or was it cool edit?) what seems like an eternity ago, but my years thinking I should crack any software just because I could are gone(like my hair). so if possible I would like to do it legit this time ^_^.
 or have somebody risking jail for me by uploading a few different bitdepth values of some nice dynamic classical track?
	

	
	
		
		

		
		
	


	




 (is it legit if I'm not the one going to jail?)
  
  
  
 Quote:


greenears said:


> rrod said:
> 
> 
> > I get about 60dB or so for the the LOTR track (taking the 99th percentile of 2s RMS values and the max peak). I don't know how LKFS loudness works (I assume it's meant to work in concert with standards for delivery systems in theaters), but it seems like it means more like "you can't have a louder perceived sound than -31dB full scale" rather than a dynamic range measurement.
> ...


 

 I think the video posted by limpidglitch makes the overall DR value vs fullscale peak, clear to people who are not familiar with all kinds of DBs and measurements.
 the -3db or more I believe (but pro dudes may have more ideas about that), is a simple safety against clipping in general. I know that I've had some mp3s that seemed to clip, and lowering the gain (whatever the way) removed that feeling. maybe one day I'll remove the gain values of everything just to find out what tracks seemed to do that an analyze the crap out of them.
 anyway if I was making a track, I wouldn't want people to think it's crap simply because they have a DAC with too much voltage output, or some upper rounding math trick clipping stuff close to zero db when oversampling, or just a guy who doesn't know how to use an EQ. better safe than sorry.


----------



## stv014

This updated version of the quantize program includes a Win32 executable, and supports 24-bit PCM samples (files created by sox and the dsputils programs should work). When processing 24-bit input, the output file is always in the same format, so it does waste some space. Another change is that adding 4 to the dither type switches from rounding towards the nearest integer to floor rounding.


----------



## RRod

stv014 said:


> This updated version of the quantize program includes a Win32 executable, and supports 24-bit PCM samples (files created by sox and the dsputils programs should work). When processing 24-bit input, the output file is always in the same format, so it does waste some space. Another change is that adding 4 to the dither type switches from rounding towards the nearest integer to floor rounding.


 
  
 \m/
	

	
	
		
		

		
		
	


	




\m/


----------



## unshavenbastard

> And 1GHz is a much higher frequency than 20kHz, but so what? Your ears can't hear it and your ears also can't hear sounds at -144dB, which are represented by the 24th bit in digital audio. Your ears can't even hear the 18th bit. This is all wishful thinking, and proven as such in controlled tests.


 

 Inaccuracies like these is what irritates me about this thread.
 The original poster made the same mistake, claiming all there was to 24 vs. 16 bits was dynamical _range._
 Well from what it looks to me anyway. I'll try to explain, maybe I myself misunderstood something.

 People here seem to confuse dB with dB-*audio*, the latter being an absolute measure (because it has an absolute reference, not depending on device), the former not.
 dB just says how much louder it is compared to a reference level, the dynamic range of e.g. a soundcard being how much louder the loudest value is from the most quiet one.
 But this could be scaled / mapped onto a differently spaced grid(e.g. when you fumble with the volume dial of your amp), not changing the dynamic range being used, as not only the loudest amplitude of a recording will be louder, but also the most quiet one (so the absolute audio level range changes, of course).
 E.g.: if your DAC has a nominal dynamic range of 8bits or 48dB let's say outputting 1mV..255mV, and you externally amplify so that you get values from 25.5mV..6553.5mV, it's still 48dB, but your speakers put it into a different dbA range, and onto a different place within the human hearing range.

 Nobody mentioned the one thing that would be deciding whether more bits can offer more _perceived_ quality or not: The _resolution_ _of the ear_ itself. And it is indeed about _resolution_, not just range.
 You could make a DAC which can produce only two different output levels, and space those such that they span 100 dB, so you have a "dynamic _range_ of 100dB", but only two steps, which is obviously useless. It's about how many different amplitude values within the healthy dynamic range of the ear the latter can discern, similar to color formats' bit depths and the number of different shades of colors the eye can discern.
 So more finely grained steps along the amplitude axis can matter, as long as it does not exceed the number of amplitude steps the ear can discern within the region of the overall hearing range of sound pressures that this granularity is mapped onto.
 If, hypothetically, the ear could discern 1 bilion different audio levels within the 140 or so dB*Audio* level range of the ear, you could easily map those 144 dB range of a 24bit DAC onto, say, the center third of the ear's dbA range, and the ear could still discern all the steps - i.e. if there were less steps resolution there, the ear might notice the equivalent of visual banding in low number of color shades images!
  
 Note that I am not making any specific claims about bit rates and their supposed advantages here.
 Just generally speaking, the explanations given in this thread so far seem inaccurate.

 From what I remember (can't find a reference) the ear actually has a _resolution_ of about 17 bits, which would mean it could discern twice as many different audio levels than 16bit wave can. So 24 bits would be 128 times overkill - assuming that during playback it is all amplified perfectly into just the sound pressure range of hearing where it matters.
 How this all really works out when people playback their music, I'm not sure ^^
 .


----------



## castleofargh

unshavenbastard said:


> > And 1GHz is a much higher frequency than 20kHz, but so what? Your ears can't hear it and your ears also can't hear sounds at -144dB, which are represented by the 24th bit in digital audio. Your ears can't even hear the 18th bit. This is all wishful thinking, and proven as such in controlled tests.
> 
> 
> 
> ...


 

 no you have the wrong idea about resolution or increased precision. well you're right if we only look at values for being values, so on the digital side, you're right.
 but wrong when you look at it as soundwaves. when you end up with the DAC making the sine wave to output the analog signal, it will be a continuous signal so it won't matter that you have fine tuned each sample or not with a precision up to 17 or 24bit. the maximum difference between the wave generated from the 24bit signal and the 16bit signal will be some noise 16bit down and below, as that's the LSB in 16bit and de facto, the biggest error in precision you should expect(at least in theory).

 meaning that of course you're increasing the precision of the analog signal by using 24bit, but the difference is audible only below -96db. and it works like that in part because of how waves can mix together. the wrong signal can be seen as the wrong signal, or as the right signal+some added noise. looking at it that way, you see that you end up with the right signal from 0 to -96db, and then you have a mess as loud as the errors from the DAC(or signal's bitdepth) so as loud as 16bit down = -96db noise. and that's quantization noise. we're back on our feet.


----------



## RRod

unshavenbastard said:


> [text]
> .


 
  
 As far as dynamic range, it is intimately related to the number of bits, as a lower number of bits simply cannot represent the same minimal and maximal waveforms that a higher number of bits can (techniques like noise-shaping notwithstanding). Take 16 vs. 8 bits and square waves. The maximal amplitude symmetric (around 0) square wave you can get at 16bits has its peaks at ±32767, while the minimal has peaks at ±1. That's a 90dB difference in amplitude. At 8bits, you get maximal peaks at {0,254} and minimal at {126,128}, which is a 42dB difference in amplitude. So we can get 48dB more difference between peak values by going to 16bit for this waveform.
  
 Resolution, as you are using it, would come down to being able to detect the smallest change possible for a given signal. So for square waves that would mean, at 16bit, detecting the difference between ±32767 and ±32766. Maybe I'll make that example.


----------



## castleofargh

rrod said:


> unshavenbastard said:
> 
> 
> > [text]
> ...


 

 and I was afraid my explanation wouldn't be clear enough for him ^_^.


----------



## RRod

castleofargh said:


> and I was afraid my explanation wouldn't be clear enough for him ^_^.


 
  
 I'm not all good with the baking analogies as you are 
  
 Here are two files at 16/44100. One is a square wave with amplitude 255; the other with amplitude 254 (software can easily distinguish them; they are about 0.03dB RMS apart).
 https://drive.google.com/file/d/0BwmVtb5IwniESGV5ODdvZ3pROUk/view?usp=sharing
 https://drive.google.com/file/d/0BwmVtb5IwniES3RNakVoLV9CUmc/view?usp=sharing


----------



## lamode

unshavenbastard said:


> Inaccuracies like these is what irritates me about this thread.
> The original poster made the same mistake, claiming all there was to 24 vs. 16 bits was dynamical _range._
> Well from what it looks to me anyway. I'll try to explain, maybe I myself misunderstood something.
> 
> ...


 
  
 You are theoretically correct about the distinction between resolution and DR. I used to try and explain this to photographers who thought that a 16-bit/channel image had higher DR than the same file converted to 8-bit. No, the DR was the same but the increments were less smooth.
  
 But... all of this becomes moot as no proper double blind test has ever revealed a single test subject who could tell the difference between a live audio feed, and the same feed fed through a 16/44 AD-DA loop, which is evidence enough for me that although many problems exist in theory, in real life they are inaudible.


----------



## Greenears

stv014 said:


> This updated version of the quantize program includes a Win32 executable, and supports 24-bit PCM samples (files created by sox and the dsputils programs should work). When processing 24-bit input, the output file is always in the same format, so it does waste some space. Another change is that adding 4 to the dither type switches from rounding towards the nearest integer to floor rounding.


 

 Cool - thanks I downloaded it.  I don't have time now but I'll try it out later.


----------



## Greenears

lamode said:


> You are theoretically correct about the distinction between resolution and DR. I used to try and explain this to photographers who thought that a 16-bit/channel image had higher DR than the same file converted to 8-bit. No, the DR was the same but the increments were less smooth.
> 
> But... all of this becomes moot as no proper double blind test has ever revealed a single test subject who could tell the difference between a live audio feed, and the same feed fed through a 16/44 AD-DA loop, which is evidence enough for me that although many problems exist in theory, in real life they are inaudible.


 
 All this back and forth on DR can be settled by this new utility posted.  Simply keep chopping of a few bits at a time and till you pass the ABX test, then run the track through a DR analyzer.
  
 I think people will find the standard equation of DR = N bits * 6 in dB only works at the limit where DR is at the max of N*6, but not at lower values.  You'll be able to hear noise at levels lower than the equation predicts.
  
 But .... I've been wrong before :--) Or have I?


----------



## castleofargh

greenears said:


> lamode said:
> 
> 
> > You are theoretically correct about the distinction between resolution and DR. I used to try and explain this to photographers who thought that a 16-bit/channel image had higher DR than the same file converted to 8-bit. No, the DR was the same but the increments were less smooth.
> ...


 

 you just gave me the audiophile idea of 2015. making a box that you plug between the DAC and the amp that will add tape hiss. you will be able to set the value so that it's slightly above both the amp noise floor and the DAC bit depth. the end of THD, IMD etc, pure hiss over all frequencies just a tad louder to cover it. analog dithering!!!!!!!!!! 
 I guess 3500$ the box is a good starting price? maybe I should really put a tape in it so people would have to turn it from time to time like a true audiophile. and then we launch a market for different tapes with different hisses. hiss rolling is the new tube!!!!!!!!!
 man I'm on fire. and the worst part is that with an ok marketing that's the kind of stuff that could probably sell and make the sound more "natural".


----------



## stv014

> Originally Posted by *unshavenbastard* /img/forum/go_quote.gif
> 
> You could make a DAC which can produce only two different output levels, and space those such that they span 100 dB, so you have a "dynamic _range_ of 100dB", but only two steps, which is obviously useless. It's about how many different amplitude values within the healthy dynamic range of the ear the latter can discern, similar to color formats' bit depths and the number of different shades of colors the eye can discern.


 
  
 Actually, it is possible to implement high quality audio playback with only two output levels. That is how DSD works, and most modern DACs also have only a few bits of output resolution when playing PCM. The use of dithering (which can be psycho-acoustically optimized with noise shaping, so that most of the quantization noise is at frequencies where the threshold of hearing is high) allows for encoding any fractional level at the cost of adding noise to the signal.
  
 The dynamic range of an audio device is the ratio of the highest output level it is capable of without clipping and the noise floor. Therefore, if the dithering is well implemented, the equation of 'X bits = X * 6.02 + Y dB of dynamic range' (Y is a constant depending on the dithering/noise shaping used, and possibly other factors like the use of A-weighting, and the sample rate) is indeed generally correct for high enough values of X. It can be proven that by adding triangular distribution white noise with a peak to peak level of 2 LSB before the quantization, as long as clipping is avoided, any input level will result in the same output level on average, with the addition of white noise at a constant 0.5 LSB RMS level.
  


> Originally Posted by *unshavenbastard* /img/forum/go_quote.gif
> 
> From what I remember (can't find a reference) the ear actually has a _resolution_ of about 17 bits.


 
  
 It was probably calculated from an assumed "good enough" dynamic range. Other than that, the concept of "bits of resolution" in the time domain cannot really be applied to hearing, as audio is not perceived in PCM format. However, the smallest audible change in the amplitude of a tone is actually about 1%, or 0.1 dB. Lossy compression algorithms successfully take advantage of this limited resolution by quantizing (after normalization) the signal in the frequency domain, even to a very low number of bits when masking makes it possible without audible artifacts.


----------



## FFBookman

I've read more this huge thread, and not a single one of you seem to talk about stereo. Soundstage. Room ambience.
  
 You all live in mono test tone land. Strange place that is, and nothing like how we actually enjoy sound in the form of music, stereo music, as heard binaurally.
  
 That's the huge blind spot in DSP, and why 24bit played on a capable system outshines 16bit. You guys don't live in the world of stereo music enjoyment or creation, where timing and depth is very much based on spatial awareness, not frequency awareness.


----------



## cjl

ffbookman said:


> I've read more this huge thread, and not a single one of you seem to talk about stereo. Soundstage. Room ambience.
> 
> You all live in mono test tone land. Strange place that is, and nothing like how we actually enjoy sound in the form of music, stereo music, as heard binaurally.
> 
> That's the huge blind spot in DSP, and why 24bit played on a capable system outshines 16bit. You guys don't live in the world of stereo music enjoyment or creation, where timing and depth is very much based on spatial awareness, not frequency awareness.


 

 OK, I'll indulge you.
  
 How does soundstage and room ambience require >100dB of dynamic range or frequencies above 20kHz?


----------



## Tuco1965

ffbookman said:


> I've read more this huge thread, and not a single one of you seem to talk about stereo. Soundstage. Room ambience.
> 
> You all live in mono test tone land. Strange place that is, and nothing like how we actually enjoy sound in the form of music, stereo music, as heard binaurally.
> 
> That's the huge blind spot in DSP, and why 24bit played on a capable system outshines 16bit. You guys don't live in the world of stereo music enjoyment or creation, where timing and depth is very much based on spatial awareness, not frequency awareness.


 
  
 What does that even mean?


----------



## mindbomb

I still have a few questions. There is a lot of talk about dithering with 16 bit, but also a lot of suggestion that 96db is enough dynamic range. This seems like a contradiction. What's the deal, why even dither? Furthermore, is noise shaping generally implied when talking about dithering? It seems like free performance, but I don't hear it mentioned as much.


----------



## The Walrus

mindbomb said:


> I still have a few questions. There is a lot of talk about dithering with 16 bit, but also a lot of suggestion that 96db is enough dynamic range. This seems like a contradiction. What's the deal, why even dither? Furthermore, is noise shaping generally implied when talking about dithering? It seems like free performance, but I don't hear it mentioned as much.


 
 http://xiph.org/video/vid2.shtml
  
 Maybe this can help.


----------



## cjl

Dithering with noise shaping is unnecessary with 16 bit typically, yes, since even the quantization noise is inaudible, but there's really no reason not to do it. It eliminates signal correlated noise, and it decreases the audibility of the noise to even further below audibility, and it does this without any impact on file size or playback requirements.


----------



## FFBookman

cjl said:


> OK, I'll indulge you.
> 
> How does soundstage and room ambience require >100dB of dynamic range or frequencies above 20kHz?


 

 I don't think it does. I think this is in the time domain, not the frequency domain.
  
 Clock jitter is a suspect, perhaps the prime suspect.
  
 Dither is a suspect.
  
 Aliasing is a suspect.
  
 But only when applying to the 2-channel stereo mix of music. Not sine waves. Not mono. Not test tones. It won't show up there.
  
 It won't show up in random ABX tests either, due to the confusion and false positives.
  
 It simply doesn't display well in 2D space so it won't show up on your scopes. BTW the scopes aren't even at 24bit!
  
 It's the time and space of music imparted by the musician, the instrument, the room it was recorded in, the mix engineer, and the mastering engineer.
  
 A lot of it is not in the frequency domain, at least not solely in the frequency domain.


----------



## RRod

ffbookman said:


> But only when applying to the 2-channel stereo mix of music. Not sine waves. Not mono. Not test tones. It won't show up there.


 
  
 I can take great 2-channel albums of mine, down-convert them to something like 14bit/39ksamples, and still not here any differences.


----------



## FFBookman

rrod said:


> I can take great 2-channel albums of mine, down-convert them to something like 14bit/39ksamples, and still not here any differences.


 

 Interesting, I don't know, I can't speak to what you are hearing. There's still plenty of variables once it gets out of the DAC.
  
 I respect the range of people's hearing abilities. I don't think we have answers for who can hear what, at what particular resolution, based on what particular playback system, using what source. I think it's a silly endeavor.  Perhaps the odd shape of my head or the floppiness of my ears give me different spatial recognition abilities than you.
  
 I don't know if I'd pass your test either. 
  
 I'd rather just take what has been a studio standard for almost 20 years now and push for it's availability to consumers. Most professionals were recording at 24/88 by the mid 90's, or to analog tape. 
  
 If we could stream 24bit audio most people would. It's still about convenience and file size when you go wireless.


----------



## RRod

ffbookman said:


> Interesting, I don't know, I can't speak to what you are hearing. There's still plenty of variables once it gets out of the DAC.
> 
> I respect the range of people's hearing abilities. I don't think we have answers for who can hear what, at what particular resolution, based on what particular playback system, using what source. I think it's a silly endeavor.  Perhaps the odd shape of my head or the floppiness of my ears give me different spatial recognition abilities than you.
> 
> ...


 
  
 Well it's very different arguments to say "people can hear what 24/96 can offer" versus "we might as well just not down-covert." I'm pretty much down with the latter, but then music should cost LESS, not more. Not that doing a Redbook conversion takes any effort, but still, what EXTRA are they doing to charge more for just recording at higher specs? Same thing with not applying loudness-war type processing: so you don't have to do the extra work needed to f@#$ up the album, and then you charge more for it? The whole economic side of hi-res is just too infuriating to me as it stands.


----------



## FFBookman

rrod said:


> Well it's very different arguments to say "people can hear what 24/96 can offer" versus "we might as well just not down-covert." I'm pretty much down with the latter, but then music should cost LESS, not more. Not that doing a Redbook conversion takes any effort, but still, what EXTRA are they doing to charge more for just recording at higher specs? Same thing with not applying loudness-war type processing: so you don't have to do the extra work needed to f@#$ up the album, and then you charge more for it? The whole economic side of hi-res is just too infuriating to me as it stands.


 

 Can't argue with any of that. I agree totally. My anger is towards the down-conversion, and the constant attempts at charging us for less than the original.
  
 As far as the "loudness wars", it's not a single thing that can be ended easily. I argue that the loudness wars really started with redbook. The time-bandwidth constraint of redbook along with the feedback-free lowend and dust free high end allowed far more aggressive limiting than vinyl or cassette.
  
 Things got louder and louder through the 80's and 90's as the digital noise floor was so low, and multi tracking increasing with CPU power.
  
 By the late 90's people were listening to low-quality mp3 and getting DAW's on their laptops. DAW's could apply parallel compression at scales that were never seen before. They could sidechain devices in ways the analog world could not. They could route tracks through many more inserts and busses than even the largest analog studios.
  
 Every step of the way, louder. Each new plugin version pumped up more. Each new track has compressors pumped by default. Throw in the 'master it yourself' plugins that add a ton more multi band limiting and expanding and you have no more air. Gone. Never there in the first place.


----------



## mindbomb

cjl said:


> Dithering with noise shaping is unnecessary with 16 bit typically, yes, since even the quantization noise is inaudible, but there's really no reason not to do it. It eliminates signal correlated noise, and it decreases the audibility of the noise to even further below audibility, and it does this without any impact on file size or playback requirements.


 
  
 For uncompressed files, yes, dithering has no impact on size. However, I am curious if it effects the compression of lossless codecs. Also, for the noise shaping question, I've read that it can be harmful in certain cases, I presume if an unshaped dither is added downstream. So I think that is why dithering when going to 16 bit is considered a standard process and noise shaping is not.


----------



## bigshot

ffbookman said:


> I've read more this huge thread, and not a single one of you seem to talk about stereo. Soundstage. Room ambience.
> That's the huge blind spot in DSP, and why 24bit played on a capable system outshines 16bit. You guys don't live in the world of stereo music enjoyment or creation, where timing and depth is very much based on spatial awareness, not frequency awareness.


 
  
 I talk about physical soundstage and room acoustics all the time. The room is as important a part of clear soundstage as the speakers are, and the proper DSP can make it even better. But secondary depth cues are a function of the original miking and mixing, not the bit depth of the playback file. As long as a file format is audibly transparent, which I've found even high rate lossy is, then the speakers, room and DSP have everything they need to create an incredibly vivid and defined soundstage.


----------



## bigshot

ffbookman said:


> I don't think it does. I think this is in the time domain, not the frequency domain.
> Clock jitter is a suspect, perhaps the prime suspect.
> Dither is a suspect.
> Aliasing is a suspect.
> ...


 
  
 I'm sorry, but you're just throwing out completely irrelevant points there. I thought you were going to talk about room acoustics and speaker placement. That's actually something that can have a great effect on soundstage. Jitter, aliasing, ABX testing, in the frequecy domain but not entirely in the frequency domain? It sounds like you're free associating on terms that are used by people who almost know what they're talking about. Do you actually have a suggestion for how soundstage works and how people with home stereos can achieve it, or are you just filling up a bucket with dung and flinging it?


----------



## bigshot

ffbookman said:


> Can't argue with any of that. I agree totally. My anger is towards the down-conversion, and the constant attempts at charging us for less than the original.


 
  
 A CD is a LOT closer to the original than LPs ever were.


----------



## RRod

mindbomb said:


> For uncompressed files, yes, dithering has no impact on size. However, I am curious if it effects the compression of lossless codecs. Also, for the noise shaping question, I've read that it can be harmful in certain cases, I presume if an unshaped dither is added downstream. So I think that is why dithering when going to 16 bit is considered a standard process and noise shaping is not.


 
  
 I took my latest recording of Stravinsky's Firebird, appended to 24bit, dithered with noise shaping back down to 16, then converted both WAVs to FLAC. The original compresses down to 33.6% and the dithered version down to 37.6%. So yes, a bit of effect but nothing mind-boggling for this track.


----------



## castleofargh

ffbookman said:


> cjl said:
> 
> 
> > OK, I'll indulge you.
> ...


 

 looks to me like you have an idea and only then fish for some random technical stuff to blame for it.
 how many of those suspects did you actually experiment on?
  
 yes sound is amplitude per time per channel in our gears. many stuff can go wrong but what values are really audible when listening to music at 80 or 85db loud? are stuff 2db loud so very important to you knowing that no music content is recorded that low on the album?get out of your general concepts and think about reality, values, and magnitude.  do you hear somebody fart quietly when the music is loud EDM in a discotheque and everybody's jumping, or do you need the bias of other senses like smell and the guilty look on someone's face to actually notice? just like you need to know you listen to highres to hear your stuff because the sound wasn't really what betrayed the bad guy?
 your post suggests that you didn't really understand digital audio and that from that point you made(or were told) theories based on false hypothesis. what do you think of the xiph videos? what in them doesn't go with your own understanding of digital audio?
  
 I'll take the worst example I can think of, to show in practice that most of your points can't be the problem you think they may be: your honor I call to the stand my nemesis, vinyl!
 high noise, high distortion, time error so big it has it's own name, crosstalk very poor and a great deal of mono in it, rolled off trebles with unreliable amplitudes. yet it can sound really nice and "natural", and spacious, with a good sense of position cues. if dither or jitter or 16bit were guilty then none those feelings should show up on vinyls, soundstage and stuff should be a total mess as the specs you talk about are much worst than on CD.
 see it's not hard to be rational and draw conclusions to eliminate a few suspects. even if you're apparently also against ABX for again something that didn't really pass the conceptual stage, there are often ways to compare things for what they are and tell if an idea might be true or if it's just wind.


----------



## analogsurviver

castleofargh said:


> looks to me like you have an idea and only then fish for some random technical stuff to blame for it.
> how many of those suspects did you actually experiment on?
> 
> yes sound is amplitude per time per channel in our gears. many stuff can go wrong but what values are really audible when listening to music at 80 or 85db loud? are stuff 2db loud so very important to you knowing that no music content is recorded that low on the album?get out of your general concepts and think about reality, values, and magnitude.  do you hear somebody fart quietly when the music is loud EDM in a discotheque and everybody's jumping, or do you need the bias of other senses like smell and the guilty look on someone's face to actually notice? just like you need to know you listen to highres to hear your stuff because the sound wasn't really what betrayed the bad guy?
> ...


 
 Now, can you please elaborate on time error in vynil ?


----------



## lamode

ffbookman said:


> I don't think it does. I think this is in the time domain, not the frequency domain.
> 
> Clock jitter is a suspect, perhaps the prime suspect.
> 
> ...


 
  
 Actually some scopes ARE 24 bit, e.g. http://spectraplus.com/
  
 Clock jitter is inaudible in most commercial devices, and in most cases would still be inaudible even with 100x more jitter. I have posted the results of a jitter study elsewhere. All of these things which you mention are 'suspect' do exist, but they exist as inaudible noise.
  
 If they "don't show up in ABX tests" then by definition they are inaudible.


----------



## lamode

ffbookman said:


> I'd rather just take what has been a studio standard for almost 20 years now and push for it's availability to consumers. Most professionals were recording at 24/88 by the mid 90's, or to analog tape.


 
  
 Well analogue tape, even at 15ips and on the best available machine gives you around 72dB DR, equivalent to 12 bits only. So CD is a step up from analogue master tape.
  
 And please understand that the extra bits in a studio are not their for playback SQ, they are there to allow an engineer to add headroom, so he can peak the levels at -6dB instead and have plenty of headroom. 24 bit also allows for mixing artefacts to be kept to a minimum.


----------



## castleofargh

analogsurviver said:


> Now, can you please elaborate on time error in vynil ?


 
 wow and flutter 
	

	
	
		
		

		
			





.


----------



## limpidglitch

mindbomb said:


> For uncompressed files, yes, dithering has no impact on size. However, I am curious if it effects the compression of lossless codecs. Also, for the noise shaping question, I've read that it can be harmful in certain cases, I presume if an unshaped dither is added downstream. So I think that is why dithering when going to 16 bit is considered a standard process and noise shaping is not.


 
  
  
 "Each noise shaper changes the white spectrum of TPDF dithering noise at the expense of increasing the total noise power. Usually the noise is shifted into high-frequency bands, so at the output we get a high-frequency noise with a significant power. If the power of high-frequency noise becomes too high, it can damage some audio gear (e.g. tweeters at high listening levels). Also high power HF-noise can degrade the quality of interpolation of audio data in CD-players, which cannot correctly read digital data from CD medium and try to interpolate invalid samples. There is no standard for the maximal dithering noise power but we propose to use a reasonable threshold of -60 dB FS for noise peak values. It prevents most VU meters with an amplitude range of 60 dB from flickering when playing noise shaped recordings."
  
 From this RMAA article.
  
 And dither should not be applied when bouncing tracks between software. _Only_ when exporting the end master for distribution to listeners should dither be used.


----------



## lamode

limpidglitch said:


> And dither should not be applied when bouncing tracks between software. _Only_ when exporting the end master for distribution to listeners should dither be used.


 
  
 +1


----------



## analogsurviver

castleofargh said:


> wow and flutter
> 
> 
> 
> ...


 
 The same as if I lumped (almost) everything I do not like in digital in "digital" 
	

	
	
		
		

		
		
	


	




.
  
 Yet I had to learn at least basic about sampling frequency, bit depth, dithering, jitter, etc, etc.
  
 "Wow and flutter" in vinyl replay has at least as many "faces" as digital and can not be lumped into a single number without knowing its origins/causes - and these , although anything but easy, can be brought down to inaudible level. But not to  figures obtainable by digital.


----------



## castleofargh

analogsurviver said:


> castleofargh said:
> 
> 
> > wow and flutter
> ...


 

 eheh, sure I just went with some nasty over exaggerated examples because I'm that kind of bad boy, and talking about jitter abx or stereo separation perception wouldn't have worked on my anti ABX client.


----------



## analogsurviver

castleofargh said:


> eheh, sure I just went with some nasty over exaggerated examples because I'm that kind of bad boy, and talking about jitter abx or stereo separation perception wouldn't have worked on my anti ABX client.


 
 I am not that anti ABX - what I am really against is its use as the sole final arbiter.


----------



## castleofargh

limpidglitch said:


> mindbomb said:
> 
> 
> > For uncompressed files, yes, dithering has no impact on size. However, I am curious if it effects the compression of lossless codecs. Also, for the noise shaping question, I've read that it can be harmful in certain cases, I presume if an unshaped dither is added downstream. So I think that is why dithering when going to 16 bit is considered a standard process and noise shaping is not.
> ...


 

 that's why my spanish tape hiss box would be right after the DAC, because you need to do those stuff last.


----------



## castleofargh

analogsurviver said:


> castleofargh said:
> 
> 
> > eheh, sure I just went with some nasty over exaggerated examples because I'm that kind of bad boy, and talking about jitter abx or stereo separation perception wouldn't have worked on my anti ABX client.
> ...


 

 not you, my post was about bookman. sorry for the messy confusion.


----------



## analogsurviver

castleofargh said:


> not you, my post was about bookman. sorry for the messy confusion.


 
 Thank you for the clarification - I may have not followed the last posts that attentively either.


----------



## Krutsch

castleofargh said:


> that's why my spanish tape hiss box would be right after the DAC, because you need to do those stuff last.





 Clever... what show is that from?


----------



## castleofargh

krutsch said:


> Clever... what show is that from?


 

 it's called IT crowd. the UK one(even though you have the same guy on the US attempt at importing british humor).


----------



## mindbomb

limpidglitch said:


> And dither should not be applied when bouncing tracks between software. _Only_ when exporting the end master for distribution to listeners should dither be used.


 
  
 Well, yea, ideally, but sometimes the audio playing software itself adds a dither. Like windows 7 if set to a 16 bit default format. I imagine in a browser like chrome, the html5 player outputs 16 bit audio with a dither if you were to do a software volume change. So in these cases, there would be no benefit to using noise shaping, since low frequency noise removed by the noise shaping would later be re-added.


----------



## Greenears

ffbookman said:


> I've read more this huge thread, and not a single one of you seem to talk about stereo. Soundstage. Room ambience.
> 
> You all live in mono test tone land. Strange place that is, and nothing like how we actually enjoy sound in the form of music, stereo music, as heard binaurally.
> 
> That's the huge blind spot in DSP, and why 24bit played on a capable system outshines 16bit. You guys don't live in the world of stereo music enjoyment or creation, where timing and depth is very much based on spatial awareness, not frequency awareness.


 
  
 Not correct.  All my testing, and those of many others, was in 24 bit and 16 bit stereo.


----------



## Greenears

ffbookman said:


> Interesting, I don't know, I can't speak to what you are hearing. There's still plenty of variables once it gets out of the DAC.
> 
> I respect the range of people's hearing abilities. I don't think we have answers for who can hear what, at what particular resolution, based on what particular playback system, using what source. I think it's a silly endeavor.  Perhaps the odd shape of my head or the floppiness of my ears give me different spatial recognition abilities than you.
> 
> ...


 
  
 That's actually a deep thought, and really hard to find a logical argument against it. File size can really be the only complaint, but the reality is that for audio file size is trivial for
 in-home use. It's only for mobile phones and players, and frankly by the end of this year 128G will be standard with expansion to 384G and more, so the file size argument is fading.  Let the user do the conversion and compression. 
  
 Frankly stuff produced in the last decade probably the native format of the master is 32 bit float.  You could even argue to just give us that for download.  More can be done to give the end user control over compression and other effects to better match their playback environment.


----------



## Greenears

rrod said:


> Well it's very different arguments to say "people can hear what 24/96 can offer" versus "we might as well just not down-covert." I'm pretty much down with the latter, but then music should cost LESS, not more. Not that doing a Redbook conversion takes any effort, but still, what EXTRA are they doing to charge more for just recording at higher specs? Same thing with not applying loudness-war type processing: so you don't have to do the extra work needed to f@#$ up the album, and then you charge more for it? The whole economic side of hi-res is just too infuriating to me as it stands.


 
  
 Hear Hear!


----------



## Greenears

lamode said:


> *Well analogue tape, even at 15ips and on the best available machine gives you around 72dB DR, equivalent to 12 bits only. So CD is a step up from analogue master tape.*
> 
> And please understand that the extra bits in a studio are not their for playback SQ, they are there to allow an engineer to add headroom, so he can peak the levels at -6dB instead and have plenty of headroom. 24 bit also allows for mixing artefacts to be kept to a minimum.


 
  
 This type of analysis that 72 dB = 12 bit (6 * N bits) is so often repeated on forums.  But I don't think it is true. If you do the exercise and ABX a 12 bit vs 16 bit version from tape you will hear the difference.  Luckily I am now in possession of a program to make this test easy, so I can try it on some albums that were originally all tape.  Rumors is supposed to be well recorded, I can try some others from the era.


----------



## FFBookman

bigshot said:


> I talk about physical soundstage and room acoustics all the time. The room is as important a part of clear soundstage as the speakers are, and the proper DSP can make it even better. But secondary depth cues are a function of the original miking and mixing, not the bit depth of the playback file. As long as a file format is audibly transparent, which I've found even high rate lossy is, then the speakers, room and DSP have everything they need to create an incredibly vivid and defined soundstage.


 

 But can you show me measurements of the soundstage?  Can you show me a chart of how vivid it was, based on when during the day you listened? Can you give me a mathematical breakdown of how the different mastered versions of the same song sound different? 
  
 I don't think you can because we haven't developed measurements for such things yet. But the soundstage exists in our ears, and the format and resolution of the source can alter that soundstage.
  
 Since you can't measure it, you insult those that hear it and can manipulate it, and hear it's improvement at higher resolutions. 
  
 This is with all things being equal - knowing your systems and not moving speakers around.
  
 It's what it does to our simulated stereo in a music mix. That's affects the timbre of the instruments and the reverbs and colors the sound. Many describe 16/44 as being "in a box", or "thin", and many hear lossy compression as a smaller, noisier box.


----------



## bigshot

ffbookman said:


> But can you show me measurements of the soundstage?


 
  
 I am a producer. I've supervised sound recording, sound editing and mixing. Soundstage is something that is created in recording and mixing, not in the playback. It's good or bad depending on the skills and hard work of the artists and engineers creating the mix. Whenever I finalized a mix and bounced it down to 16/44.1 for release, I didn't leave the studio until I was convinced that the sound on the release version was *identical* to what we were hearing on the mixing stage. High bit or sampling rates help a lot in the mixing stage. They don't make a lick of difference for playing music in your home.
  
 If a well recorded and mixed CD doesn't sound good in your system, you should look at correcting the problems in your system, not upping the file size.


----------



## castleofargh

greenears said:


> lamode said:
> 
> 
> > *Well analogue tape, even at 15ips and on the best available machine gives you around 72dB DR, equivalent to 12 bits only. So CD is a step up from analogue master tape.*
> ...


 

 why wouldn't that be true? on tape obviously they measured the noise floor and came up with the bit equivalent(unless we're talking digital signal on tapes). it's not saying anything about where the music is relatively to 0db. or how dynamic the track is, making us listen louder, or how loud we listen to the track in general. all those stuff will pull up the noise floor for our listening, but it doesn't have to make the bit/db dynamic wrong. the tape will still be able to output the value, how it's used isn't the tape's fault.


----------



## stv014

greenears said:


> This type of analysis that 72 dB = 12 bit (6 * N bits) is so often repeated on forums.  But I don't think it is true. If you do the exercise and ABX a 12 bit vs 16 bit version from tape you will hear the difference.


 
  
 If you convert a sample that has already only 72 dB dynamic range (note however that tape recordings may be allowed to go a few dB above the reference 0 dB level) to 12 bits, adding the -72 dBA noise floor once more, then you are basically reducing the ENOB from 12 to 11.5. If the noise floor of the tape is not pure white noise, then it may also be lower than that of 12-bit dithered PCM at some frequencies. However, recording at 16 bits should likely be enough of a safety margin.


----------



## cspirou

Is digital volume control relevant to this argument? I came across this slide show by ESS.
  
 http://www.esstech.com/PDF/digital-vs-analog-volume-control.pdf
  
 In other words if the DAC output is going to a regular amp with an analog volume control, than 24 bits is probably irrelevant. However digital volume control leads to a decrease in S/N ratio for 16 bit audio. They work around this by using a 32 bit dac to process the signal and the volume must be controlled through the DAC.
  
 Now I admit that I am not an expert at this topic but my naive interpretation is that you will have an audible difference in noise between a 16 bit file and a 24 bit file if you control the volume through iTunes or the digital volume controls on a laptop or smart phone. Would 24 bit audio actually make a difference in these cases?


----------



## stv014

> Now I admit that I am not an expert at this topic but my naive interpretation is that you will have an audible difference in noise between a 16 bit file and a 24 bit file if you control the volume through iTunes or the digital volume controls on a laptop or smart phone. Would 24 bit audio actually make a difference in these cases?


 
  
 16-bit resolution before the digital volume control is not a problem. A well implemented software volume control may look like this:
  
 16-bit input file -> conversion to floating point -> volume control, any other DSP -> conversion to 24-bit integers -> DAC
  
 It is when the processed audio is converted back to 16-bit (even though the DAC would probably support 24) that there is potentially an avoidable loss of sound quality.


----------



## castleofargh

cspirou said:


> Is digital volume control relevant to this argument? I came across this slide show by ESS.
> 
> http://www.esstech.com/PDF/digital-vs-analog-volume-control.pdf
> 
> ...


 
 I agree with everything that is in the pdf. but in practice I tend to find it less dramatic than what they make it to be.
 first, on many amps I've used, the noise generated by the amp itself was the audible noise for me, not the quantization noise that was below and not audible. so I would have to really lower the digital volume a good deal before it would overthrow the amp's internal noise in my ears.
 also the readings are for the DAC, so to us that would be the same as if we assume the amp's noise to be below those values, and also assume that changing the amps pot will not change the noise level of the amp itself.
 so again while I agree with the pdf as far as the DAC is concerned and will not advise people to set their computer to 1% volume^_^, I also believe that it might not be so obvious of a problem as mentioned.
 just set your output to stream 24bit to the DAC and you already have a nice little safety cushion even when playing a 16bit track.


----------



## RRod

greenears said:


> This type of analysis that 72 dB = 12 bit (6 * N bits) is so often repeated on forums.  But I don't think it is true. If you do the exercise and ABX a 12 bit vs 16 bit version from tape you will hear the difference.  Luckily I am now in possession of a program to make this test easy, so I can try it on some albums that were originally all tape.  Rumors is supposed to be well recorded, I can try some others from the era.


 
  
 I just tried two of my Appalachian Spring recordings: NYPO/Bernstein (ADD) and Cincinnati/Kunzel (DDD). I put them down to 12/44.1 and gave them a listen. I haven't done an ABX since I'm away from my preferred program, but the DDD recording will be easy to pass; the noise is so obvious in the quiet passages at 12 bits. The ADD, on the other hand, I can tell will be tricky, and I'll need the quick-switch ability of my usual ABX program to have any chance of passing. And that's knowing what to listen for and where, and knowing that I have to jack the volume as high as is tolerable in the loud sections to have a chance at hearing the hiss in the soft sections. A newbie probably wouldn't have a shot.


----------



## bigshot

Straining to hear a difference and jacking the volume level around looking for noise is a worst case scenario.  It probably wouldn't make much of a difference for the purposes of listening to music at a normal volume level in a living room.


----------



## RRod

bigshot said:


> Straining to hear a difference and jacking the volume level around looking for noise is a worst case scenario.  It probably wouldn't make much of a difference for the purposes of listening to music at a normal volume level in a living room.


 
  
 Note I don't mean adjusting the volume during the track; that's a no-no. Rather, I mean setting the volume to where the loudest part is perhaps a bit louder than ideal with fresh ears, but that might be ok after actually listening to the track before getting to the peak. Still, audible is audible as long as the test is set up correctly. And yeah, someone doing this on speakers in the living room wouldn't be missing those 4 bits much for the NY recording.


----------



## bigshot

16/44.1 crosses the line into overkill by a bit. It's good that it does, but with normal listening to music, high end that rolls off above 17kHz probably wouldn't be noticed much. Nor would knocking it down to 14 bit. That's the beauty of redbook. It even covers worst case scenarios.


----------



## RRod

bigshot said:


> 16/44.1 crosses the line into overkill by a bit. It's good that it does, but with normal listening to music, high end that rolls off above 17kHz probably wouldn't be noticed much. Nor would knocking it down to 14 bit. That's the beauty of redbook. It even covers worst case scenarios.


 
  
 There's a goodly number of tracks I've tried at 14/38 with the expected results. This is one thing I always find eye-rolling when reading hi-res apologetics: they do everything BUT down-sample the darn song themselves and do a blind test. It's really not hard: make the two versions, get a switcher of your liking, and have someone randomly turn on and off the hi-freq stuff.


----------



## bigshot

That's a really good point. Everyone struggles and strains to hear the differences between redbook and high rate tracks. They might have a better perspective if they went the other direction first and tried to determine the line where there are differences they can actually hear.


----------



## Greenears

cspirou said:


> Is digital volume control relevant to this argument? I came across this slide show by ESS.
> 
> http://www.esstech.com/PDF/digital-vs-analog-volume-control.pdf
> 
> ...


 
  
 I was curious about this a few months back ... my review of the leading 3 24-bit DACs showed they all had some kind of internal digitally-controlled volume circuit.  It's semantics but I wouldn't call them pure digital, I would call them digitally-controlled analog.  Pure digital to me is just number crunching - where you divide every PCM value by some amount which will clearly reduce the SNR.   Digitally controlled analog won't affect the SNR.  I think that's the point the presentation is trying to make - if you do it right it's not an issue.  Their specs showed that and I moved on.
  
 Although .... if I'm wrong and my computer didn't implement it that way it could explain why I had trouble passing the ABX tests.  Sooo many variables....


----------



## Greenears

rrod said:


> I just tried two of my Appalachian Spring recordings: NYPO/Bernstein (ADD) and Cincinnati/Kunzel (DDD). I put them down to 12/44.1 and gave them a listen. I haven't done an ABX since I'm away from my preferred program, but the DDD recording will be easy to pass; the noise is so obvious in the quiet passages at 12 bits. The ADD, on the other hand, I can tell will be tricky, and I'll need the quick-switch ability of my usual ABX program to have any chance of passing. And that's knowing what to listen for and where, and knowing that I have to jack the volume as high as is tolerable in the loud sections to have a chance at hearing the hiss in the soft sections. A newbie probably wouldn't have a shot.


 
  
 Right, so I think you just proved my point.  I need to try it too.  If you have an obvious pass at 12 bits, that is 72 dB DR by the famous formula 6 * N bits.  So we can agree that there almost nothing in audio everyone in any forum agrees on, except that there is no recorded music with more than 60 dB DR and 20 or less is common for popular music.  So how can you hear it? At most 10 bits should be sufficient. Hmmm.   It's not so simple, that formula is an approximation and DR is more complicated than a single measurement.


----------



## RRod

greenears said:


> Right, so I think you just proved my point.  I need to try it too.  If you have an obvious pass at 12 bits, that is 72 dB DR by the famous formula 6 * N bits.  So we can agree that there almost nothing in audio everyone in any forum agrees on, except that there is no recorded music with more than 60 dB DR and 20 or less is common for popular music.  So how can you hear it? At most 10 bits should be sufficient. Hmmm.   It's not so simple, that formula is an approximation and DR is more complicated than a single measurement.


 
  
 Yeah I said that like several pages ago… The formula is convenient for putting bit-rate comparisons into perspective, but there is of course more to dynamic range. And I think many people understand that 72dB peak-to-peak will be less is you look at rms-to-rms. If we take common musical crest factors from 8-10, then we need 18-20dB* above the RMS for the peaks, so if our RMS range is 60-65dB (which is what people tend to mean with the 60dB dynamic range #, at least that's what I look at), then that's 80-95dB peak-to-peak. So 16bits works with certainty, but 12 would not (for very dynamic music), 13 might barely, and 14 would work in most cases. It all adds up, really.
  
 *too lazy to calculate; shamelessly lifted from wikipedia :3


----------



## castleofargh

greenears said:


> rrod said:
> 
> 
> > I just tried two of my Appalachian Spring recordings: NYPO/Bernstein (ADD) and Cincinnati/Kunzel (DDD). I put them down to 12/44.1 and gave them a listen. I haven't done an ABX since I'm away from my preferred program, but the DDD recording will be easy to pass; the noise is so obvious in the quiet passages at 12 bits. The ADD, on the other hand, I can tell will be tricky, and I'll need the quick-switch ability of my usual ABX program to have any chance of passing. And that's knowing what to listen for and where, and knowing that I have to jack the volume as high as is tolerable in the loud sections to have a chance at hearing the hiss in the soft sections. A newbie probably wouldn't have a shot.
> ...


 

 because you start to be able to perceive the noise maybe?????????  I know I've only said that like 20times by now. but you still try to find complicated stuff while rejecting the most obvious(and the actual answer).
  
 anyway, on a circuit ladder, to add one bit you add a new path with half the voltage that can be turned ON or OFF. if the step above was 1V, that new "bit" will output 0.5V and you will be able to do 0v/0.5v/1v/1.5v that we will code as 00/01/10/11, telling to the circuit what path to close. it's pretty straightforward. that 2times ratio between the path/bits gives 6db(well 6.02060709562....damn logs). it doesn't guaranty a 5db value by itself or a 3.025db value, but it does guaranty one first value and another one 6.02db below. whatever the support, the equivalence isn't going to change and each new bit will effectively offer 6more db as maximum amplitude.


----------



## Greenears

stv014 said:


> This updated version of the quantize program includes a Win32 executable, and supports 24-bit PCM samples (files created by sox and the dsputils programs should work). When processing 24-bit input, the output file is always in the same format, so it does waste some space. Another change is that adding 4 to the dither type switches from rounding towards the nearest integer to floor rounding.


 
  
 OK - I got around to try it.  This is embarrassing!!!
  
 I simply took HD Tracks 2014 Sampler, track 02 Vivaldi's spring 24 bit (free from HD Tracks anyone can try this).  Renamed it spring24.flac
  
 My incantation:
  
 sox spring24.flac -t wav spring24.wav
 quantize spring24.wav spring12.wav 12 0
  
 Quantize.exe works like a charm, easy as pie.
  
 foo_abx 1.3.4 report
 foobar2000 v1.3.6
 2015/02/06 22:21:02
 File A: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\spring12.wav
 File B: C:\Users\Public\Music\HDtracks\Various Artists HDtracks Sampler\HDtracks 2014 Sampler\spring24.wav
 22:21:02 : Test started.
 22:24:04 : 01/01  50.0%
 22:24:30 : 01/02  75.0%
 22:25:53 : 01/03  87.5%
 22:27:34 : 01/04  93.8%
 22:30:03 : 01/05  96.9%
 22:31:12 : 01/06  98.4%
 22:33:13 : 02/07  93.8%
 22:35:52 : 03/08  85.5%
 22:36:40 : 03/09  91.0%
 22:37:04 : 03/10  94.5%
 22:37:16 : Test finished.
  ----------
 Total: 3/10 (94.5%)
  
 Shocker, can't tell 12 from 24, and this is a track HD tracks holds up as a great sample of 24 bit quality.  I tried a run earlier and got 3/6 and stopped.
  
 I did try really hard, I was sure I could pass this.  Now in full disclosure on this track there is an artifact.  There is 2 second quiet before the first note. Something in that probably destabilizes my DAC at 12 bit and you can hear a little noise there.  That's my theory, since zero should not be affected by chopping 12 bits.  I only noticed it after a while at 12 bit, but it is very obvious at 10 bit.  However, if you restrict yourself to 2 seconds and onwards playing music at medium to loud sustainable listening levels I failed badly.  Not sure if the noise is not there or masked when the music is playing. 
  
 Try it yourselves.  If anyone can pass I want to know.  These days I consider 9/10 a clear pass. I'll discuss 8/10, 17/20 or maybe 16/20 if you're really convincing.  But if you have to listen to each trial for more than 60 seconds frankly you're not hearing a clear difference.  It needs to be music, not a gap, and at a level you can listen to the whole piece for an hour.


----------



## James-uk

http://archimago.blogspot.ca/2015/02/measurements-bob-dylans-shadows-in.html?m=1

this link is for those among us that belive 24 bit is valid. dont want you wasting money on this album in 24 bit.


----------



## Greenears

james-uk said:


> http://archimago.blogspot.ca/2015/02/measurements-bob-dylans-shadows-in.html?m=1
> 
> this link is for those among us that belive 24 bit is valid. dont want you wasting money on this album in 24 bit.


 

 Quantize it to 12 bit like I did above and pass an ABX.  It's really easy to do it didn't take me along at all.  If you can't pass that it doesn't matter what they did to it and whether it's really 15 bit or 16 bit as the article says.   Try it with both the 24 and 16 bit versions.  Dare you. Double dare.


----------



## bigshot

Ah! I remember the fabled past when an innocent lad by the name of green ears came into Sound Science proclaiming his intention to ABX 24 from 16. We all laughed and he got angry! Now the tables have turned! He's the one challenging expectations. Ah memories!


----------



## James-uk

greenears said:


> Quantize it to 12 bit like I did above and pass an ABX.  It's really easy to do it didn't take me along at all.  If you can't pass that it doesn't matter what they did to it and whether it's really 15 bit or 16 bit as the article says.   Try it with both the 24 and 16 bit versions.  Dare you. Double dare.




You are preaching to the converted .I don't need to, I know 24 bit makes no difference therefore I know I will fail that ABX test just like everyone else. I posted that link so the ones that have been proclaiming they prefer 24 bit don't waste money on an album that isn't 24 bit. I mean if you are buying some BS snake oil it may as well be the full fat version!


----------



## lamode

greenears said:


> Shocker, can't tell 12 from 24, and this is a track HD tracks holds up as a great sample of 24 bit quality.  I tried a run earlier and got 3/6 and stopped.


 
  
 I think it would be helpful to list the playback equipment used in such tests too. The worse a system gets, the more difficult it becomes to judge quality, as the audio chain is only as good as the weakest link. I can't hear a difference between MP3 and CD if I use my $20 headphones


----------



## OddE

While debating whether 16 or 24 bits are what it takes to reach earvana, it is all too easy to forget the plight of others.
  
 Look at this cake I found in the local supermarket today - apparently, the cake industry is stuck in the stone ages...
  

  
 Still have high hopes for it, though.


----------



## bigshot

lamode said:


> I think it would be helpful to list the playback equipment used in such tests too. The worse a system gets, the more difficult it becomes to judge quality, as the audio chain is only as good as the weakest link. I can't hear a difference between MP3 and CD if I use my $20 headphones


 

 I can't hear the difference between AAC 256 VBR and CD using my best speaker system or headphones. I think most of us have good systems, not $20 headphones. There is an old saw in audiophilia... "If you can't hear it, you either have lousy equipment or hearing loss." Usually, this is hogwash because even with the best equipment and ears, lots of things that can be measured are still completely inaudible. Human hearing, even the best human hearing, only goes so far. Too many people spend all their time examining the specs for their electronics and don't spend a moment to find out the specs for human hearing. Without that, performance of home audio has no context at all.


----------



## lamode

bigshot said:


> I can't hear the difference between AAC 256 VBR and CD using my best speaker system or headphones. I think most of us have good systems, not $20 headphones. There is an old saw in audiophilia... "If you can't hear it, you either have lousy equipment or hearing loss." Usually, this is hogwash because even with the best equipment and ears, lots of things that can be measured are still completely inaudible. Human hearing, even the best human hearing, only goes so far. Too many people spend all their time examining the specs for their electronics and don't spend a moment to find out the specs for human hearing. Without that, performance of home audio has no context at all.


 
  
 All of that might be true, but I still don't like to make assumptions. I'll bet you a day's wages that at least one person on this board has tried the MP3 v CD test on Apple earbuds, hence my request for clarification


----------



## castleofargh

lamode said:


> bigshot said:
> 
> 
> > I can't hear the difference between AAC 256 VBR and CD using my best speaker system or headphones. I think most of us have good systems, not $20 headphones. There is an old saw in audiophilia... "If you can't hear it, you either have lousy equipment or hearing loss." Usually, this is hogwash because even with the best equipment and ears, lots of things that can be measured are still completely inaudible. Human hearing, even the best human hearing, only goes so far. Too many people spend all their time examining the specs for their electronics and don't spend a moment to find out the specs for human hearing. Without that, performance of home audio has no context at all.
> ...


 

 apple earbuds are great now(for the price), and beats by dre sound nice. the world has changed!!!!!!  ^_^.
 I remember trying to abx 2 files with a pair of PL30. "ok no trebles at all on that track, and... no treble at all on the other one". not fun to look for a 16khz cut when you're not sure you can hear 10khz.


----------



## Greenears

bigshot said:


> Ah! I remember the fabled past when an innocent lad by the name of green ears came into Sound Science proclaiming his intention to ABX 24 from 16. We all laughed and he got angry! Now the tables have turned! He's the one challenging expectations. Ah memories!




Ah. Those were the days...


----------



## Greenears

odde said:


> While debating whether 16 or 24 bits are what it takes to reach earvana, it is all too easy to forget the plight of others.
> 
> Look at this cake I found in the local supermarket today - apparently, the cake industry is stuck in the stone ages...
> 
> ...




Did you ABX the cake? Was it a bit crunchy?


----------



## Greenears

james-uk said:


> You are preaching to the converted .I don't need to, I know 24 bit makes no difference therefore I know I will fail that ABX test just like everyone else. I posted that link so the ones that have been proclaiming they prefer 24 bit don't waste money on an album that isn't 24 bit. I mean if you are buying some BS snake oil it may as well be the full fat version!


 

 You said "this link is for those among us that believe 24 bit is valid" above which is why I dared you.  Sounds like it was a typo.  Still, it's fun to play with the quantize.c program it's really cool.


----------



## sonitus mirus

Greenears said:
			
		

> Did you ABX the cake? Was it a bit crunchy?


 
 Reminds me of this...


----------



## Greenears

lamode said:


> I think it would be helpful to list the playback equipment used in such tests too. The worse a system gets, the more difficult it becomes to judge quality, as the audio chain is only as good as the weakest link. I can't hear a difference between MP3 and CD if I use my $20 headphones


 
  
 It would be nice in the ABX utility if you could fill out some equipment options and put them in the log.
  
 Acer Ultrabook Haswell i5 Aspire S7-392-6411, Windows 8, Sox, Foobar, Realtek HD Audio 24 bit DAC, AKG K301.   This is the Realtek mentioned in the Tom's Hardware article were he did blind testing of many DACs up to $2000 and couldn't separate them from HD Audio.  Which was disconcerting since he owned one of the expensive ones.  I also looked at the specs to the extent I could.  It should be sufficient to tell 12 bit.
  
 I was concerned earlier that I didn't have the hardware to tell 24 from 16 and I've posted on that.  But this is an eye opener - it's really hard to suggest I don't have the hardware for 12 bit. It's pretty good hardware much better than the average consumer.  To me it sounds really good, the best I ever had.  I could use a headphone upgrade and I'm getting some new speakers with an amp I already have, but after this testing I don't feel much need to do anything else.  It's hard to know for sure but now I have a portable setup that I can try.  If someone claims a new headphone will resolve more than 12 bit I have a really quick way of testing that.  Once you have a known good source all you need are transducers that have the equalization and distortion that is pleasing to you. 
  
 Quantize.exe controls for a lot of things.  Quantize-man --- you're a genius! 
  
 One other HW note: All the modern DACs are multi-bit high frequency SD types.  What this means is that your PCM24/16/12 is up-sampled, usually to 64x the sampling frequency.  This involves padding zeros and then running it through a big FIR filter.  This interpolation filter will fill in the missing values, and it's doing it at the equivalent of 24 bit.  It's possible this is restoring some of the bits that got chopped by quantize.exe, and even where it is slightly off it's very hard to hear.  In short, if I had a crappier 16 bit non-SD DAC I might be able to ABX 16 to 12 bit .... maybe.  Any takers out there to try that?


----------



## bigshot

greenears said:


> I was concerned earlier that I didn't have the hardware to tell 24 from 16 and I've posted on that.  But this is an eye opener - it's really hard to suggest I don't have the hardware for 12 bit.


 
  
 The equipment is able to reproduce 24 bit fine. It isn't the equipment. It's your ears. Perfectly normal human ears.


----------



## jugate

I listen and the point is that music recorded in 24bits sounds better that 16. And so better if its recorded in 48 khz (dvd quality)
  
 If u upsampling music recorded in 16 bits to 24 they dont sounds better (maybe worse)


----------



## jugate

bigshot said:


> The equipment is able to reproduce 24 bit fine. It isn't the equipment. It's your ears. Perfectly normal human ears.


 
  
  
 i was asistant in the aes surround seminar in buenos aires, argentina.
  
 aes suport the hd audio
  
 http://www.realhd-audio.com/?p=3405
  
 even they tested the 24/96 vs 16/44.1 saying that was no diference. But after admits that the el original master audio was recording on 16/44.1 and upscaling for the test (great error).


----------



## Greenears

jugate said:


> i was asistant in the aes surround seminar in buenos aires, argentina.
> 
> aes suport the hd audio
> 
> ...


 
  
 Your link doesn't say what you claim.  It just says high-res should be created from sources that are more than 16 bit.  Not that you can hear the difference.
  
 Try and ABX 12 bit vs 16 bit - it avoids the problem above.


----------



## OddE

greenears said:


> Did you ABX the cake? Was it a bit crunchy?


 
  
 -We did obtain some interesting results; after my wife had truncated two bits, our five-year old claimed that there hardly was any cake left; however, no tastible difference was discerned when he got around to taking the bit count from six to five. Once I joined the fray and we suddenly had only three bits left, my wife insisted on a second helping and declared that a two bit cake tasted just the same as the full eight bits, and that this HD cake thingamajig is just a fad.
  
 Our son then proceeded to reduce the dynamic range to 0dB, only to find himself emitting whimpering noises at some 75dB(A) upon finding that three bits was way more than his stomach could safely handle.
  
 All told, a successful test. The whimpering has since subsided.


----------



## Greenears

odde said:


> -We did obtain some interesting results; after my wife had truncated two bits, our five-year old claimed that there hardly was any cake left; however, no tastible difference was discerned when he got around to taking the bit count from six to five. Once I joined the fray and we suddenly had only three bits left, my wife insisted on a second helping and declared that a two bit cake tasted just the same as the full eight bits, and that this HD cake thingamajig is just a fad.
> 
> Our son then proceeded to reduce the dynamic range to 0dB, only to find himself emitting whimpering noises at some 75dB(A) upon finding that three bits was way more than his stomach could safely handle.
> 
> All told, a successful test. The whimpering has since subsided.


 
  
 I'm glad you proved you can have your cake and eat 24-bits without any discernable difference.


----------



## jugate

greenears said:


> Your link doesn't say what you claim.  It just says high-res should be created from sources that are more than 16 bit.  Not that you can hear the difference.
> 
> Try and ABX 12 bit vs 16 bit - it avoids the problem above.


 
  
  
 more that 16 bits = 24
  
 the point is that if u record in 24 bits native and playback in a good system (JBL Es 30 or above. KRK 6 G3 or above) will sound more fidelic and life-like


----------



## jcx

yes its a long thread - but you are making thoroughly debunked assertions - many posts citing real references, analysis - only a few tracks in the Meyer/Moran study lacked high frequency content - most did have high rez originals
  


jugate said:


> i was asistant in the aes surround seminar in buenos aires, argentina.
> 
> aes suport the hd audio
> 
> ...


----------



## Lespectraal

More bits. Better psychological response.

The increase in resolution is not at all about the sound quality itself, but, it seems to me, more of a desire to acquire what the market suggests is best and to be well updated with the latest happenings so as to be guaranteed that the best possible sound quality is maintained.


----------



## threephi

Speaking of cake:

 http://channel.nationalgeographic.com/channel/brain-games/videos/cakes-of-deception/


----------



## bigshot

The absolute best sounding disk in my entire collection is the Japanese multichannel SACD of Donald Fagen's The Nightfly. It leaps out of my speakers with complete naturalness and presence. No distortion. Perfect frequency balances from the lowest low to the highest high.
  
 The Nightfly was one of the first records to be recorded digitally... at 16/44.1.


----------



## castleofargh

threephi said:


> Speaking of cake:
> 
> http://channel.nationalgeographic.com/channel/brain-games/videos/cakes-of-deception/


 

  
 you could post that video on every single FOTM discussion and still be on topic.


----------



## Greenears

jugate said:


> more that 16 bits = 24
> 
> the point is that if u record in 24 bits native and playback in a good system (JBL Es 30 or above. KRK 6 G3 or above) will sound more fidelic and life-like



If you would like to back up your assertions please quote the text where AES says that anywhere. They do not say these things.


----------



## jugate

greenears said:


> If you would like to back up your assertions please quote the text where AES says that anywhere. They do not say these things.


 
  
 I saying that, no aes.
  
 Aes failed in the comparation of sacd vs dvd audio and cd quality because they use a original 44.1/16 master piece.
  
 so AES begins to understand that audio hd is inminent like a stardard, at last in 48/24 format.


----------



## bigshot

Spelling counts.


----------



## lamode

jugate said:


> the point is that if u record in 24 bits native and playback in a good system (JBL Es 30 or above. KRK 6 G3 or above) will sound more fidelic and life-like


 
  
 Every properly conducted blind test has shown the opposite to be true. There is no difference. If you understand the technology, you will understand why 16 bit quantization error is inaudible (even before dithering).


----------



## jugate

lamode said:


> Every properly conducted blind test has shown the opposite to be true. There is no difference. If you understand the technology, you will understand why 16 bit quantization error is inaudible (even before dithering).


 
  
 that blind tests used native recorded 16 bits audio with a upsampling of 24
  
  
  
 dts hd ma and dolby trueHD are smoke too?


----------



## Greenears

Well Head-Fiers I've thoroughly enjoyed 24 v 16 argument, 163 pages in.  Next time we should take on something easier and less controversial, like Darwin or Global Warming.
  
 I felt I owed it to everyone who gave the many serious responses to my 24v16 ABX testing to write up my conclusions, FWIW.  And for the help and wonderful program, and even some of the funnier barbs.  I wrote in a file, and I'm not sure if there is a post length limit, so I'll probably chop it into 3 or 4 pieces.  Sorry for the verboseness, but again, we ARE 163 pages in so what do you expect?
  
 The most fascinating part of this is how hard it is to get a simple answer.  Anyhow, here it comes....


----------



## Greenears

*Final Post to Forum on 24b vs 16b ABX Testing*
 I thought I should write a summary of my thoughts now that I’m at a hiatus in my 24 bit vs 16 bit ABX testing.  I am keeping my setup, and if I get some different hardware or new tracks of known provenance I might try another hack at it, but for now I think I’ve exhausted all the known tracks and current hardware.
  
 The reason for writing this is to address a later reader that searches the question “Is 24 bit better than 16 bit audio?”.  Our poor imaginary reader sees the many many many pages of back and forth but no conclusion.  So I offer my own thoughts after incorporating my results, the research papers and specs I’ve read, and all the opinions and back and forth from these and other pages.
  
 So first, I will summarize a few of the barbs I’ve taken and my reaction to them, just for grins:
  
 It’s not as simple as dynamic range.  A jackhammer at 1m is 100 dB SPL, and a mosquito in the corner of a sound-proof room is 10 dB SPL.  Those two things are true, but the resulting 90 dB difference converted to bits by dividing by ~6 yielding ~15 bits are not the exact same bits you are listening to for sound quality.  These are not the droids you are looking for, really they aren’t.  I’m not going to debate this any further.
  
 This is not a Kobayashi Maru test, and I’m not Captain Kirk.  Really it is not and I’m not going to debate that further either.  If you don’t know what a KM test is (a) Google it, or (b) consider yourself lucky and don’t expose yourself to that giant intellectual black hole.  Can you tell how I really feel? J
  
 No I didn’t come into this with a strong expectation bias either way that the tests should pass or fail.  This was thrown at me a _lot_.  I would say my bias was about 60/40 and frankly I can’t remember which way.  I thought there was a chance I could pass but I never thought it would be easy.  Just because you think something is close doesn’t mean you don’t think there is a difference, or that the difference is not worth measuring.
  
 Next, ah … Nyquist. Sigh.  I’ve passed actual tests on Nyquist so I thought I knew something but I did’t know this: Most people that post about Nyquist and Shannon don’t know much about it.  On both sides.  The key point that never misses an opportunity to be missed is that Nyquist is analog!  Yes, analog!   It is sampled in discreet time intervals, but the levels of the samples are perfectly continuous analog samples with infinite resolution.  Quantization/digitization does not enter the picture at all that was a later addition.  Nyquist is mathematically proven and perfect and wonderful and has almost nothing to do with what I’m testing.  Really I’m not joking just open a textbook or even Wikipedia has it right. End sigh.
  
 Then finally, the big one:  What question was I answering with my test.  Many strawman questions were invented for me, and then they were efficiently shredded by one side or the other.  Some of the answers were so convincing I forgot the question.  
 So let me answer the last one directly:  I intentionally came in with a very simplified question that I thought I had a chance to answer.  *Is there ANY difference between 24b and 16b that ANYONE can provably hear.*  If I can find just one animal, vegetable or mineral to pass the ABX test convincingly, I have answered my question.  Of course, the word “convincingly” leaves me some room for maneuver, but really truly that was the question.  As simple as that.  I know well that if we did find a difference, the next issue would be that if it was so small does it matter or if only 1 person out of 100 could hear it does it matter to the other 99 trees in the forest.  My answer is we can have that debate later, let’s just get past the first hurdle.
  
 So now, my summary of thoughts.  
  
 (page 2 coming next)


----------



## Greenears

So now, my summary of thoughts.  
  
 First, let me say I learned a LOT.  More than I thought I would.  ABX testing is much more instructive if you actually try it than just read about it.  So I highly recommend anyone spending significant time and/or money on this hobby to try it honestly just once, especially on any long held belief or equipment or recording you are very familiar with and sure about the differences.
  
 Second, I have concluded that ABX testing while obviously valid for a PASS, is not so useful with a FAIL (no difference detected).  In short you can’t really prove a negative.  It doesn’t answer the question if there is some other combination of track, hardware and ears out there that might have passed the test.  As much as you try to design the test in a way to control for a lot of those what-ifs, they never really go away even if they are reduced.  This is a real weakness of ABX that requires some serious thought by the proponents.
  
 Third, audiophiles are an opinionated lot.  Understatement of the decade? Century? When I started I felt that ABX testing was kind of irrefutable.  That is until I was buried under an avalanche of refutation!  The most interesting part was much of the avalanche landed before I published my first result.
  
 Fourth, after digging out from under the avalanche, I noticed the snow piled in two sections:  (a) 24 bit is heaven (b) 24 bit is snake oil.  No snow whatsoever on the middle ground.  Actually the two piles were on opposite ends of the Grand Canyon.  If there are any doubters, please identify your lonely self…
  
 Fifth, there are people that talk about their ABX or similar testing, and people that publish logs.  First group big, second group tiny.  Thank you heartily to the second group, keep it coming.  Every single one really helped.  As I’ve posted many times, all my logs are available just ask it takes me seconds to post it.  I have kept all my files and logs in a directory, it’s not a material amount of space on my internal flash drive.  Any serious debaters should offer the same to be taken seriously.  Attempts to reproduce are the most valuable inquiry that I can think of, and you can’t do that without proper logs and file names.
  
 So what about the answer: Is 24 bit better than 16?
  
 (Page 3 final page to follow)


----------



## lamode

jugate said:


> that blind tests used native recorded 16 bits audio with a upsampling of 24
> 
> 
> 
> dts hd ma and dolby trueHD are smoke too?


 
  
 One famous test used a live feed from a microphone in another studio, and the audience compared the live feed with a volume-matched 16/44 AD/DA loop, and no-one in the audience could tell the original from the 16/44.
  
 A live feed, direct from the microphone, is the ultimate comparison source. If that doesn't sound better than 16/44 then no amount of HD will sound better either, as every HD format is technically still inferior to the analogue original, no matter how many bits.


----------



## Greenears

So what about the answer: *Is 24 bit better than 16?*
  
 Well, you’re not going to like it, but I’m still on the fence.  My current answer is that probably the limit of audibility is somewhere in the 18-20 bit range, I would guess closer to 18 than to 20.  I can and will be swayed by future testing.  I base this on my own testing, a few posted test logs on forums, and one paper that few people cite.  The rest of the commonly cited 5 papers kind of split down the middle or were not really testing 24 vs 16 in the modern sense.  I really wish there were some solid updated papers.  Yes I read all the ones I could access, and discounted the ones I couldn’t.  Sorry such is life.
  
 But, I am much firmer in another conclusion.  Anything over 16 bit is extremely tiny, very difficult to detect, and even if you can detect it not clear you will like it better.  The differences you can easily hear or are otherwise obvious to you in a 24 bit file are still there if you chop 8 bits off the 24 bit file.  And maybe even another 4 bits, yes 12 bit!  So it is not the format, it’s the recording and mastering in that case.  For those new to this, the difference of 20 bit / 16 bit is not 25%, it is more like 0.00001%, if present at all.  This is highly exponential, not a straight line. You learn this very quickly when attempting the tests yourself and all the testers that claimed success said the same thing.  But as noted above that is not the question I asked but it is an interesting side conclusion.
  
 So now what?
  
 The reason I embarked on the Illiad, Odyssey and every other mythical journey all rolled into one, is that I am at a junction that I have to decide on the format for my bucket archive.  This is the last re-rip or re-encode I will ever do.  The idea is to have a golden copy of a subset of my collection that I consider to be “definitive” or “best” however subjective that it and archive that on a master drive backed up sufficiently. So that I never have to muck with this again in my life.
  
 Main Conclusions:

 The good news is that I was convinced there is a digital frontier after which format change is absolutely inaudible.We are either there already or extremely close to attaining it.If I was easily passing this test on a handful of tracks I would have started wondering if 32 bit format or other improvements will be there in the future.I am certain it will not (at least not legitimately).
  
 The other even better news is that current hardware is incredibly amazing, to the point of perfection.The current crop of flagship 24-bit DACs and ADCs from the leading three makers (TI/Burr Brown, Cirrus/ESS, Wolfson) yield effective end-to-end system performance from mic to speaker of well over 120 dB which loosely corresponds to 20 bit.I am convinced this is both the theoretical and practical limit of human hearing.And with their reference designs, this can be had cheaply certainly for $100-200 and I hesitate to say even $10-20 range now or very soon.So we have entered the golden age of audio.This is a big change from 10 and 15 years ago.
  
 One weakness of my testing nobody pointed out but I think is important:
  
 Bear in mind there is no such thing as a 24 or 16 bit parallel DAC/ADC.  Everything is roughly described as 2.5 bit equivalent noise shaped and over-sampled at a high rate, usually around 64x the sampling rate.  So when I played back my 16 bit file it was being upsampled and going through what is known as a digital interpolation FIR filter. This may actually smooth over differences, because that is literally what it does – smooth the signal between input 1x samples.  Frankly it doesn’t matter since if you can’t ABX the final output it doesn’t matter.  But interesting thought.
  
 More conclusions:

 Don’t throw away your Redbook.16/44 FLAC is excellent, and when you listen to it don’t feel at all compromised.Mastering and compression add way more distortion than another few bits of accuracy ever will overcome even if I eventually pass the test (which maybe I will never manage).So if you got a good clean bit-perfect 16/44 rip, FLAC it, archive it and you’re done forever.
 FLAC is the format of choice, forever, done.Make your master archive and fix your tags all in FLAC, and then maintain some other formats converted from FLAC just to appease certain recalcitrant hardware vendors.Eventually the others will melt away.It would help if all the readers would only buy FLAC to help it along, wishful thinking.
  
 Use flagship 24 bit DACS from last 2 years, and try to get as much 24 bit recordings as well, even if the file is 16 bit. The reason is that 24 bit is really about 20 bit of system accuracy and 16 bit is maybe 14.5-15 bit.I can’t prove it’s audible, but the hardware is cheap so why not try that going forward.However, see above – don’t throw away your 16 bit recordings.They are still dandy.I also think that playing a 16 bit recording and 16 bit file back on a 24 bit DAC is worth it.For the reason (a) it’s cheap so why not (b) technical reason see below.
  
 Transducers are everything.This was always true to an extent, and it is only ever more true as semiconductor technology has advanced.Just to be controversial, I’ll state that all my research the last few months tells me that for under $1000 you should be able to get a completely transparent DAC, preamp and amp combination for any transducer that won’t blow an eardrum.Yes, all three together! If you like some coloration ridiculous 32 bit floating point performance EQ can be had for $200 in any state-of-art laptop.Another bonus of the last 20 years of Moore’s law.
  
 Analog (read vinyl) is silly.Oh yes, this will start the storm.Bring it on.This is easily proved – take your best-in-class vinyl setup, and loop the analog output through a $100 sound card on a PC.Channel A is analog straight out, Channel B is recorded on 16 bit ADC and back out through a DAC.Level match and you will not be able to ABX the difference A vs B.This experiment has been done as far back as the 80s.So it’s not the audio format, there is no “digititis”, the evil Nyquist gremlins don’t live in a DAC-house.In the end the “vinyl sound” is the equalization and coloration added by the end-to-end system of tapes, vinyl mastering, phone etc.It can be mimicked with a good floating point equalizer.
  
 So what will I do next?  Well I’ll probably spend a little effort to make sure my DAC is really state of the art 24 bit which I think it is already, and upgrade a few transducers.
  
 When buying a new recording I might buy the 24 bit version if it is under 20% more cost than the 16 bit.  That’s completely arbitrary but if the price difference is under a cup of coffee I might do that until I’ve done more ABX testing …. Just in case.  But I will not pay $40 for 24b vs $20 for 16 – no way.  That kind of pricing offends me based on my results so far.  Remember I was ABX’ing examples held up by those companies as state-of-the-art 24 bit, not just any random file.  Think about that.
  
 Also, in recent testing, I tried that fantastic quantize.c (quantize.exe in Windows) software.  Thanks @stv014!! Even 12 bit is hard to ABX.  So I am not sold on ever buying more than 16/44 – part of me just wants to boycott the whole thing.  In reality from time to time I’ll probably try one or two that are held up as really good, and ABX them to see if the “good” is in the format or the imagination.
  
 Finally it’s back to the music for me, to find better masterings, recordings and versions.  I’m quite dejected by the state of dynamic range compression and I’ll be spending more effort checking or measuring DRC and evaluating what posters consider good masterings.  I wish the powers that be would release uncompressed tracks, and provide parameters so that you can compress on playback.   The audio industry would do itself justice defining a standard model equalizer and compressor, and transmit a number of settings along with raw bit files.  So each bit file comes with say 5 settings – studio monitor (no processing), party (high compression), audiophile (medium-low compression), vinyl equivalent (lots of colorization) and whatever else creativity can muster.  THIS is the promise of digital and THIS I would pay for. 
  
 Ah but we can wistfully wonder wish and wait…. And wait…. And wait…. (fade out to sound of snoring in the distance)..


----------



## sonitus mirus

greenears said:


> Fifth, there are people that talk about their ABX or similar testing, and people that publish logs.  First group big, second group tiny.  Thank you heartily to the second group, keep it coming.  Every single one really helped.  As I’ve posted many times, all my logs are available just ask it takes me seconds to post it.  I have kept all my files and logs in a directory, it’s not a material amount of space on my internal flash drive.  Any serious debaters should offer the same to be taken seriously.  Attempts to reproduce are the most valuable inquiry that I can think of, and you can’t do that without proper logs and file names.


 
  
 What is the point of posting ABX logs unless they can show that a difference can be identified?  I can't hear a difference between 24 and 16-bit audio, so how would my ABX logs be of any use?


----------



## Greenears

sonitus mirus said:


> What is the point of posting ABX logs unless they can show that a difference can be identified?  I can't hear a difference between 24 and 16-bit audio, so how would my ABX logs be of any use?


 
  
 It doesn't mean your opinion is wrong.  It's just that some people will go back and forth 5 times on posts about ABX testing, and state they did it, but never post any actual data.  It's observational.
  
 I'll take 3/6 as solid signs of failure.  5/10 is absolute.  But no fibbing - you have to really try.   
  
 You (and others similar) should maybe realize by not contributing test logs you just help perpetuate the debate.  Because the 24 bit proponents will say "you have bad ear/equipment/material" or "you just didn't try" or "you don't know what to listen for".  The log also contains track info, format, tool versions etc (I wish it would also log hardware but you can add that manually as I did).  If you post your test, the proponents can try to prove you wrong detail by detail.  As in "aha - you used the xyz master of that track, I just tried the abc master and passed" then you or I can try the abc ourselves and see if we pass.  That's all I'm really saying.
  
 Just IMHO.  Doesn't make you evil.


----------



## castleofargh

greenears said:


> Fifth, there are people that talk about their ABX or similar testing, and people that publish logs.  First group big, second group tiny.  Thank you heartily to the second group, keep it coming.  Every single one really helped.  As I’ve posted many times, all my logs are available just ask it takes me seconds to post it.  I have kept all my files and logs in a directory, it’s not a material amount of space on my internal flash drive.  Any serious debaters should offer the same to be taken seriously.  Attempts to reproduce are the most valuable inquiry that I can think of, and you can’t do that without proper logs and file names.


 
 TBH it never occurred to me that my ABX results could be anything but for myself. I do not trust other people's logs much, so I don't expect others to trust mine ^_^.


----------



## sonitus mirus

I have posted some logs in the past here and elsewhere.  There is even a Foobar ABX log of me testing a 24/96 FLAC file vs a converted 16/44 320 kbps mp3 in this forum.   If you fail the ABX, the logs are mostly useless.  I don't have to prove to anyone that my ears are bad, I don't know what to listen for, or that I didn't even try.  The ABX test will not reliably show this information.  The ABX test can only show that someone can identify a difference.


----------



## Greenears

castleofargh said:


> TBH it never occurred to me that my ABX results could be anything but for myself. I do not trust other people's logs much, so I don't expect others to trust mine ^_^.


 
  
 My thought in this is why would someone fabricate a log? In some back-and-forth about logs my general impression is people are honest.  People rant in free form text endlessly defending a strongly held belief and seem to relish in it, but they don't spend 10 minutes altering a log file.  Not sure why - something about human nature or humans-that-post-in-forums nature.  A few would post a log and then later joke about it, revealing later they heard some artifact or unrealistic high volume.  Jokes yes, altering results not as far as I could tell.
  
 The truth is in them logs?   Interesting to me that this minor point of my endlessly long windbag posting is what people are picking up on.


----------



## sonitus mirus

The problem is that an overwhelming number of people can't hear a difference in most of the ABX tests that bring about many vehement debates in these forums.  When you can't hear a difference, but are certain that you must be able to, attempts at cheating are more likely to occur.  
  
 I've done all of the testing that I needed to do to convince myself that I could not hear any differences between a CD, a CD ripped to FLAC, a CD ripped to a 320 mp3 file, or a Hi-Rez 24/96 file from the same master.  I followed many discussions and researched a plethora of scientific observations that strongly suggested that there was no audible difference to be heard.  I'm converted, as my initial opinion was heavily swayed by industry-wide marketing and popular misbelief.
  
 I'm coming from a position where further proof is no longer necessary to affirm my understanding of what is audible and what is not.  Failed ABX tests with subject matter that has already been properly observed, evaluated, and recorded is not solidifying anything for me.  It is already a truth.  I'm only interested in those that refute these claims.


----------



## FFBookman

lamode said:


> Every properly conducted blind test has shown the opposite to be true. There is no difference. If you understand the technology, you will understand why 16 bit quantization error is inaudible (even before dithering).


 

 "properly conducted blind test" still gives garbage results.
  
 ears don't work the way blind tests assume they do.
  
 unless there's a better test than AB or ABX as they are currently constructed I will continue to dismiss the results as useless.


----------



## FFBookman

sonitus mirus said:


> The problem is that an overwhelming number of people can't hear a difference in most of the ABX tests that bring about many vehement debates in these forums.  When you can't hear a difference, but are certain that you must be able to, attempts at cheating are more likely to occur.
> 
> I've done all of the testing that I needed to do to convince myself that I could not hear any differences between a CD, a CD ripped to FLAC, a CD ripped to a 320 mp3 file, or a Hi-Rez 24/96 file from the same master.  I followed many discussions and researched a plethora of scientific observations that strongly suggested that there was no audible difference to be heard.  I'm converted, as my initial opinion was heavily swayed by industry-wide marketing and popular misbelief.
> 
> I'm coming from a position where further proof is no longer necessary to affirm my understanding of what is audible and what is not.  Failed ABX tests with subject matter that has already been properly observed, evaluated, and recorded is not solidifying anything for me.  It is already a truth.  I'm only interested in those that refute these claims.


 

 that's sad man.  
  
 go into a recording studio, or record some drums that you yourself are hitting, and tell me there's still no difference between mp3 and 24bit audio.
  
 make sure you do a stereo mix too.  too much test tone mono signal crap in this discussion, and both are woefully inadequate compared to listening to music.


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## FFBookman

Unless you know to listen for the room, the silence, the delays, the decays, and the timbre of instruments, you probably will think an mp3 sounds as good as 24bit. 
  
 Also best to avoid modern popular music for these tests, as the MP3 era production is built to hide all the bandwidth limitations.
  
 But if you really focus on the soundstage, really focus on the depth and character of the space, that's the tell for higher-definition digital.
  
 MP3 comes out of the speakers, it sounds like the instruments are inside of the speakers.
  
 Redbook starts to make the speakers disappear and brings something similar to the instruments into your room, but your room sounds flat, square, empty, with no discernible center, and the instrument isn't really in focus and not quite in your room, but a projection from the speakers. At loud volumes you will experience fatigue as your ears are trying to determine the spatial degradation.
  
 24bit focuses the center and widens the soundstage, and along with it the EQ. The speakers disappear and the instrument forms in your room. Additional details like breaths and fret noise and other squawks present themselves. So do buried 3rd and 4th guitar tracks, various layers of planned distortion, and hand percussion.
  
 This is called many things, but I call it "stereo accuracy". Stereo informs every single instrument and every single moment of silence in music. Dither/downsampling/aliasing/filtering -- all affects the stereo accuracy in ways they can't (and don't try to) measure.  But I hear it, and most mixing engineers and professional musicians hear it. So do many people that listen to vinyl. Vinyl has many issues, but presenting an accurately-timed stereo signal is not one of them.
  
 Someday your math and science will catch up with our ears, but at this point it is not the case. There's plenty more going on in well recorded music than 16/44 can accurately recreate. 
  
 Standard disclaimer - some systems won't show much advantage with any file upgrade since they are such a mess to begin with. Laptops and phones come to mind. There's really no improving that $4 signal chain.


----------



## sonitus mirus

ffbookman said:


> that's sad man.
> 
> go into a recording studio, or record some drums that you yourself are hitting, and tell me there's still no difference between mp3 and 24bit audio.
> 
> make sure you do a stereo mix too.  too much test tone mono signal crap in this discussion, and both are woefully inadequate compared to listening to music.


 

I did all of that, just now, and neither I nor the 500 listening experts with me could hear a difference. Now how can I prove this?


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## FFBookman

greenears said:


> It doesn't mean your opinion is wrong.  It's just that some people will go back and forth 5 times on posts about ABX testing, and state they did it, but never post any actual data.  It's observational.
> 
> I'll take 3/6 as solid signs of failure.  5/10 is absolute.  But no fibbing - you have to really try.
> 
> ...


 
 First off kudos on working this hard. Regardless of if I agree with your conclusion, I can accept and honor the hard work you've put into the subject.
  
 I read through your conclusions and I do agree with several of them. No haterade here. But I just don't see you discussing the tones and the presentation of any music.
  
 I feel there is a huge divide between how our ears ingests music that we love and what our scientific instruments are able to show us.  
  
 I believe people when they show that they focused on the sounds, the performances, the rests, and the overall emotional content of the music, not some number or scope reading.
  
  
 If you know exactly how a guitar solo sounds, how a drum fill sounds, how a horn part sounds, how a vocal line makes you cry almost every time,  and you are able to AB test that piece in a slow, natural, multi-day test you will go towards the higher quality, whatever it might be. 
  
 50% of the time, 80% of the time, 20% of the time? I don't know. I don't have the budget nor staff to conduct such tests. But I hope someone does soon.
  
  
 I can and have "failed" an AB test trying to pick out higher resolutions. It's easy to trick and annoy the ear. Read about psycho-accoustics and tell me again how the ear likes to be tested with parlor games. The fact that you can't prove HD exists with an AB test shows that the AB test is flawed.


----------



## FFBookman

sonitus mirus said:


> I did all of that, just now, and neither I nor the 500 listening experts with me could hear a difference. Now how can I prove this?


 

 You are trying to tell me that you can literally :
  
 -- Sit at a drum set and play
 -- Record that with at least 3 mics at 24bit into a decent interface. Apply EQ, compression, pan, and reverb as needed, all at 24bit.
 -- Mix down a 2-track master at 24bit, save as WAV or FLAC.
 -- Dither that mix to 16-bit, save as WAV or FLAC.
 -- Take the 16bit file and compress to mp3 so you have 3 files -- 24bit drums, 16bit drums, and MP3 drums.
 -- Playing back on that same mix system - you hear no difference between those three files?  
  
 The reverbs, the wood tones, the snare snap, the kick drum, the decays on each drum would be different to my ears. Are you claiming that to you, and to 500 "listening experts", they couldn't hear that difference?
  
 I agree that on your laptop or iPhone they would sound very similar. But on that production system (with the decent 24bit DAC) or on a good modern playback system like a PP, there should be clear differences that even a layman could hear.
  
 I think there's so much audio masking, production tricks and logical confusion going on that such a bare bones test with just drums would show a lot.


----------



## Krutsch

greenears said:


> The other even better news is that current hardware is incredibly amazing, to the point of perfection.The current crop of flagship 24-bit DACs and ADCs from *the leading three makers (TI/Burr Brown, Cirrus/ESS, Wolfson)* yield effective end-to-end system performance...


 
  
 When did Cirrus Logic acquire ESS Technology? Did I miss something in other news?


----------



## Greenears

sonitus mirus said:


> The problem is that an overwhelming number of people can't hear a difference in most of the ABX tests that bring about many vehement debates in these forums.  When you can't hear a difference, but are certain that you must be able to, attempts at cheating are more likely to occur.


 
  
 Well there is no way to know about anything online for certain.  But if someone posts a log you can try to reproduce it, which I tried on a few occasions. A few people posted passes but admitted they weren't hearing it in music.  But then, one or two said they passed but only on certain equipment.  I can't prove it because I don't have those phones he mentioned, but it sounded plausible to me.  So there is still a small chance there is something audible, that was my takeaway. But even he said the difference is very small and hard to hear.  Which is one of my main conclusions.
  
 I guess if the passer had said "Oh it was night and day, so much more air and this and that" (like they always say) and posted a 10/10 log I would have thrown it out.  But that wasn't the tone or the result.  That's just my perspective.


----------



## Greenears

ffbookman said:


> I read through your conclusions and I do agree with several of them. No haterade here. But I just don't see you discussing the tones and the presentation of any music.
> 
> I feel there is a huge divide between how our ears ingests music that we love and what our *scientific instruments *are able to show us.
> 
> I believe people when they show that they focused on the sounds, the performances, the rests, and the overall emotional content of the music, not some number or scope reading.


 
  
 Well it became like a game. I really wanted to pass. Some people get hooked on Candy Cruncher, I did some ABX.
  
 I didn't discuss it in the conclusions, but it's in the original posts if you go back.  I did try to listen for tone, soundstage, timber, air, presence, all these things.  Remember many of these files were held up by Linn and HD Tracks as good examples of 24 bit.  Many were classical.  I did the best listening for timbre of a clarinet on a short section (but still very hard and never to 1% at best around 5%).  I listened for tone on strings.  I also tried to just let the piece run and go with whatever "felt" better to me.  Vocal tone. Bass. Those air/presence/emotion things sure felt real to me on first listen, but when you actually try to pick A from B blind I couldn't with that approach. None of the other passers did either, they all grabbed onto specific things in segments.  Can't prove it will never be possible, just that I didn't yet.
  
 The key thing to keep in mind is that ABX is not a "scientific instrument" per se.  Audio Precision is that, but not ABX.  The instrument is only your ears.  The switch could in theory be done manually with cables.  No real instrument really.  You can listen to whatever you like in the way you like as long as you like.  Days if you want.  But if you feel something different but your brain can't tell you which is A or B ... does the tree make a sound?
  
 I don't really get these objections to the concept of ABX testing, as I wrote that both surprised and perplexed me.  Oh well.


----------



## Greenears

krutsch said:


> When did Cirrus Logic acquire ESS Technology? Did I miss something in other news?


 
  
 Cirrus bought Wolfson last year.  Should be Cirrus/Wolfson my bad.


----------



## Greenears

sonitus mirus said:


> I did all of that, just now, and neither I nor the 500 listening experts with me could hear a difference. Now how can I prove this?


 
  
 Sonitus,
  
 I would encourage you to think about the inherent weakness of ABX testing: You can't prove some pair of ears and equipment out there won't pass where another subject failed. I wrote about that.  I think it's better to put some thought how to work around that.  I intentionally posted the marks where I seemed to be doing better, so that others could try to hone in on that and maybe achieve the 9/10 that is conclusive.  Also recording fresh material would be a really good idea, even just short passages.  Better control of the material and equipment can help offset this weakness IMO.


----------



## Greenears

ffbookman said:


> I think there's so much audio masking, production tricks and logical confusion going on that s*uch a bare bones test with just drums would show a lot*.


 
  
 So do the test!  If you have such skill to do all this setup and work regularly, then the ABX is just 5 minutes of nothing at the end.
  
 Better yet, post the 24 bit file and let us also try it.
  
 Remember, we are not talking about 24 bit production.  We're just saying take the 24 bit output file you mention and ABX it against a 16 bit version that is a simple chopping of the last 8 bits in sox.  Easy as pie.....


----------



## threephi

Well I too have read through this entire thread from the beginning over the last few weeks, and it has been quite entertaining, so thank you all for putting on such a fine show. 





 I am not going to tell anyone they hear something that they don't, or that they don't hear something that they do.  We each experience the world differently.  When it comes to audio gear, everyone can choose for themselves what works best for them, and if you find something that works for you and allows you to enjoy your music, great!
  
 I will also allow that it is conceivable that some people may have so finely trained their sense of hearing, and have such intimate familiarity with music production, and have access to absolute top-level gear in ideal listening conditions, that with all those stars being aligned, it _may _be possible to detect differences between otherwise identical 16- and 24-bit recordings.  I, however, am not one of those people on any of those accounts, and I surmise most people aren't either.
  
 Using my own high-end DAP and IEM's, I personally can distinguish 320kbps MP3 from 16-bit FLAC.  The differences as I hear them are restricted to an absolutely minuscule tiny bit more detail just on the edge of audibility in the quietest parts of certain tracks.  I cannot distinguish at all between 16-bit and 24-bit FLAC.
  
 Like Greenears, I also find a bit disturbing some comments which have smugly dismissed the scientific method, I assume because the writer believes its conclusions contradict their experience.  It is absolutely true that there is a lot we don't understand about the sense of hearing, but an ABX listening test isn't relying on oscilloscopes, waveforms, equations, graphs, or scientific instruments of any kind; it relies on your ears.  It seeks to answer the very simple question: can you hear a difference in a blind test?
  
 If one can't distinguish between different tracks (or cakes, or colas, or detergents) in a blind test, in other words removing all physical or social cues that can subconsciously tip which is which, there is no _material _difference, full stop.
  
 However, note my italics on "material" just there.  I do not dismiss the reality of the experience that some people have reported, that higher resolution music sounds better to them.  It has been demonstrated many, many times in a variety of contexts, not just audio, that our perceptions can be tangibly altered by external factors.  One example is in the video I linked to a few pages back: when asked to compare two identical cakes which had been marked with different price tags, people overwhelmingly preferred the cake they believed was more expensive.  And they used very detailed and specific language to describe the differences they *tasted*.  That tells me that the experience was real--people really did sincerely taste a difference.  But that difference was *not *because the cake was different.  They had an unshaken expectation that the more expensive cake would taste better, and the power of the mind is such that it *did*.  That doesn't make those people crazy, it makes them normal.
  
 And that's the reason blind tests are necessary, to take those subconscious, unintentional tricks that our brains constantly play on us out of the equation.  It is the only way to truly tell if two things are different in their actual selves, and not owing to unrelated or uncontrolled variables.  The mind is a powerful and mysterious thing which will fill in the gaps when given the chance.


----------



## Greenears

threephi said:


> I am not going to tell anyone they hear something that they don't, or that they don't hear something that they do.


 
  
 Can someone eject this guy from Head-fi?  Doesn't he know that is all we do around here?


----------



## bigshot

threephi said:


> I am not going to tell anyone they hear something that they don't


 
  
 I'm not going to tell anyone that if they order the X Ray Specs from the ad at the back of the comic book, they won't be able to see the bone in their arm or peek at pretty girls naked under their clothes!


----------



## threephi

greenears said:


> Sonitus,
> 
> I would encourage you to think about the inherent weakness of ABX testing: You can't prove some pair of ears and equipment out there won't pass where another subject failed. I wrote about that.  I think it's better to put some thought how to work around that.  I intentionally posted the marks where I seemed to be doing better, so that others could try to hone in on that and maybe achieve the 9/10 that is conclusive.  Also recording fresh material would be a really good idea, even just short passages.  Better control of the material and equipment can help offset this weakness IMO.


 

 This is where statistics and sample size come in.  Individual results aren't really relevant so much as what can be seen from testing a large population.  
  


greenears said:


> Can someone eject this guy from Head-fi?  Doesn't he know that is all we do around here?


----------



## castleofargh

ffbookman said:


> lamode said:
> 
> 
> > Every properly conducted blind test has shown the opposite to be true. There is no difference. If you understand the technology, you will understand why 16 bit quantization error is inaudible (even before dithering).
> ...


 
 "ears don't work the way blind test assume they do". sure we all know that, that's why they are blind tests. because they are an attempt at letting you use only your ears for once. but I'm sure you know all about sound, all about how our brain mixes all our senses, all about how sensitive we are to suggestion and bias. and from all that you determined that the best way to test audio wasn't to test only audio in an abx, but instead to sit in a chair and do a long session of sighted listening where you know everything in advance and just have to check that your brain agrees with itself based on what you had picked even before starting. I imagine you like that method because of how often it makes you agree with yourself, but as a testing method, it's crap.
  
 doesn't it trouble you slightly not to have anything to offer as an alternative, yet dismissing what is accepted as the most efficient subjective method? where is the logic behind that?


----------



## RRod

castleofargh said:


> "ears don't work the way blind test assume they do". sure we all know that, that's why they are blind tests. because they are an attempt at letting you use only your ears for once. but I'm sure you know all about sound, all about how our brain mixes all our senses, all about how sensitive we are to suggestion and bias. and from all that you determined that the best way to test audio wasn't to test only audio in an abx, but instead to sit in a chair and do a long session of sighted listening where you know everything in advance and just have to check that your brain agrees with itself based on what you had picked even before starting. I imagine you like that method because of how often it makes you agree with yourself, but as a testing method, it's crap.
> 
> doesn't it trouble you slightly not to have anything to offer as an alternative, yet dismissing what is accepted as the most efficient subjective method? where is the logic behind that?


 
  
 It always seems to come down to arguments about euphoria: some "feeling" of rightness one gets when bathing in hi-res for hours, a kind of aural sous vide. People never seem to have done ABX in an environment where passes ARE expected, like 16 vs 8 bit or 44.1 vs 22.05 ksamples. It's obvious the paradigm works in differentiating these conditions with statistical uncertainty. Suddenly it fails as a method when we go to 16 vs 24 or 44.1 vs 88.2? It's poppycock, of course. I would bet dollars to dogecoins that a vast majority of the hi-res crowd couldn't ABX drums at 16/44.1 vs 14/38. But we'll never know, because it's always "oh you just can't hear what I hear" or "human ears don't work with sudden shifts in frequency content," instead of "let me try to pass the simple test honestly."


----------



## FFBookman

The test I'd pay attention to is the one in the marketplace, first and foremost. Just give consumers both versions and they can decide if they hear it or care. 16/44 streaming will soon compete with MP3 streaming. If DAPs stay alive and car audio goes 24bit, consumers will be able to choose which content they want to pay for or store (if any at all).
  
 If you want a test that falls into your sociological framework, it has to be a long-term blind test done in the listeners multiple environments. On the listeners gear.
  
 The source files must be encrypted to not show their quality.
  
  
  
 How about this:
  
 The user should sign up with a list of their 10 favorite albums and their 5 listening systems (amp, speakers, location).
  
 They are encouraged to familiarize themselves with those 10 albums the week of the test initialization. To this end they are given a listening log and told to make notes and review the performances in each song, for instance what caught their ear first, what is the peak of the song, where does the song drag, are there any missed notes? They aren't scoring the playback, but they are noting place, time, and emotional cues before and after playback.
  
 The testers prepare a set of those 10 albums on an SD card with several resolutions. I'd do 24/192, 24/44, and 16/44 but anything can be tried.
  
 The SD card is in a modern DAP like the PonoPlayer, modified to not display resolutions, just a code. When playing back the album, they pick which code (version) they want and record their thoughts and feelings in the log, along with that code. They repeat this across as many of their playback systems, times of the day, and emotional states as time allows.
  
 Perhaps they manage to listen to each album 5 or 6 times in a month. This will generate enough emotional responses across enough natural listening cases that I would pay attention to those results.
  
 The final part is they get to select which version of the album they want to own, as payment for taking part in the test. They can review their log or just pick based on which they've come to prefer. Record if they review notes or not, and of course which one they select.
  
 I believe in everything I know about mixing, playing, listening, and DSP math that such an in-depth study would show that most people do indeed hear the quality differences when it is given to them in a natural way.
  
 You wanted alternatives? There's one. Is anyone doing music quality listening tests of this nature? Please point me to them if they are.


----------



## sonitus mirus

ffbookman said:


> You are trying to tell me that you can literally :
> 
> -- Sit at a drum set and play
> -- Record that with at least 3 mics at 24bit into a decent interface. Apply EQ, compression, pan, and reverb as needed, all at 24bit.
> ...


 
  
 My point was that as long as we are simply making claims with no proof, why does it matter?


----------



## FFBookman

sonitus mirus said:


> My point was that as long as we are simply making claims with no proof, why does it matter?


 

 Understood. I have proven it to myself in both the mixing/recording world and the listening world, and if I'm unable to convince you it's no loss for me, other than the time for an interesting debate.
  
 Listening to sample libraries at different resolutions on a production system, that's an easy way to hear differences.
  
 Listening to your own mixed productions mastered at different resolutions, that's an easy way to hear differences.
  
 Listening to multiple formats (vinyl, MP3, tape, FLAC, etc.) is an easy way to tell differences too. 
  
 Listening to live music, making live music, those are great ways to keep your ears in tune with the world of pleasing sound.
  
 I'm not doubting the progress and coolness that lossy compression is. I rely on it for many things. But it is lossy so it's a temporary format. 
  
 I'm also not doubting the awesome compromise and impressive technology (for 1978) that redbook represents. It's still the best digital format for most sound.
  
 I just know, like almost anyone who's done any production knows, that's there's better digital than 16/44. But I get in these fights because I'm fascinated by those who stands so strongly behind this belief that the human senses have been replicated by digital. Locking onto a single max resolution for 35+ years is odd.


----------



## jugate

lamode said:


> One famous test used a live feed from a microphone in another studio, and the audience compared the live feed with a volume-matched 16/44 AD/DA loop, and no-one in the audience could tell the original from the 16/44.
> 
> A live feed, direct from the microphone, is the ultimate comparison source. If that doesn't sound better than 16/44 then no amount of HD will sound better either, as every HD format is technically still inferior to the analogue original, no matter how many bits.


 
  
 link please. Tell me: is there any difrence audible between flac and mp3 320?
  
  
 why dts hd and dolby truehd  (formats of 24 bits/48-96 khz) exist if they dont sound better that a cd track?


----------



## RRod

jugate said:


> link please. Tell me: is there any difrence audible between flac and mp3 320?
> 
> 
> why dts hd and dolby truehd  (formats of 24 bits/48-96 khz) exist if they dont sound better that a cd track?


 
  
 Are you asking why humans keep things around despite lack of extra utility?


----------



## cjl

ffbookman said:


> The test I'd pay attention to is the one in the marketplace, first and foremost. Just give consumers both versions and they can decide if they hear it or care. 16/44 streaming will soon compete with MP3 streaming. If DAPs stay alive and car audio goes 24bit, consumers will be able to choose which content they want to pay for or store (if any at all).
> 
> If you want a test that falls into your sociological framework, it has to be a long-term blind test done in the listeners multiple environments. On the listeners gear.
> 
> ...


 
  
 First off, what possible purpose could 24 bit audio serve in a car? Cars have very high background noise levels - even 16 bit is dramatic overkill for them. Secondly, you just described a form of blind testing.


----------



## bigshot

ffbookman said:


> The test I'd pay attention to is the one in the marketplace, first and foremost.


 
  
 Look how well letting the marketplace decide is working for measles vaccinations!


----------



## bigshot

ffbookman said:


> I just know, like almost anyone who's done any production knows, that's there's better digital than 16/44.


 
  
 Better for what? Is the first question this fella who's done production is going to ask.
  
 Better for mixing and mastering? Yes. Better for listening to music on your home stereo? No.


----------



## FFBookman

bigshot said:


> Better for what? Is the first question this fella who's done production is going to ask.
> 
> Better for mixing and mastering? Yes. Better for listening to music on your home stereo? No.


 
  
 In the old days of megabytes and bad DAC's, I agree.  It didn't really matter in 1999, or 2004, or probably 2010. 24bit was for "purists" and rich guys only. And production, you are right. "Good converters" it's called, and its been important in pro audio for at least 25 years. 
  
 But it's not just the headroom. When mixing we hear that all the instruments and voices and timbre/tone of various details can suffer some loss in the conversion to 16/44, and get lost further in the conversion to lossy formats.
  
 Modern music tries to hide the bandwidth limitation, and succeeds in appearing bigger, louder, wider, more dynamic, but it's built for ADD generation. It fatigues quickly and actually has very little dynamic range compared to analog masters transferred to 24bit digital.
  
 So if average music lover can get 24bit playback these days with a good signal chain, why not?  Our speakers and many of our amps can handle it just fine.  We have been feeding them crap for the past 15+ years now.
  
 No one will take your iPhone away from you, but it should be able to play 24bit files any version now. Some android phones already can. The rest of it comes down to DAC and then analog circuit design.  
  
 If I was an android phone maker (or Apple) I'd be spec'ing out my audio upgrade for the next revision with someone like Allen & Heath or Soundcraft, to get the Pono-style analog stage. Anyone can buy the DAC and the storage space, the last piece you need is the nice analog stage to drive various speakers.
  
 There's no reason we can't have 24bit audio with a discreet analog stage on other devices. Maybe not all devices, but anything tasked with playing music. That would be a success in my book, if 24bit FLAC is the standard stereo file format for the next 20 years.


----------



## Greenears

threephi said:


> This is where statistics and sample size come in.  Individual results aren't really relevant so much as what can be seen from testing a large population.


 
  
 OK, now we are getting somewhere.  But there needs to be some statisticcs science put around it.  There needs to be a protocol, something along the lines of a 2-phase protocol: First you write a paper announcing the test, and posting your intended hardware and post a list of files that the public can download and details of how they were recorded. Then people give their criticism, and maybe some additional files are submitted by the public in the review period.  This avoids the common provenance issue.  In the 2nd phase a panel of maybe 20 recording engineers evaluates the files and hardware and announced the final selection.  That is also posted publicly and then the tests are done. You do some statistics say on the assumption 1 person in 100 is capable of passing, how many testers do you need to have a 99% confidence of catching that 1 in the sample set.
  
 All this can be done, it's not so hard and not actually that expensive compared to the investment made in the hardware and music for all his 24 bit stuff.
  
 The other advantage is that if that one pair of ears writes in and says he passed using the publicly available files, you can always invite him in and add to the pool.  Adding people never invalidates an ABX.  It's not like other sampling where you need a representative demographic necessarily. 
  
 I know this will never happen but it's nice to daydream.


----------



## bigshot

ffbookman said:


> But it's not just the headroom. When mixing we hear that all the instruments and voices and timbre/tone of various details can suffer some loss in the conversion to 16/44


 
  
 Absolutely incorrect. Within the audible range of frequency response, there is no more "resolution" with a high sampling rate than with redbook. Per Nyquist double the sampling rate PERFECTLY recreates the waveform of any given frequency or combination of frequencies. You can pile on as many samples as you want and it doesn't make a lick of difference. Two samples is all it takes to perfectly define a waveform of any given frequency. From 20 to 20, 44.1 is perfect.
  
 And lowering the noise floor by going from 16 bit to 24 certainly doesn't affect any details that you can hear at a normal listening level, unless your listening level exceeds the threshold of pain.
  
 Back to digital audio school!


----------



## FFBookman

bigshot said:


> Absolutely incorrect. Within the audible range of frequency response, there is no more "resolution" with a high sampling rate than with redbook. Per Nyquist double the sampling rate PERFECTLY recreates the waveform of any given frequency or combination of frequencies. You can pile on as many samples as you want and it doesn't make a lick of difference. Two samples is all it takes to perfectly define a waveform of any given frequency. From 20 to 20, 44.1 is perfect.
> 
> And lowering the noise floor by going from 16 bit to 24 certainly doesn't affect any details that you can hear at a normal listening level, unless your listening level exceeds the threshold of pain.
> 
> Back to digital audio school!


 

 Please mr big shot professor, explain dither to me.
  
 Looks like noise to me. Fuzz. Cover up.
  
 What are you covering up when you dither? 
  
 Is that dither stereo matched?  Can you actually stereo match random noise?
  
 Or is it so randomly in stereo that it affects the imaging of the signal it obfuscates?


----------



## threephi

greenears said:


> OK, now we are getting somewhere.  But there needs to be some statisticcs science put around it.  There needs to be a protocol, something along the lines of a 2-phase protocol: First you write a paper announcing the test, and posting your intended hardware and post a list of files that the public can download and details of how they were recorded. Then people give their criticism, and maybe some additional files are submitted by the public in the review period.  This avoids the common provenance issue.  In the 2nd phase a panel of maybe 20 recording engineers evaluates the files and hardware and announced the final selection.  That is also posted publicly and then the tests are done. You do some statistics say on the assumption 1 person in 100 is capable of passing, how many testers do you need to have a 99% confidence of catching that 1 in the sample set.
> 
> All this can be done, it's not so hard and not actually that expensive compared to the investment made in the hardware and music for all his 24 bit stuff.
> 
> ...


 
 Wait, is your goal to try and find a method to identify a golden-eared outlier who can reliably and repeatably distinguish hi-res in an ABX test?  In that case statistics don't really matter since you're looking for one exceptional result.  All you need is to test people who claim they can do it.  I think it's safe to say it would be a colossal waste of time testing people who do not make that claim, since it would be repeating and re-verifying many existing experiments.  It is noteworthy, however, that the people who claim they can hear a difference _always_ allege exceptions or flaws in the testing paradigm to explain away their negative results in a controlled test.

 I think the power of the placebo effect has not been fully understood or adequately discussed throughout this thread.  It is a truly amazing, mysterious, and powerful phenomenon.  It's not crazy or abnormal to experience it, and the _effects_ can be very, very real.

 http://www.radiolab.org/story/91539-placebo/


----------



## cjl

ffbookman said:


> Please mr big shot professor, explain dither to me.
> 
> Looks like noise to me. Fuzz. Cover up.
> 
> ...


 
  
 When you dither, you're not covering up anything. You're decorrelating the quantization noise from the signal. It doesn't affect stereo image at all, since that noise will be at something like -100dBFS or lower. Stereo image is not sensitive to noise at such a low level (hell, it's not even affected by crosstalk at a much higher level than that, say -50dB or so). The biggest thing that can kill stereo image is channel imbalance, but that's not a format issue.


----------



## bigshot

ffbookman said:


> Please mr big shot professor, explain dither to me.


 
  
 Would you like me to start making fun of you too? Because I am pretty good at that if you give me permission to follow your lead.
  
 Re: Dither: See the videos in my signature file. In one of them Ethan Winer does a demonstration of a redbook file with and without dither. Either way, you are never going to hear the noise floor at any sort of comfortable listening level. Dither is handy for pushing the noise floor down a little bit further, but it isn't absolutely necessary because in redbook, noise never gets anywhere near the level of being audible, even with no dithering at all.
  
 Another test in that video you might find interesting is where he takes a very annoying buzzing sound and progressively lowers the level under music. Where do you think the worst kind of noise possible becomes inaudible? It really helps to have an understanding of what specs like -96dB actually means in real world terms.


----------



## jcx

ffbookman said:


> Please mr big shot professor, explain dither to me...


 
 um, could you explain exactly why you are posting at Sound Science
  
 why anyone would want to help given your attitude problem - for a easily researched topic treated quite thoroughly on the web, already linked by several contributors in this thread, repeatedly


----------



## bigshot

troll


----------



## lamode

Has anyone else noticed that the people who claim to hear a difference are always self-testing?
  
 No-one has ever been tested by a neutral party, and been able to tell the difference.
  
 Says it all, really.


----------



## Greenears

threephi said:


> Wait, is your goal to try and find a method to identify a golden-eared outlier who can reliably and repeatably distinguish hi-res in an ABX test?  In that case statistics don't really matter since you're looking for one exceptional result.  All you need is to test people who claim they can do it.  I think it's safe to say it would be a colossal waste of time testing people who do not make that claim, since it would be repeating and re-verifying many existing experiments.  It is noteworthy, however, that the people who claim they can hear a difference _always_ allege exceptions or flaws in the testing paradigm to explain away their negative results in a controlled test.
> 
> I think the power of the placebo effect has not been fully understood or adequately discussed throughout this thread.  It is a truly amazing, mysterious, and powerful phenomenon.  It's not crazy or abnormal to experience it, and the _effects_ can be very, very real.
> 
> http://www.radiolab.org/story/91539-placebo/


 
  
 Mmmm.... I think the goal would be to provide a stastical metric for the likelihood of a golden-eared outlier existing somewhere in the universe even if all the tests were fail.  Obviously if we catch said GEO in the net, we can stop the testing because if they pass the ABX with enough trials (say 9/10 or 17-18/20) then we've proven the difference is audible, even if only few people can hear them.  Statisticians know how to do all this.  You can ask them if on average in a large population of fish, 99% have green tails and 1% have blue tails how many fish do you have to catch to have a 95% chance of having a blue-tail fish in your net?  The answer is not 100 it is something like 250 give or take 100.   Someone can crunch this for us - how many people you have to test to get the 99%/1% level.  I'm guessing it's in the 200-300 range - totally feasible if you test at any audio conference or CES.     Your answer comes out something like: I am 99% confident our tests would have caught a GEO in the population if the % of GEOs is >1%.  So you are saying effectively somewhere between 0-1 in 100 people could pass (maybe), but no more than that.  Increase the numbers it goes to 0-0.5 etc. 
  
 So I don't think their claims matter much, but testing experts (or maybe half experts half random) should only increase your confidence.  Most large listening studies I see test a mix of experts and novice, and group them in results.
  
 Placebo/suggestion call it what you will is very strong.  I commented on that and that's why I think ABX testing is good for everyone. I also heard things I was sure must be in the 24, and only ABX could convince me those features were in the 16 as well.  The mind a powerful thing it is, my young padawan.


----------



## RRod

greenears said:


> Mmmm.... I think the goal would be to provide a stastical metric for the likelihood of a golden-eared outlier existing somewhere in the universe even if all the tests were fail.  Obviously if we catch said GEO in the net, we can stop the testing because if they pass the ABX with enough trials (say 9/10 or 17-18/20) then we've proven the difference is audible, even if only few people can hear them.  Statisticians know how to do all this.  You can ask them if on average in a large population of fish, 99% have green tails and 1% have blue tails how many fish do you have to catch to have a 95% chance of having a blue-tail fish in your net?  The answer is not 100 it is something like 250 give or take 100.   Someone can crunch this for us - how many people you have to test to get the 99%/1% level.  I'm guessing it's in the 200-300 range - totally feasible if you test at any audio conference or CES.     Your answer comes out something like: I am 99% confident our tests would have caught a GEO in the population if the % of GEOs is >1%.  So you are saying effectively somewhere between 0-1 in 100 people could pass (maybe), but no more than that.  Increase the numbers it goes to 0-0.5 etc.
> 
> So I don't think their claims matter much, but testing experts (or maybe half experts half random) should only increase your confidence.  Most large listening studies I see test a mix of experts and novice, and group them in results.
> 
> Placebo/suggestion call it what you will is very strong.  I commented on that and that's why I think ABX testing is good for everyone. I also heard things I was sure must be in the 24, and only ABX could convince me those features were in the 16 as well.  The mind a powerful thing it is, my young padawan.


 
  
 The 1%-level for 10 trials is 10/10. For 100 trials, it's 63/100. That's testing if a single person has a >50% chance of differentiate a given track at two specs. Note that the larger sample will also have better power as well (lower false negative rate).
  
 You have to be exact about what question you are asking. Is your parameter of interest the overall probability that people (not a specific person) can identify a given track at different specs? If so, then you'd want a hierarchical setup, where each person's particular probability is a realization of some latent overall probability. There's all kinds of stuff you can do. And note you have to specify the number of trials as well as the number of people. A thousand people doing 5 trials each is way different than 5 people doing 1000 trials.
  
 Note your fish example is probability more than statistics: you start off knowing the parameters. If you catch 299 fish, then there is 95% chance you will have at least one blue tail (1 - probability of all green tails).


----------



## Greenears

rrod said:


> The 1%-level for 10 trials is 10/10. For 100 trials, it's 63/100. That's testing if a single person has a >50% chance of differentiate a given track at two specs. Note that the larger sample will also have better power as well (lower false negative rate).
> 
> You have to be exact about what question you are asking. Is your parameter of interest the overall probability that people (not a specific person) can identify a given track at different specs? If so, then you'd want a hierarchical setup, where each person's particular probability is a realization of some latent overall probability. There's all kinds of stuff you can do. And note you have to specify the number of trials as well as the number of people. A thousand people doing 5 trials each is way different than 5 people doing 1000 trials.
> 
> Note your fish example is probability more than statistics: you start off knowing the parameters. If you catch 299 fish, then there is 95% chance you will have at least one blue tail (1 - probability of all green tails).


 
  
 I think what is being proposed above is to figure out how many test subjects you need to be able to say you will catch a GEO (Golden-eared outlier) that can pass the ABX.  Nobody knows how many GEOs there are in the population, since they need to be caught to find out.  But if you don't find any GEOs you can say something about how rare they need to be in the population in order to have been missed. 
  
 Then you still have the problem of material and hardware.  It's no so simple.  It'll never happen.  Neither the proponents or detractors have much interest in conducting this test.


----------



## bigshot

I just worry about myself. If I can't hear it, even after I sweat and strain and try to hear it, it doesn't matter. I am looking for ways to make my system sound better, not ways to split the atom for humanity as a whole.


----------



## lamode

greenears said:


> Mmmm.... I think the goal would be to provide a stastical metric for the likelihood of a golden-eared outlier existing somewhere in the universe even if all the tests were fail.  Obviously if we catch said GEO in the net, we can stop the testing because if they pass the ABX with enough trials (say 9/10 or 17-18/20) then we've proven the difference is audible, even if only few people can hear them.


 
  
 If you ABX enough guessers, you will eventually have some with scores like 9/10. It's like flipping a coin 10 times and getting 9 heads (around 1% chance). It's never "proof", but if you repeat the test enough times with one person, the odds of it being chance become vanishingly small.
  
 For those people who claim the difference is "clearly audible", I would suggest the pass rate is 20/20.


----------



## RRod

greenears said:


> I think what is being proposed above is to figure out how many test subjects you need to be able to say you will catch a GEO (Golden-eared outlier) that can pass the ABX.  Nobody knows how many GEOs there are in the population, since they need to be caught to find out.  But if you don't find any GEOs you can say something about how rare they need to be in the population in order to have been missed.
> 
> Then you still have the problem of material and hardware.  It's no so simple.  It'll never happen.  Neither the proponents or detractors have much interest in conducting this test.


 
  
 Well, to be fair the hi-res detractors at least have auditory testing on their side. If you define a GEO as someone who can get through a test successfully, then you have to set up the statistical properties of the test to fail him accidentally with low probability. That's regardless of how many people you decide to put into the test. A good way to do this is to have an initial run that has a low false negative rate. This means you'll tend to pass all the GEOs, but also a fair number of people who got lucky. You then do a second round on these people, this time with a low false negative rate. If you want to actually estimate the # of GEOs in a population, then you'd need to do some probabilistic sampling on the entire population of interest. This is probably beyond the means of any individual or small organization, since a phone survey wouldn't exactly suffice.


----------



## lamode

jugate said:


> link please.


 
  
 http://www.aes.org/e-lib/browse.cfm?elib=14195


----------



## Roly1650

lamode said:


> http://www.aes.org/e-lib/browse.cfm?elib=14195



Here's another one, by the (in)famous Ivor Teifenbrun, Mr. Linn himself, who did as much as anyone to get the whole subjectivist ball rolling. According to him digital wasn't a patch on his LP12 turntable. The trouble for him was that he couldn't pick out the pure analog loop from the analog/digital loop and that with early '80's consumer level 16 bit digital technology. A lot of his other myths got busted in the same sessions.

www.bostonaudiosociety.org/bas_speaker/abx_testing2.htr

As far as I'm aware he never made any public prognostications as to why he failed, not even the limp wristed "too stressful" cop out.


----------



## analogsurviver

lamode said:


> Has anyone else noticed that the people who claim to hear a difference are always self-testing?
> 
> No-one has ever been tested by a neutral party, and been able to tell the difference.
> 
> Says it all, really.


 
 Nope. 
  
 A producer for the choir, who later in time became a friend, DID NOT take my word for granted. For several claims of my own, he tested me. I will not go into specifics, but I decided to turn around away from any possibility to see any switching or computer screen or anything that might allow me to cheat. In some cases, he did not even tell me WHAT should be the core of the test. I used either Stax Lambda Pro or AKG K 1000 headphones - both mine and familiar with.  ( Later on, my producer friend bought his own K 1000 - and we both described the sound of each set equally. K 1000 DOES differ sample to sample - but under certain circumstances can not be bested by anything else. )
  
 Although I can not say I was right 1000/999, I did well. I have noticed that this ability of mine does get worse after some 15 minutes of so of ABing or ABXing or whatever you might want to call it.
  
 This went so far that when I arrived with my equipment to our first recording together, he had already set up another setup - borrowed from the national radio. You can bet I did not like that surprise - not at all. And although the whole of my setup has NEVER been in a recording venue before,  it , in less than 5 minutes, BLEW AWAY that other setup to kingdom come. You would get the Jackass Of The Millenium award  for even _*thinking *_about ABX - THAT different/better. 
  
 A friend is also a singer - tenor - in one of the best and most highly prized amateur male choirs on the planet.
  
 No, we did not keep the AB or ABX scores; he merely wanted to test me I was not BS him. Although not 100/100, I did more than well enough to satisfy any statistics.


----------



## bigshot

I stand in awe of your auditory abilities.


----------



## analogsurviver

bigshot said:


> I stand in awe of your auditory abilities.


 
 It is not that difficult - it is quite logical after all.
  
 What is hard is to "swallow" that something that should by all logic sound better - does not. Specially if that something is "my child". That's why I always ask friends and acquaintances to frankly say what they think - there is always PLENTY of those who will, one or more generations of equipment later, separated by few years, finally drop that diplomatic position and will say what they did not like back then - if they had been frank there and then, the progress might be faster.


----------



## bigshot

I stand in awe of your logic too.


----------



## analogsurviver

bigshot said:


> I stand in awe of your logic too.


 
 Ever been to a concert ? With people criticizing certain playing techniques of X, then going to congratulate this very same X for the great performance, only to resume criticizing the "impossible" playing of the still very same X - just around the corner.
  
 I try to flush as much/many flaws out of my recording by encouraging critic, be it positive or negative - the worst thing out there is no response at all.


----------



## bigshot

I am in awe of your ability to flush flaws.


----------



## sonitus mirus

bigshot said:


> I am in awe of your ability to flush flaws.


 
  
 Hmmm...


----------



## castleofargh

analogsurviver said:


> bigshot said:
> 
> 
> > I stand in awe of your auditory abilities.
> ...




in all honesty, I would expect everybody to believe high resolution is better and will sound better. it s obviously our failure to observe an audible difference under control that pushes us to become skeptical and start to refuse paying more and wasting storage for something we do not hear.
I can't claim that highres is never audibly different. after all, the tests I've done can't prove a null, and maybe some real life daredevil(the red marvel hero) do exist who can hear more than the average joe. 
but each time I see somebody claiming he can hear the difference without having even passed some matter of blind test a statistically significant number of times, it gives fuel to my opinion that most pro "high res is audible" people shouldn't be trusted. following a protocol should be the number one priority for anybody trying to be taken seriously. that cannot change whatever our own opinion is on a subject.
 and each time someone makes that claim on a product I've used to do an ABX, then I admit to being prejudice and thinking the guy is a joke and should go read a book about bias.


----------



## jugate

placebo or deafness, that is the question... (willy say)
  
  
 but again, if anything more that 44.1/16 is bullshet then why exist dolby trueHD, dta MA and thats stuffs..


----------



## bigshot

marketing. same with "HD" audio.


----------



## kraken2109

jugate said:


> placebo or deafness, that is the question... (willy say)
> 
> 
> but again, if anything more that 44.1/16 is bullshet then why exist dolby trueHD, dta MA and thats stuffs..


 
 If expensive cables don't improve sound then why do they exist?
  
 Money.


----------



## analogsurviver

kraken2109 said:


> If expensive cables don't improve sound then why do they exist?
> 
> Money.


 
 I am probably one of the last persons to claim outrageous audible differences in cables. Except in cases the electrical design of the cable for the dedicated use REALLY does make measurable and easily audible difference.
  
 Yet claiming that differences in "normal" cables do not exist would be foolish. They are nowhere as inflated by the manufacturers and reviews would lead you to believe, but they are REAL.
  
 If sensibly designed, they also do not have to cost exorbitant amounts -  but they can not be the same price as commonly available wire.
  
 But ask any audio dealer where considerable portion of his income comes - cables...
  
 It is about finding "better than ZIP cord at reasonable price" which can provide some improvement.
  
 I find it laughable audiophiles spending $$$$ on cables while their boxes are full of low quality capacitors - which each and every one of them can provide for more improvement than a cable ever could. But selling cable is easy, replacing those caps is not.


----------



## headwhacker

analogsurviver said:


> I am probably one of the last persons to claim outrageous audible differences in cables. Except in cases the electrical design of the cable for the dedicated use REALLY does make measurable and easily audible difference.


 
  
 Measureable difference does not usually translate to audible difference. You are most like to hear a difference from bad solder joints than with the cable itself. I do appreciate a high quality constructed cable. But the claims to get audible difference ...


----------



## analogsurviver

headwhacker said:


> Measureable difference does not usually translate to audible difference. You are most like to hear a difference from bad solder joints than with the cable itself. I do appreciate a high quality constructed cable. But the claims to get audible difference ...


 
 Believe me, I would prefer if ZIP cord could do it all in audio - but it can not.
  
 All the voodoo of the cable dealers is not necessary voodoo after all - but they can fix this or that complaint by substituting the wire. It is debatable whether it is the right thing to do (my opinion is not), it is debatable if it is worth it ( at super inflated prices IMO not ) - but it is undeniable fact that they can tailor make sound to your liking using various cables..
  
 English is not my first language - and even in my first language is difficult to exactly describe the sound. All I know and care of is that I can go to a decent dealer and set up a good demonstration of my own recordings: and if it is quicker and easier to make the sound closer to what I remember heard live during the making of that recording using different wire than remodelling the room and replace/introduce sound treatment in that room, I am fine with it. Given the need to purchase those XYZ? cable$, I would at home probably try to exhaust all other possibilities first - and only if everything else failed, with hesitation reach for a credit card.
  
 Most of my cables are around 2-3 EUR per metre and I assemble them by myself. There are uses where this just will not do well enough - regardless my wallet having an opposite opinion.


----------



## bigshot

Any properly functioning cable sounds exactly the same. If a cable sounds different, it's defective.


----------



## analogsurviver

bigshot said:


> Any properly functioning cable sounds exactly the same. If a cable sounds different, it's defective.


 
 I would _*LOVE *_to have you attitude regarding cables.  
  
 This is to say that you can not hear difference in conductor material, in dialectric material, ... ALL of which can, at least partially, be measured thus verifying at least some differences heard. 
  
 Where cables become KILLER important is for phono, particularly for the best low output moving coil cartridges. When these spit out mere 0.04 mV/5cm/sec (0 dB, max + 18 dB ), cable better be GOOD. There is a reason why really good discontinued phono cables now command higher price when they were new - people do not exactly crave to part with $ for no reason.


----------



## bigshot

All cables that are properly functioning should sound the same *by definition*. The purpose of a cable is to conduct the electrical impulses faithfully. Any two cables that faithfully conduct will sound the same by definition. So if you are finding that cables sound different, you are getting cables that don't conduct faithfully... you are buying lousy cables.
  
 I was working with a first class sound studio in Hollywood, and got the chance to get a tour from the chief engineer who had designed the whole setup. He opened a panel in a wall and showed me the bundles of cables that connected all the recording and mixing stages together. I asked him if he used high end cables. He laughed and opened a closet door and showed me a giant spindle of cable. He told me that he buys cable and connectors in bulk from Monoprice. The whole studio was wired with it. That's why I buy cables from Monoprice. They cost a dollar or two apiece and they do a perfect job. I've bought everything from audio cables to HDMI from them. Always perfect. Always cheap.


----------



## bfreedma

bigshot said:


> All cables that are properly functioning should sound the same *by definition*. The purpose of a cable is to conduct the electrical impulses faithfully. Any two cables that faithfully conduct will sound the same by definition. So if you are finding that cables sound different, you are getting cables that don't conduct faithfully... you are buying lousy cables.
> 
> I was working with a first class sound studio in Hollywood, and got the chance to get a tour from the chief engineer who had designed the whole setup. He opened a panel in a wall and showed me the bundles of cables that connected all the recording and mixing stages together. I asked him if he used high end cables. He laughed and opened a closet door and showed me a giant spindle of cable. He told me that he buys cable and connectors in bulk from Monoprice. The whole studio was wired with it. That's why I buy cables from Monoprice. They cost a dollar or two apiece and they do a perfect job. I've bought everything from audio cables to HDMI from them. Always perfect. Always cheap.




+1

Though I have to admit I prefer the connectors from Blue Jeans for cables that go through a lot of plug in/out cycles. A little more robust, no difference in "sound".


----------



## analogsurviver

bigshot said:


> All cables that are properly functioning should sound the same *by definition*. The purpose of a cable is to conduct the electrical impulses faithfully. Any two cables that faithfully conduct will sound the same by definition. So if you are finding that cables sound different, you are getting cables that don't conduct faithfully... you are buying lousy cables.
> 
> I was working with a first class sound studio in Hollywood, and got the chance to get a tour from the chief engineer who had designed the whole setup. He opened a panel in a wall and showed me the bundles of cables that connected all the recording and mixing stages together. I asked him if he used high end cables. He laughed and opened a closet door and showed me a giant spindle of cable. He told me that he buys cable and connectors in bulk from Monoprice. The whole studio was wired with it. That's why I buy cables from Monoprice. They cost a dollar or two apiece and they do a perfect job. I've bought everything from audio cables to HDMI from them. Always perfect. Always cheap.


 
 I did _specify clearly _where any of my cables for 2-3 Eur ( Tasker, Italy ) or Monoprice would have poo-pooed themselves. Please read the figures again - and THINK before saying that anything remotely likely to be considered by Monoprice would ever be capable of transmitting that signal to the input without causing major SNAFU.
 I am not against Monoprice (except they do not deliver outside US, therefore useless for me - or if they do, shipping kills any prospect of the deal ), they are doing their job splendidly - but past certain quality, one no longer can operate in best bang for the buck mode - like it or not.
  
 If indeed run on Monoprice cables, that studio is not first class - period. At least the likes of Mogami cable is reasonable to expect in a first class studio - if not better.
 Monoprice would "pass" only if OEM is de facto Mogami. And Mogami itself is 1/10th or less of the cost of REALLY good cables - which due to the prohibitive cost, almost never get used in the studio. These would make quite an audible difference - because if nothing else, will be made of materials that age FAR better than cheap cable. As the deterioration is slow and unoticable on day to day basis, in couple of years (or decade(s)) lower quality cables basically rot away, requiring eventually the complete exchange of all cables.
  
 With their "soundmark" stamped on every recording ever made in that studio...


----------



## analogsurviver

bfreedma said:


> +1
> 
> Though I have to admit I prefer the connectors from Blue Jeans for cables that go through a lot of plug in/out cycles. A little more robust, no difference in "sound".


 
 Please look at the requirement/level I have posted to be critical for cables - it is not CD to amp where almost anything goes ...


----------



## bfreedma

analogsurviver said:


> bfreedma said:
> 
> 
> > +1
> ...




No thanks. I'm interested in actual requirements for audible transparency and both Monoprice and BJC more than achieve that standard.


----------



## bigshot

analogsurviver said:


> If indeed run on Monoprice cables, that studio is not first class


 
  
 I don't think you're really qualified to judge things that you don't know about, to be honest.


----------



## analogsurviver

bigshot said:


> I don't think you're really qualified to judge things that you don't know about, to be honest.


 
 Sorry, cables in home audio or personal audio have very short runs. Cables in studios are completely different matter - they are LONG. And the quality of the cable does matter more for long runs than for a meter or so.
  
 Mogami - and to a bit lesser degree Tasker - cables are very consistent batch to batch and the deviations from their specs are minimal. Specially capacitance which is critical for the longer runs.
  
 I have been forced once to buy C112  cable from some other manufacturer - because Tasker got sold out and I needed cable right away. Only about three times capacitance than the Tasker - at about 90 % the price ... - 100% loss in financial terms as I used it only once and with predictably poor results.
  
 Similar happened to Technics in (one, more ? ) of their last batches of the SL 1200 turntables. They run out of cable they were using for ages - replacing it with an optical "equivalent". Same story - instead of approx <150 pF, the new cables were around 500 pF - rendering the TT unsuitable for most moving magnet cartridges, including Technics' own. Anyone not hearing the havoc this excessive capacitance works with sound must be deaf.
  
 I am not 100% sure that Monoprice cable is crap - I have never seen a Monoprice product in my life due to being on the wrong continent for that - but experience with cables and cables is as described above. Mogami is NOT a high end cable with crazy prices - but is what I consider minimum to be able to safely rely on it. Specially in pro environment. You can visit any site regarding studios etc and will see how widespread Mogami really is . For a reason.
  
 One can get fancy with materials etc - but then the price will be some 10 times the Mogami "equivalent". Would the use of such cable put the studio you mentioned in _hors de categorie _class ?


----------



## bigshot

I still don't think you're qualified to comment on the quality of a recording studio, for more than one reason, chief of which being you have never worked with them.


----------



## jugate

bigshot said:


> marketing. same with "HD" audio.


 
  
 yep but with something technicaly true inside
  
 THE point is, digitally speaking the format of 48/24 or greater is better than what we had 20 or 30 years ago when the cd was invented.
 Now, we are sufficiently audiophile / demanding in terms of audio to tell the difference?


----------



## jugate

kraken2109 said:


> If expensive cables don't improve sound then why do they exist?
> 
> Money.


 
   
 balanced cables (with o/i devices) make a diference, it is a fact.
  
 The "hd" or "hi-res" format has more bit and khz, it is a fact.
  
 money always exist, but in this case would have to look beyond. Behind the marketing can be something true and concrete
  
  
  
 Quote:


bigshot said:


> marketing. same with "HD" audio.


 
  
 yep but with something technicaly true inside. The marketing it is on a audio format that are not even new (remember sacd/audio dvd), like a zombie mark.
  
 THE point is, digitally speaking the format of 48/24 or greater is better than what we had 20 or 30 years ago when the cd was invented.
 Now, we are sufficiently audiophile / demanding in terms of audio to tell the difference?


----------



## jugate

kraken2109 said:


> If expensive cables don't improve sound then why do they exist?
> 
> Money.


 
   
 balanced cables (with o/i devices) make a diference, it is a fact.
  
 The "hd" or "hi-res" format has more bit and khz, it is a fact.
  
 money always exist, but in this case would have to look beyond. Behind the marketing can be something true and concrete
  
  
  
 Quote:


bigshot said:


> marketing. same with "HD" audio.


 
  
 yep but with something technicaly true inside. The marketing it is on a audio format that are not even new (remember sacd/audio dvd), like a zombie mark.
  
 THE point is, digitally speaking the format of 48/24 or greater is better than what we had 20 or 30 years ago when the cd was invented.
 Now, we are sufficiently audiophile / demanding in terms of audio to tell the difference?


----------



## sonitus mirus

jugate said:


> Now, we are sufficiently audiophile / demanding in terms of audio to tell the difference?


 
  
 Human hearing capabilities have not changed in the last 20-30 years.  With respect to our hearing abilities, a CD was "sufficiently audiophile" then as it is now.


----------



## analogsurviver

bigshot said:


> I still don't think you're qualified to comment on the quality of a recording studio, for more than one reason, chief of which being you have never worked with them.


 
 True - because I never liked what came out of them in the first place. None of them. Did not sound anything approaching live sound. Actually, lurking on some forums about studios etc to learn what to AVOID at all costs - and % of these things is growing by the day. I am the diametral opposite of the guy who is over the moon with a yet another new component to allow him to manipulate the sound. It took me over two years before I heard some samples posted that did approximate the sound I am after -
 and strangely enough, equipment used was perhaps 1% worse than the current "standard" - at one fourth or so of the cost. In plain English - under real world conditions indistinguishable.
  
 But at least I know a decent - not exorbitant - cable is a must. I am perfectly aware the quantity of cables and their length in a studio present a sizeable % of  entire cost - so keeping the price of cable down is a reasonable decision. Question is - how much down not to compromise quality ?
  
 Signal will be only as good as the cable will transmit - no after the fact gizmo can put it right again. One can put filters for hum, RFI etc on the inputs - but as always, prevention is better than cure.


----------



## lamode

jugate said:


> placebo or deafness, that is the question... (willy say)
> 
> 
> but again, if anything more that 44.1/16 is bullshet then why exist dolby trueHD, dta MA and thats stuffs..


 
  
 Formats like Dolby TrueHD support 16/44 as well. They just support other bit rates and sampling frequencies for maximum compatibility.
  
 And don't forget the power of numbers in marketing. That's reason enough for HD to exist.


----------



## jugate

lamode said:


> Formats like Dolby TrueHD support 16/44 as well. They just support other bit rates and sampling frequencies for maximum compatibility.
> 
> And don't forget the power of numbers in marketing. That's reason enough for HD to exist.


 
  
 dolby digital is 16/48. What would be the difference with dolby hd?


----------



## bigshot

analogsurviver said:


> True - because I never liked what came out of them in the first place.


 
  
 I am a professional. I work with professional studios, staff and equipment. I guess that is the difference between us here.


----------



## analogsurviver

bigshot said:


> I am a professional. I work with professional studios, staff and equipment. I guess that is the difference between us here.


 
 Yes, it is. When I decided to to start recording, it was a firm decision not to go this route. Too much of the same I did not like - at all.
  
 The more studio equipment evolved, the more possibilities to manipulate sound became available. And with toys at their disposal, people play - whether it is in the interest of the music or not.
  
 I knew that the minimalistic approach I use will be limited to acoustic music, good sounding halls, etc - I am struggling with myself to decide to get some cardioid mics for recording in lesser halls - which are generally unsuitable for work with omnis, but seldom are concerts held in the venues I would really like to work in...
  
 From pure SQ point of view - NO even in the wildest nightmares ; from realities of real life and making a living - yes. 
  
 But I could perhaps never come to terms of working in a studio - it is way too artificial for my taste. At least for acoustic music. Genres that without mixing almost do not exist are another matter.


----------



## bigshot

Well, when I work,  I'm getting paid to do a job professionally. I have specific standards I have to meet for broadcast or CD release. I'm not just recording for the fun of it. (not that I don't have fun sometimes) It would be a lot easier to just be a duffer, but I have to adhere to functional factors as well as creative. The price of professionalism.


----------



## analogsurviver

bigshot said:


> Well, when I work,  I'm getting paid to do a job professionally. I have specific standards I have to meet for broadcast or CD release. I'm not just recording for the fun of it. (not that I don't have fun sometimes) It would be a lot easier to just be a duffer, but I have to adhere to functional factors as well as creative. The price of professionalism.


 
 I fully understand that. It is a high price to pay.
  
 But the limitations for the CD - and particularly broadcast - remind me of my high school days and our "disco" that was far too crammed for so many people. One could almost not "dance" but stepping on the the very same spot - unless you were a beautiful chick or senior equivalent of Arnold Schwarzenegger. And we , of course, adapted to this "one spot disco style".
  
 It was hilarious to the extreme to meet fellow high school folks in the summer during vacation in a very popular resort by the sea with gigantic dance floor - one could literally practice roller skate disco if so desired. 
  
 We were still performing our "one spot dance".... - one could spot us from the greatest of distance.
  
 This is how I sometimes perceive the studio people - so in the mould they usually do not even dare to think differently. "One spot (studio) dance"...


----------



## Krutsch

bigshot said:


> Well, when I work,  I'm getting paid to do a job professionally.


 
  


analogsurviver said:


> We were still performing our "one spot dance".... - one could spot us from the greatest of distance.


 
 No matter what @bigshot posts, @analogsurviver just does takes another hit and heads back for more... awesome thread...


----------



## bigshot

I am available to sign autographs and make appearances at children's birthday parties!


----------



## bigshot

analogsurviver said:


> We were still performing our "one spot dance".... - one could spot us from the greatest of distance.
> This is how I sometimes perceive the studio people - so in the mould they usually do not even dare to think differently. "One spot (studio) dance"...


 
  
 The difference is, I get paid to produce. You don't get paid to dance!


----------



## Greenears

This thread is about 24 bit v 16 bit not cables. Please take your dead horse to another thread.


----------



## audionewbi

I've made up my mind and after hearing 16bit files out of chord Hugo I am no longer valuing the 24 bit files like I used to however DSD still has potential to my ears if your DAC is capable of it.


----------



## bigshot

greenears said:


> This thread is about 24 bit v 16 bit not cables.


 
  
 Oh, we already concluded that there was no point to 24 bit several years ago!


----------



## analogsurviver

bigshot said:


> The difference is, I get paid to produce. You don't get paid to dance!


 
 That's what you think.
  
 There are 15 CDs entirely recorded by me plus 4 where at least one recording is by me. As always, there is more in the pipeline.


----------



## analogsurviver

bigshot said:


> Oh, we already concluded that there was no point to 24 bit several years ago!


 
 I agree with this - for the 99% of cases. But only in the case for PLAYBACK - recording is another story.
  
 I do not feel something significant can be gained by dithering either - try as you might, once you will be actually at recording for some time, you will find that finding a quiet place today is the hardest thing to do. Even if you go recording to some church in the middle of the nowhere, there will be at least planes - in addition to birds, dogs, cats, wildlife, drunk teenagers, farmers cutting trees in the adjacent valley, etc, etc. Which are all going to be much louder than 16 to 24 bit difference.
  
 Musician friends who have been to concerts in Carnegie Hall said it was regarding outside noise worse even than our Philharmonics - and I thought this was bad enough.


----------



## RRod

analogsurviver said:


> That's what you think.
> 
> There are 15 CDs entirely recorded by me plus 4 where at least one recording is by me. As always, there is more in the pipeline.


 
  
 Any available to listen to?


----------



## analogsurviver

rrod said:


> Any available to listen to?


 
 PM sent.


----------



## bigshot

analogsurviver said:


> I agree with this - for the 99% of cases. But only in the case for PLAYBACK - recording is another story.


 
  
 That is exactly what we're talking about. You finally realized that! Congratulations.


----------



## analogsurviver

bigshot said:


> That is exactly what we're talking about. You finally realized that! Congratulations.


 
 I was NEVER referring to 16 bit not being enough for playback - except for that 1% of cases that are extremely hard to realize in practice.
  
 But bandwidth limited to 20 kHz is the same as 8-9 bit PCM audio for me.
  
 Adamant about that !
  
 Then again - listening to a 192/32 float bounce from DSD128 sounds so good that I might reconsider that "bit" statement. But you got to have true master to play with and DAC that does support 192/32float well.


----------



## bigshot

Redbook is perfect sound for playback of music on home stereos.


----------



## jugate

is there some material of popular music recently released in 24 native? ( i mean recorded several years ago, not a remasterized part of "dark side of the moon")
  
  


bigshot said:


> That is exactly what we're talking about. You finally realized that! Congratulations.


 
 anyway... what are the benefits of 24 bits in recording and playback as welll?


----------



## lamode

analogsurviver said:


> I agree with this - for the 99% of cases. But only in the case for PLAYBACK - recording is another story.


 
  
 Did hell just freeze over?


----------



## StanD

lamode said:


> Did hell just freeze over?


 
 One down, a million to go.


----------



## analogsurviver

jugate said:


> is there some material of popular music recently released in 24 native? ( i mean recorded several years ago, not a remasterized part of "dark side of the moon")
> 
> 
> anyway... what are the benefits of 24 bits in recording and playback as welll?


 
 In recording, 24 bit are as first used as a "buffer" against dynamic bursts - allowing to operate below clipping and still take advantage of full dynamic range of music - which would have either clipped or drowned in noise with 16 bits - there is 6 dB dynamic range per bit, meaning 24 bit audio has 48 dB more dynamic range than 16. 
 http://en.wikipedia.org/wiki/Audio_bit_depth
  
 After the recording is dome, it can be normalized ( loudest peaks reach 0 dBFS or slightly below (usually not below -2 dBFS) - and the resulting music more often than not can fit into 16 bit.
  
 Audibility of 24 bit superiority is VERY hard to hear. It depends on track, on some there is no difference whatsoever, on some is using top notch equipment possible to hear it. It was on track to track basis for me, not something conclusive - and it was struggling and not fun doing these ABXes.
  
 But like I said, I did like listening to 192/32bit floating decimal point which was converted from DSD128 - the noise floor was phenomenally low, in audiophile jargon it achieved very good blackness. One really has to hear it to believe it - can not offer you anything more.


----------



## analogsurviver

lamode said:


> Did hell just freeze over?


 
 No - there is always that 1 % ...


----------



## dprimary

jugate said:


> is there some material of popular music recently released in 24 native? ( i mean recorded several years ago, not a remasterized part of "dark side of the moon")
> 
> 
> anyway... what are the benefits of 24 bits in recording and playback as welll?


 

 Anything from about the mid 90's on has a good chance of being recorded 24bit if it was digital. The problem is that mixed master might have ended up being recorded to a DAT at 44.1/ 16. For a few years there was a really not a standard transportable 24bit format to mix to.


----------



## lamode

dprimary said:


> Anything from about the mid 90's on has a good chance of being recorded 24bit if it was digital. The problem is that mixed master might have ended up being recorded to a DAT at 44.1/ 16. For a few years there was a really not a standard transportable 24bit format to mix to.


 
  
 Why was a special format necessary at all? They could have used external disk drives or Syquest, Jaz, etc


----------



## Matte82

I haven't read anything but the last few posts. As a sound engineer, yes but depth is only related to dynamic range. How it affects playback is the amount of room from the lowest volume sound to the loudest. 16-bit can easily achieve much more dynamic range than necessary for playback of any musical style. 

However when recording a higher dynamic range also moves the noise floor down. So if you can move the floor down, say 3 dB, on 60-80 tracks, which is common in modern pop music. The cumulative reduction in noise is totally worth it. 

Then there is also the discussion about fixed point versus floating point. But that really doesn't affect the sound quality of a well engineered mix. But floating point can help sloppy gain staging.


----------



## Greenears

audionewbi said:


> I've made up my mind and after hearing 16bit files out of chord Hugo I am no longer valuing the 24 bit files like I used to however DSD still has potential to my ears if your DAC is capable of it.




I'm willing to ABX DSD but don't have the software to down covert it to 16 bit PCM. Then the only way to do it then is but it through a player to get analog out A and then loop it through a 16 bit record and playback soundcard as B. This test was done in 2007 and nobody passed even though SACD sounded better. Conclusion it's not the format. Just convert any SACD/DSD to PCM it makes life easier and you won't lose quality. 

I wish people would boycott DSD.


----------



## RRod

greenears said:


> I'm willing to ABX DSD but don't have the software to down covert it to 16 bit PCM. Then the only way to do it then is but it through a player to get analog out A and then loop it through a 16 bit record and playback soundcard as B. This test was done in 2007 and nobody passed even though SACD sounded better. Conclusion it's not the format. Just convert any SACD/DSD to PCM it makes life easier and you won't lose quality.
> 
> I wish people would boycott DSD.


 
  
 All I want is to be able to rip the multichannel tracks off my SACDs. I have something at least than can go from DSD64 5.1 to PCM. DSD128+ is just annoying as a distribution format.


----------



## bigshot

rrod said:


> All I want is to be able to rip the multichannel tracks off my SACDs. I have something at least than can go from DSD64 5.1 to PCM. DSD128+ is just annoying as a distribution format.


 

 The only way I know of to rip multichannel SACDs is with an older model Xbox.


----------



## RRod

bigshot said:


> The only way I know of to rip multichannel SACDs is with an older model Xbox.


 
  
 I think it's a PS3 with older firmware, and I've never found a firm answer on whether it can grab the multichannel. For now it doesn't matter b/c a 5+.1 setup isn't anywhere in my immediate future. Hopefully by 2020 Sony will stop being ninnies… lol that isn't gonna happen ^_^


----------



## Roly1650

rrod said:


> I think it's a PS3 with older firmware, and I've never found a firm answer on whether it can grab the multichannel. For now it doesn't matter b/c a 5+.1 setup isn't anywhere in my immediate future. Hopefully by 2020 Sony will stop being ninnies… lol that isn't gonna happen ^_^



As someone has already said, you can go analog out on an SACD player to analog in to a 5.1 sound card. I know the Oppo players do multi channel out and have done for a good number of years. I ripped all my SACD's to PCM 24/96 and really didn't notice any degradation. Once you've got PCM you can do anything with it.

I agree with you DSD as a format is bloody annoying, the only reason it exists is Sony's paranoia about copy protection when cd-r came along.

Edited to add: if memory serves correctly, if you rip a multi channel SACD to stereo, you get the surround channels folded to the stereo channels, but I'm a bit foggy about that without digging out my Oppo owner manual.


----------



## bigshot

https://newtoolbox.files.wordpress.com/2014/03/sacd-ripper-primer-v4-0.pdf
  
 I think the reason it's such a PITA to rip SACDs is because not many people really care much about the format.


----------



## Krutsch

bigshot said:


> https://newtoolbox.files.wordpress.com/2014/03/sacd-ripper-primer-v4-0.pdf
> 
> I think the reason it's such a PITA to rip SACDs is because not many people really care much about the format.


 

 Well, it's because Sony didn't screw around with the copy protection aspect of SACD (you know more about this that I do); there are NO available writers and only two plants, world-wide. No PC drives ever released to the public - they knew what they were doing.
  
 Frankly, I would be ecstatic if we could rip these things to (very) lossy Dolby AC-3 5.1 - I have a bunch of surround SACDs that just sound fantastic in the surround system and would love to rip them.


----------



## Greenears

roly1650 said:


> As someone has already said, you can go analog out on an SACD player to analog in to a 5.1 sound card. I know the Oppo players do multi channel out and have done for a good number of years. I ripped all my SACD's to PCM 24/96 and really didn't notice any degradation. Once you've got PCM you can do anything with it.
> 
> I agree with you DSD as a format is bloody annoying, the only reason it exists is Sony's paranoia about copy protection when cd-r came along.
> 
> Edited to add: if memory serves correctly, if you rip a multi channel SACD to stereo, you get the surround channels folded to the stereo channels, but I'm a bit foggy about that without digging out my Oppo owner manual.


 

 >As someone has already said...
  
 That someone would be me.  Of course, I'm just the most recent.  I'm sure there are 20 other similar posts out there.  But I have no hands on with this, it's based on the test results.
  
 I do however have hands-on with 5.1-to-2 channel downmix as it is called.  This is a standard feature of all surround systems.  Your SACD player can do it, but if you're going to all the trouble of real-time analog rips you may as well grab the 5.1 it will take no more effort.  The downmix algorithm is pretty straightforward but finding reasonable software that will do it may not be so easy.   @stv014 aka Quantize-man is pretty handy with C code, maybe he can oblige?  Anyway if you find the software you end up with both a 5.1 and 2 channel 16 bit PCM and you are good to go.  No more hassle with plastic, no loss in quality, play it anywhere.


----------



## dprimary

lamode said:


> Why was a special format necessary at all? They could have used external disk drives or Syquest, Jaz, etc


 
 My first drive that could hold a complete album in the early to mid 90's was $1400. If the mastering was being done out of town you where shipping an expensive drive that you hoped to get back and not damaged in shipping. I didn't have a bunch laying around. Compare to today when you throw out a 1 gig usb drive as useless. We would back it up to data tape or burn it to CD's. At $150 an hour and up you did not want to pay the mastering engineer to spend time be moving files around. A CD master then was a 3/4" videotape recorded through 1630 processor. That would be shipped by the mastering studio to the duplication plant who would create the glass master for pressing. It was about the 96 when I could burn a fully rebook compliant CD-R with index and track points those you had to burn at real time and then later at 2 times speed.


----------



## bigshot

Back in those days, I always laid down to four track ADAT. Before that, it was on Beta. I worked on the first digitally recorded television program. It was recorded to those Beta two part things. Can't remember what the machines were called. They were made by Sony. Sorry.


----------



## stv014

For converting DSD format audio to PCM, I have created a simple utility that can be downloaded from here: dffconv.zip. It also requires the dsputils.zip package, copy the .exe file to "bin", and the .cpp (if you want to compile it) to "src". The usage of the program is:
  

```
dffconv INPUT_FILE.DFF OUTPUT_FILE.WAV [OCTAVES [LPFREQ [GAIN [BWMULT]]]]
```
  
 INPUT_FILE.DFF is the input file in DFF (DSDIFF) format. Compressed (DST) files are not supported, and the channel configuration chunk is ignored. That means multi-channel files can be converted, but the output file will simply contain all channels in the same order as they were in the DFF. A sample in this format that I used for testing can be found here.
  
 OUTPUT_FILE.WAV is the output file in standard WAV format. It is always 24-bit PCM unless OCTAVES is 0 (no downsampling), in which case the output resolution is 8-bit instead.
  
 OCTAVES (optional) is the number of octaves by which the DSD input is downsampled. It defaults to 5, or 32x downsamping to convert DSD64 to 88.2 kHz. The conversion algorithm is identical to "resample -q 7" with an integer power of two ratio, and linear phase filter. OCTAVES can be zero to disable sample rate conversion and write a 8-bit output file with no processing.
  
 LPFREQ (optional, ignored if OCTAVES=0) is the -6.02 dB corner frequency of the linear phase windowed sinc lowpass filter, as a fraction of the output sample rate in the range 0.25 to 0.4999. A value closer to 0.5 results in faster roll-off to -160 dB at the Nyquist frequency, and longer impulse response. The default setting is 0.46. For 44100 Hz output, it can be set to 0.48 to approximate the default filter of the SoX resampler.
  
 GAIN (optional, ignored if OCTAVES=0) defaults to 1, other values multiply the output level by the specified number. If the downsampled DSD has lower than 0 dBFS peak level, the gain can be greater than 1 without clipping.
  
 BWMULT (optional, ignored if OCTAVES=0) multiplies the width of the transition band of the lowpass filter. It defaults to 1, and and the allowed range is 1 to 1.5. 160 dB attenuation at the Nyquist frequency is likely to be overkill in practice, so the roll-off can be made slightly slower with this parameter without the risk of significant aliasing.
  
 To compile the program on Linux, use the following command:
  

```
g++ -Wall -O2 -fomit-frame-pointer -ffast-math -DUSE_SIMD=1 -DUSE_OOURA_FFT=1 -I. dffconv.cpp -o dffconv -lsndfile -lm -s
```
  
 On 32-bit x86, *-march=native -mfpmath=sse -msse -msse2* can be added to enable the use of SSE2.


----------



## jugate

matte82 said:


> I haven't read anything but the last few posts. As a sound engineer, yes but depth is only related to dynamic range. How it affects playback is the amount of room from the lowest volume sound to the loudest. *16-bit can easily achieve much more dynamic range than necessary for playback of any musical style.*
> 
> However when recording a higher dynamic range also moves the noise floor down. So if you can move the floor down, say 3 dB, on 60-80 tracks, which is common in modern pop music. The cumulative reduction in noise is totally worth it.
> 
> Then there is also the discussion about fixed point versus floating point. But that really doesn't affect the sound quality of a well engineered mix. But floating point can help sloppy gain staging.


 
  
 and what about 44.1 vs 96 khz?
  
 i found that 48 is a best sound (dolby/dts) but is there sometihng better with 96 khz (again, talking abount native recorded, not mastering upsampling)


----------



## Matte82

On paper, sample rate mostly determines the highest frequency that can be recorded or played back. Look up nyquist theory for more info. 44.1 can easily get up to 20khz. Which is the theoretical limit of human hearing. But most people start to drop off in the 14-16khz range. And some even sooner. Fwiw 10khz is the top end of cymbals and really quiet high. 14-16k is what gives sounds "air". 

However there are some claims that even though we can't measure our hearing above this range. That there is sonic info included in some sounds well above 20k and our ears still "feel" it. Even though we can't actually measure response in that range. I've seen some tests done up to 50khz. Which as far as I know is pushing how far equipment can even measure. And there is sonic activity in live music above 20k. So it would take a 96khz sample rate to pick up roughly 48 kHz sounds. Provided the recording equipment has the ability to go that high. And the playback system has to be able to reproduce it. Even if something can record or playback 96k audio files, that doesn't mean there isn't a low pass filter somewhere along the way cutting off any sounds above 20k. 

And then there is the matter of jitter. This is getting into the tech side. Basically every digital device has a clock that is telling the converters when to either record or play back a sample. (Hence sampling rate). How evenly spaced those samples are, is the measure of jitter. Higher end equipment tends to have better designed clocks with less jitter. However jitter can be reduced in almost any recording or playback devices by using higher sample rates. So not only does higher sample rates improve frequency response, but it can lower jitter. Which makes sound clearer and cleaner. 

But now the bigger question is, which of those is really making the difference in the sound quality. The higher theoretical frequency response, or the lower jitter? I'm really not sure. And there's a lot of double blind tests that say we can't hear these differences. But anyway that's my .02 on the subject.


----------



## jcx

increased sample rate at the same bit depth does lower audio frequency noise by spreading some of the quantization noise over inaudible ultrasonic frequencies - noise shaped dither can take advantage of the extra bandwidth to push even more audio frequency noise beyond our hearing range
  
 16/96 would be the much better direction in my opinion for hi rez instead of the stupid 24/44.1 Beatles offering
  
  
 jitter's theoretical effects vs sample frequency depends on the DAC technology - BitStream DSD is more sensitive than multibit DACs, audio DAC are all NRZ but jitter effects are reduced with balanced pulse Return to Zero schemes used in some industrial, RF DAC


----------



## bigshot

For the purposes of mixing and mastering, a lower noise floor is MUCH more useful than higher sampling rates. A lower noise floor means you have more room to boost an element in the mix without dragging noise up with it. Frequency response above the range of human hearing is notoriously spiky in a lot of electronics, and in playback inaudible spikes can cause harmonic distortion lower in the audible range. Because of this, it's standard procedure when mastering to put music through a low pass filter to get rid of inaudible frequencies. They can only hurt sound quality. They can't improve it.


----------



## Greenears

stv014 said:


> For converting DSD format audio to PCM, I have created a simple utility that can be downloaded from here: dffconv.zip. It also requires the dsputils.zip package, copy the .exe file to "bin", and the .cpp (if you want to compile it) to "src". The usage of the program is:
> 
> 
> ```
> ...


 
  
 Whoa..... we're not worthy!  (doing the Wayne's world bow-down frantically)


----------



## analogsurviver

matte82 said:


> On paper, sample rate mostly determines the highest frequency that can be recorded or played back. Look up nyquist theory for more info. 44.1 can easily get up to 20khz. Which is the theoretical limit of human hearing. But most people start to drop off in the 14-16khz range. And some even sooner. Fwiw 10khz is the top end of cymbals and really quiet high. 14-16k is what gives sounds "air".
> 
> However there are some claims that even though we can't measure our hearing above this range. That there is sonic info included in some sounds well above 20k and our ears still "feel" it. Even though we can't actually measure response in that range. I've seen some tests done up to 50khz. Which as far as I know is pushing how far equipment can even measure. And there is sonic activity in live music above 20k. So it would take a 96khz sample rate to pick up roughly 48 kHz sounds. Provided the recording equipment has the ability to go that high. And the playback system has to be able to reproduce it. Even if something can record or playback 96k audio files, that doesn't mean there isn't a low pass filter somewhere along the way cutting off any sounds above 20k.
> 
> ...


 
 Great post !
  
 Just an update to current technology - there is a microphone capable of 100 kHz response (Sanken C100K), there is a recorder capable of recording it ( Merging Technology Horus and Hapi in DSD256 mode; exact response is not known (yet), but it certainly could reach 100 kHz @ - 3 dB if not better and a gentle roloff above, no brick filtering as required by PCM  ). 
  
 One has to ENSURE there is no low pass filtering "somewhere". In my cassette day recording, I made absolutely sure there is NO low pass  (or high pass for that matter ) filtering anywhere. Very few cassette decks allow for this, as only a handful allow the signal without bias etc to be put out at all. It is only available during playback, monitoring while recording does require using such filter. I was and occasionally still am using this cassette deck - Technics RS-AZ 7. Since I do not like Dolby A to Ž, I use Telefunken>Nakamichi High Com II for noise reduction. Naka has copy pasted much of the original design, all the flaws and omissions included. Whereas  Technics deck is "transparent", High Com II is not - even if MPX filter is switched off, it still represents low pass filter - which IS audible. I had to remove and short circuit the coils of the MPX filter to finally arrive at the "open" sounding noise reduction.
  
 Due to the above mentioned requirement of having to use low pass filter during the recording/monitoring, the sound is never as open and lifelike than upon playback.
 Removing these two otherwise "hidden and permanent" high pass filters turns otherwise cassette's cat meow into lions's roar. Needless to say, MPX filter on the deck itself is also always set off when recording from microphone feed.
  
 To those studio cats with big R2R machines - NO R2R ever made can compete in the bass with very few cassette decks fitted with amorphous heads. These came about long time after the last newly designed R2R was made. In the bass - zero irregular response ( no "head bumps"), output limited by the electronics - to DC if required.  This is the prime reason why a R2R machine set up in an ABX is so easy to spot in a straight wire bypass test - bass will reveal it instantly. An amorphous head cassette machine is next to impossible to tell from the original as far as bass is concerned.


----------



## bigshot

Of course you're talking about using home equipment for amateur recordings, not studio R2R decks. (And Teac made a consumer R2R with cobalt amorphous heads.)


----------



## analogsurviver

bigshot said:


> For the purposes of mixing and mastering, a lower noise floor is MUCH more useful than higher sampling rates. A lower noise floor means you have more room to boost an element in the mix without dragging noise up with it. Frequency response above the range of human hearing is notoriously spiky in a lot of electronics, and in playback inaudible spikes can cause harmonic distortion lower in the audible range. Because of this, it's standard procedure when mastering to put music through a low pass filter to get rid of inaudible frequencies. They can only hurt sound quality. They can't improve it.


 
 That is WHY I do not like studios with their spiky and iffy electronics. Which is a direct consequence of the "above 20 kHz does not matter"- adopted also in studio equipment. And that is why I prefer audiophile gear that is usually MUCH better designed than studio gear.
  
 The studio gear that is OK is usually beyond the reach of almost anybody. Starting with microphone preamps...
  
 I agree there is no such thing as too low noise floor - it always comes handy.


----------



## Greenears

I think if we turn @stv014 loose on this 24 vs 16 problem, with help from a few recording engineers that have access to high quality mics and 24 bit recording chain, we should be able to create a series of ABX setups & fles and post them as a challenge to all comers to get the mythical 9/10 ABX log of any equipment of their choosing.  Frankly this whole question is relatively easily solved, the puzzle is why nobody really wants to solve it
  
 Anyhow, @stv014 you reference a DSD file that I think is a needle drop from an LP.  Any recommendations on "good" DSD files legitimately downloadable that were recorded DSD natively that we can use your new DSD conversion utility to do a DSD vs PCM24 vs PCM16 vs PMC12 ABX?
  
 I haven't tried DSD ABX since I thankfully never went down the DSD path so I don't have any DSD files.  But, there does seem to be a fair bit of consensus out there that there are some very nice sounding SACD/DSD recordings.  I would never want a DSD formatted file in my library, but if your DSD2PCM conversion util ABX's out to strong fail on all of the above (as all the previous data indicates it will) then I might be tempted to buy a few DSD files and convert them to PCM if they sound really good.


----------



## RRod

greenears said:


> I think if we turn @stv014 loose on this 24 vs 16 problem, with help from a few recording engineers that have access to high quality mics and 24 bit recording chain, we should be able to create a series of ABX setups & fles and post them as a challenge to all comers to get the mythical 9/10 ABX log of any equipment of their choosing.  Frankly this whole question is relatively easily solved, the puzzle is why nobody really wants to solve it


 
  
 Pretty much all the 24bit recordings I have just end up giving distortions at about -120dB when truncated to 16bit and re-converted. No one is going to hear that, which is why nobody bothers to do the tests on either side. The "I love 24-bit side" just doesn't believe in ABX, and the "16-bit is enough side" knows that no one is going to hear -120dB differences.
  
 And thanks for the DSD program, stv. Can DSD128+ be handled just by playing appropriately with the octaves parameter?


----------



## analogsurviver

greenears said:


> I think if we turn @stv014 loose on this 24 vs 16 problem, with help from a few recording engineers that have access to high quality mics and 24 bit recording chain, we should be able to create a series of ABX setups & fles and post them as a challenge to all comers to get the mythical 9/10 ABX log of any equipment of their choosing.  Frankly this whole question is relatively easily solved, the puzzle is why nobody really wants to solve it
> 
> Anyhow, @stv014 you reference a DSD file that I think is a needle drop from an LP.  Any recommendations on "good" DSD files legitimately downloadable that were recorded DSD natively that we can use your new DSD conversion utility to do a DSD vs PCM24 vs PCM16 vs PMC12 ABX?
> 
> I haven't tried DSD ABX since I thankfully never went down the DSD path so I don't have any DSD files.  But, there does seem to be a fair bit of consensus out there that there are some very nice sounding SACD/DSD recordings.  I would never want a DSD formatted file in my library, but if your DSD2PCM conversion util ABX's out to strong fail on all of the above (as all the previous data indicates it will) then I might be tempted to buy a few DSD files and convert them to PCM if they sound really good.


 
 What @stv014 did is great.
  
 But you will never get the whole quality of the DSD file is capable of by converting it to PCM - it is a lossy process. 
  
 Then again, if you at least keep the PCM at least at 88.2 kHz sampling, DSD bounced to 88.2 will still sound better than CD Redbook. 
  
 Whether DSD or PCM, it is always better to stick to the original file played natively than using conversion.


----------



## analogsurviver

bigshot said:


> Of course you're talking about using home equipment for amateur recordings, not studio R2R decks. (And Teac made a consumer R2R with cobalt amorphous heads.)


 
 Studio R2R decks. All of which predate amorphous heads by more than enough.
  
 But I admit, did not know Teac made consumer R2R deck with amorphous heads. That should be interesting.
  
 My dream R2R deck would be Technics 1500 or 1700 mechanics (wow and flutter better than anything else ) with amorphous heads and up to date electronics.


----------



## bigshot

Amorphous heads were an option on Nagras.


----------



## analogsurviver

bigshot said:


> Amorphous heads were an option on Nagras.


 
 Did not know that. At the time, Nagra was so above my budget I really did not want to look at it. Still is.
  
 But today, in times of DSD, there are other manufacturers offering similar or better quality for less money. It no longer offers the same supremacy as in analog days.


----------



## Greenears

rrod said:


> Pretty much all the 24bit recordings I have just end up giving distortions at about -120dB when truncated to 16bit and re-converted. No one is going to hear that, which is why nobody bothers to do the tests on either side. The "I love 24-bit side" just doesn't believe in ABX, and the "16-bit is enough side" knows that no one is going to hear -120dB differences.
> 
> And thanks for the DSD program, stv. Can DSD128+ be handled just by playing appropriately with the octaves parameter?


 
 Well @RRod, we've disagreed on this about 24 times?   Or was it 42?
  
 I don't think that's a valid reason not to do updated tests with 2014 or newer vintage equipment.  By not doing it you leave that little sliver of doubt.


----------



## RRod

greenears said:


> Well @RRod, we've disagreed on this about 24 times?   Or was it 42?
> 
> I don't think that's a valid reason not to do updated tests with 2014 or newer vintage equipment.  By not doing it you leave that little sliver of doubt.


 
  
 To me the bigger sliver is accepting a 95% confidence level and no statistical power specification


----------



## stv014

> And thanks for the DSD program, stv. Can DSD128+ be handled just by playing appropriately with the octaves parameter?


 
  
 I have not tested it, but if the only difference is the sample rate (2.8224/5.6448/11.2896 MHz), then it should work fine in theory. If the file format is not DSDIFF, or the data is compressed (DST format), that would be a problem.


----------



## stv014

> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> But you will never get the whole quality of the DSD file is capable of by converting it to PCM - it is a lossy process.


 
  
 Fortunately, the loss is usually just high frequency shaped noise. This is what the spectrum of a short section of the above linked DSD64 sample looks like at about 2:06 (I tried to find a part where there is as much ultrasonic content as possible, using the spectrogram view):

 For comparison, a "quieter" part is also shown in green, and a 176.4 kHz conversion with a slow roll-off in red. Above 40 kHz, the noise floor increases so much that it swamps anything that could be useful even to a theoretical human with bat-like ears. This could be tested by shifting the 50-70 kHz band into the audible range and listening to it. Arguably, removing the noise is beneficial because it could be a source of IMD issues in some amplifiers. That is why it is recommended for DSD playback to implement a lowpass filter at 50 kHz.
  


> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> Then again, if you at least keep the PCM at least at 88.2 kHz sampling, DSD bounced to 88.2 will still sound better than CD Redbook.


 
  
 Have you tried an ABX test between DSD->88.2 and DSD->88.2->44.1->88.2 (or even 176.4 instead of 88.2) ? Since the comparison is now between PCM files of identical format, that should not be difficult.


----------



## wehaveyourpants

bigshot said:


> For the purposes of mixing and mastering, a lower noise floor is MUCH more useful than higher sampling rates. A lower noise floor means you have more room to boost an element in the mix without dragging noise up with it. Frequency response above the range of human hearing is notoriously spiky in a lot of electronics, and in playback inaudible spikes can cause harmonic distortion lower in the audible range. Because of this, it's standard procedure when mastering to put music through a low pass filter to get rid of inaudible frequencies. They can only hurt sound quality. They can't improve it.


 
  
 Huh? Have you done much mixing and/or mastering? Level control (fader) is not a boosting mechanism. While I'm at it, the word boost doesn't belong in audio discussion outside of what an engineer does during tracking. Even then you are not supposed to boost, but we do it anyways. You can boost your sales, boost your ego and boost your kid's seat but STANDARD practice in the studio is to CUT CUT CUT... 
  
 You and your crazy distortion theories are killing me. Harmonics only happen in one direction and that's up. Harmonic distortion is caused by clipping and the presence of harmonic order indicates the symmetry of the clip compared between the PEAK (top) and the TROUGH (bottom). A symmetrically clipped wave will produce only odd numbered harmonics and the presence and amplitude of even order harmonics provides clues to the cause of the clipping that occurs. However... None of this is necessarily a bad thing. Harmony is a beautiful thing when it's in phase and predictable.
  
 You say that it is standard practice to "get rid" of inaudible frequencies during mastering... Are you sure? What happens to the rest of the collective waveform when you "get rid" of a part of it? Are you sure filtering low frequencies in the mastering booth is intended eliminate inaudible sound?
  
 Those were rhetorical of course, because the reason (if an engineer decides to cut to sub-20 range period) is to provide a natural boost to the remaining wave. Harmonic distortion is only bad in two scenarios: It sums and diminishes the presence of the fundamental frequency (Muddy) or it is out of phase causing the sinusoidal fundamental wave to distort (Attenuate)... Other than that harmonies ROCK. Want some cross-genre, generational-gap bridging proof?
  
 Cash-Carter, Supremes, Donny & Marrie, Simon & Garfunkel, The Eagles, Boyz 2 Men, TLC... Any Catholic Boys choir. The reason why these harmonies are pleasant and the electrical kind are not is entirely due to phasing. Humans have brains that allow them to remain dynamically in phase while controlling amplitude if your amp could do this then people would stop saying harmonic distortion and just say harmonics that I don't want in my mix. We enjoy vocal harmonies, duel pianos and battling guitars but two bass players is too much... Even if the second one only turns up to 7.
  
 Bandpass filters do not eliminate the wave. This is a bad visual. Instead, filters push down the waves at the peak by a measurable range (attenuate). 1st order (12db) 2nd order (24db) and so on. Because a filter is incapable of eliminating the waves mass, a gain is measured on the output (low-pass) or input (high-pass) sides. In other words, when you attenuate one frequency you boost the others.
  
 Coming full circle, this is why you should never boost a signal via EQ... When you do you attenuate the rest of the wave. Has to come from somewhere right? While this may seem pleasant at first it will over time increase listening (or frequency) fatigue. When you boost in the studio it is always on a percussion instrument and never matters because it is an isolated boost on a single channel.
  
 Harmonics from inaudible sub-frequencies will harmonize at audible frequencies and as long as they are in phase will not be noticeable apart from the slight increase in volume at that frequency. Too much low frequency harmonics can definitely get out of hand but is not BAD.
  
 Typically an engineer will filter sub 20Hz frequencies in an effort to improve the max level of the recording without amplifying the noise ceiling or to eliminate sub-sonic harmonics that cause phase shift which in turn lead to intermodulation and artificial attenuation at the harmonic order. If a 100Hz wave produces a third order harmonic distortion because it was symmetrically clipped and that harmonic is out of phase with the 300Hz waves being produced at the same time it will distort the 300Hz band as long as it is present.
  
 Harmonics are a result of clipping, which is the most notorious of the 6 deadly distortions... Comparably harmonics are like women: most of the time they are beautiful and the rest of the time they talk all over your music.
  
 Here's an article that should help:
  
 http://www.aes-uk.org/forthcoming-meetings/harmonic-phase-the-missing-factor-in-distortion-measurement/
  
 -Jason
  
 Just kidding just in case.


----------



## limpidglitch

I think bigshot meant _in_harmonic distortion, like IMD, which does occur and can fold downwards.
 None-the-less, interesting info on why one would want to cut infra-sonic frequencies.


----------



## analogsurviver

stv014 said:


> Fortunately, the loss is usually just high frequency shaped noise. This is what the spectrum of a short section of the above linked DSD64 sample looks like at about 2:06 (I tried to find a part where there is as much ultrasonic content as possible, using the spectrogram view):
> 
> For comparison, a "quieter" part is also shown in green, and a 176.4 kHz conversion with a slow roll-off in red. Above 40 kHz, the noise floor increases so much that it swamps anything that could be useful even to a theoretical human with bat-like ears. This could be tested by shifting the 50-70 kHz band into the audible range and listening to it. Arguably, removing the noise is beneficial because it could be a source of IMD issues in some amplifiers. That is why it is recommended for DSD playback to implement a lowpass filter at 50 kHz.
> 
> ...


 
 This is precisely WHY I said DSD64 ( SACD ) should have NEVER been let out. The first really usable DSD is DSD128 - and it still is not "enough". But it is some 20 dB quieter at the same frequency (say 40 kHz) than DSD64. By the time DSD512 is reached, any noise should be low enough in level and high in frequency to allow for a "quiet" performance to 100 kHz without noise above that to be a problem for IMD with amplifiers, even without now mandatory DSD low pass filter.
  
 Your graphs make DSD look good - looking at the display of an oscilloscope to a signal at say -30 or so dB with DSD64 without filtering reveals nothing - but noise...
  
 I did write on head-fi that SACD/DSD64 IS a problem with direct drive high voltage amps for electrostatics (which are overstretched in the high frequencies by the nature of the load and legally limited power level to begin with ). I heard/observed it with SACD players ( like ...What !!!....), I did hear/observed it with Korg MR series of recorders ( MR1 is a good example, it is DSD64 only and offers zero filtering and a sharp and shallow filter - each sounding distinctly audibly different from each other ). DSD128 is less of a problem, but...-see above.
  
 I agree it is a tradeoff among as unperturbed as it gets frequency response (DSD) and low noise above audio spectrum (PCM ). Once we get to DSD512, this tradeoff will for all practical purposes cease to exist. PCM file of comparable size will still be ringing due to the requirement to use brick filtering - it is a slower process than DSD no matter what and will never be able to have as good pulse response as DSD of comparable file size. Given infinite resources, PCM would be perfect - which is to say never in practice.
  
 There is the middle ground - DXD. 
  
 I did those comparisons. The biggest difference is ALWAYS from 44.1 to whatever at least twice higher sampling frequency - the differences among those higher sampling rates and DSD are much more subtle but can be heard with good material.
  
 Audible results can vary - due to amplifiers. I have seen at least once mentioned that out of band noise can trigger a class AB amp into almost permanent class A amp or at least into allowing at  higher level to be still operated in class A mode - due to the out of band HF noise - yielding BETTER sonics. I have yet to run across such a case I could hear or measure. I have heard amps that using DSD turned into soft mush - where PCM was CLEARLY preferable. Me like it or not, but this is an honest account.
  
 It is still early in the game for DSD. Requirements for DSD256, let alone DSD512, seem today utopistic for general consumer - storage is overwhelming. Dowloads are also snail speed compared to MP3s and Redbook - which will all get better in the future. 
  
 But the potential definitely IS there.


----------



## bigshot

Now we have two of them.


----------



## wehaveyourpants

limpidglitch said:


> I think bigshot meant _in_harmonic distortion, like IMD, which does occur and can fold downwards.
> None-the-less, interesting info on why one would want to cut infra-sonic frequencies.


 
  
 This is actually something that fascinates me about music... That said:
  
 Inharmonicity is a function of naturally occurring sound in musical instruments. It's not actually a real word but a *pseudonym for* partial frequencies that make-up tone and first appears in text around 1942. It is the generation of partial harmonics on both sides of a frequency that enables the fundamental frequency to sustain and is the reason we (ambiguous) prefer real pianos to synthesized ones.
  
 Partials are my favorite thing about music, but not what he is referring to. How do I know? Because harmonic distortion is an electrical phenomenon caused by the materials and components of an amplifier and are a direct result of clipping. They are artificial and humans tend not to prefer them over the real thing specifically because they lack tone and typically only when they are, out of phase... or out of order.
  
 The real culprit is clipping in his case which is an easier form of distortion to understand. Kept in phase and under control 2nd order harmonics in the lower 2/3 are the way to go. In the studio I prefer clean/tight bass, preferably from an SM57 running through a Pultec EQA-1P and then through a Fairchild Compressor. In this way you amplify the partials surrounding the fundamental and compress the resonance and control roll-off + harmonics. This keeps the low frequencies clear and ordered so that you point out the differences.
  
 This conserves energy and makes for a more powerful recording... The best example of recorded partials I can think of is Time by Pink Floyd. The recorded clocks used in the song were originally recorded as part of a quadraphonic experiment years before. The tick-tock that leads into the song is actually Roger Waters picking bass. To recreate that you need to amplify the tone while compressing the resonance.
  
 By shaping roll-off you can predict harmonics and accurately mix the frequencies together. This keeps harmonics from one tone from stepping on the fundamental of another. The fundamentals are what's important, but without harmonics music sounds dry.
  
 Listen to anything produced by Rick Rubin and you'll instantly hear what I mean. Stadium Arcadia is a great example of masterful bass tracking and a favorite to listen to in the cans. So much detail.
  
 Harmonics and harmonic distortion are different things. One is good, while the other is only perceivable by the enlightened. Enlightenment in my case came from listening to my father record the worst sort of commercial music for thousands of hours in the studio. The primary argument of this thread however is Bit rate perception and on that subject the original poster is again using conjecture to explain the subtleties of physics.
  
 Because he cannot perceive a change MUST mean one doesn't exist. Anyone looking to KNOW the truth can complete this experiment:
  
 1. Pick a song you enjoy that has a native resolution of 24-bits, 192 kHz 
 2. Convert that song into a 16-bit version at 96 kHz
 3. Create a playlist that has the 24-bit version listed 3 times and the 16-bit version last
 4. Turn it up to the preferred level and rock out
  
 If when the fourth song plays you can't tell the difference between the two then you should call an audiologist and trade in your cans for hearing aids... You are going deaf.
  
 White noise is no substitution for tone. Partials are what make music and why Audiophiles (another word made up in the early 1940's) prefer it in it's raw form.
  
 Sensation is perception that is affected by experience. To assume that our ears, which are capable of sensing an infinite number of half-tones over a given range can't be trained to hear the difference between 24-bit & 16-bit is laughable. Now 24-bit and 32-bit... Might be more difficult. That said, detecting the difference between analog and digital playback is actually quite easy when done side by side.


----------



## Roly1650

bigshot said:


> Now we have two of them.



Yawn! Yep we sure do.


----------



## RRod

So now 24bits has to do with harmonic content? I give up.


----------



## StanD

bigshot said:


> Now we have two of them.


 
 Not to worry, they grow like mushrooms, more will pop up.


----------



## jugate

bigshot said:


> For the purposes of mixing and mastering, a lower noise floor is MUCH more useful than higher sampling rates. A lower noise floor means you have more room to boost an element in the mix without dragging noise up with it. Frequency response above the range of human hearing is notoriously spiky in a lot of electronics, and in playback inaudible spikes can cause harmonic distortion lower in the audible range. Because of this, it's standard procedure when mastering to put music through a low pass filter to get rid of inaudible frequencies. They can only hurt sound quality. They can't improve it.


 
  
  
 summarizing: 16 bits and 48 khz is the perfect combination for music but...
  
  
 why we found that dts and dolby HD formats uses 24 bits (even at 48 khz) ??


----------



## lamode

greenears said:


> I wish people would boycott DSD.


 
  
 +1,000,000


----------



## analogsurviver

lamode said:


> +1,000,000


 
 And I wish CD was never invented - or better said adopted - for pushing the overall quality back for say a quarter century.


----------



## lamode

matte82 said:


> And then there is the matter of jitter.


 
  
 ...which has been shown to be inaudible at the levels present in most digital equipment.


----------



## StanD

I always ask the jitterbugs if they can tell me me how many PPM of clock jitter manifests in which audible artefacts and what the expected measurable values should be. I never get an answer.


----------



## lamode

greenears said:


> I think if we turn @stv014 loose on this 24 vs 16 problem, with help from a few recording engineers that have access to high quality mics and 24 bit recording chain, we should be able to create a series of ABX setups & fles and post them as a challenge to all comers to get the mythical 9/10 ABX log of any equipment of their choosing.  Frankly this whole question is relatively easily solved, the puzzle is why nobody really wants to solve it


 
  
 It has been resolved ad nauseum and links to the papers have been posted in this thread many times. The 24-bit evangelists never seem to find those links though.


----------



## bigshot

jugate said:


> summarizing: 16 bits and 48 khz is the perfect combination for music but... why we found that dts and dolby HD formats uses 24 bits (even at 48 khz) ??


 
  
 Why do they put "New and Improved!" on boxes of laundry detergent when it's the same soap as ever inside the box?


----------



## bigshot

stand said:


> Not to worry, they grow like mushrooms, more will pop up.


 

 I think these are deliberate trolls who happen to be in love with the sound of their own voice. I don't make it through reading the first couple of lines without deciding to move on to greener pastures.


----------



## sonitus mirus

bigshot said:


> I think these are deliberate trolls who happen to be in love with the sound of their own voice. I don't make it through reading the first couple of lines without deciding to move on to greener pastures.


 
  
 It seems like a common marketing strategy to me.  There is always a lengthy retort for a comment that refutes any notion that goes against the general idea that spending money on their product is necessary.  Any link to a scientific study is ignored when it challenges their position, and eventually it comes around to some test that is basically impossible to conduct without bias for an overwhelming number of readers.


----------



## Krutsch

bigshot said:


> I think these are deliberate trolls who happen to be in love with the sound of their own voice. I don't make it through reading the first couple of lines without deciding to move on to greener pastures.


 

 I am starting to wonder if some of these are paid bloggers / posters. Look at the Pono thread to see what I mean. A number of new members with a low post count gushing about how great "Sticky Fingers" sounds in DSD on their Pono.
  
 I've been suspicious of some of the posts I've read from the iFi Audio evangelists and it's not hard to imagine the 24-bit marketers aren't doing something similar. I am speaking from first-hand experience from a different industry (i.e. paid promotional influencers) - it's more common than many would like to believe.
  
 Hopefully I don't get banned for the above comment...


----------



## Matte82

lamode said:


> matte82 said:
> 
> 
> > And then there is the matter of jitter.
> ...




http://www.cranesong.com/jitter_1.html

And fwiw, my default recording format is 24-bit 48khz. And i typically bounce down to 44.1k 16-bit files. The daw I use has a 64-bit float mix engine and my interface actually has a built in 80-bit float mix engine. I don't use it much, but have.


----------



## RRod

To put some money where my speaky-parts are, I went ahead and bought the first CS&N album off of teh Ponostore. It's remastered at 24/96; so far I've gotten it down to 12/32 and haven't even bothered doing an ABX because I'd really just be guessing. I assume this is the same remaster as on HDTracks; the DR numbers appear the same. SMH.


----------



## Greenears

bigshot said:


> Why do they put "New and Improved!" on boxes of laundry detergent when it's the same soap as ever inside the box?


 

 Because it's a new and improved box design.


----------



## 1c3d0g

krutsch said:


> I am starting to wonder if some of these are paid bloggers / posters. Look at the Pono thread to see what I mean. A number of new members with a low post count gushing about how great "Sticky Fingers" sounds in DSD on their Pono.
> 
> I've been suspicious of some of the posts I've read from the *iFi Audio* evangelists and it's not hard to imagine the 24-bit marketers aren't doing something similar. I am speaking from first-hand experience from a different industry (i.e. paid promotional influencers) - it's more common than many would like to believe.
> 
> Hopefully I don't get banned for the above comment...


 
 Damn right. I bought an iDSD Micro and honestly my onboard Realtek chip truly sounds much better.
	

	
	
		
		

		
		
	


	




 I don't know why, but iDSD Micro + Sony MDR-XB1000 are not an improvement over integrated laptop audio.
  
 I have a Fostex TH900 on the way. Hopefully that will open up a whole new world, but if it doesn't, this baby will go on sale.


----------



## StanD

bigshot said:


> Why do they put "New and Improved!" on boxes of laundry detergent when it's the same soap as ever inside the box?


 
  
  


greenears said:


> Because it's a new and improved box design.


 
 Sometimes (often) I feel that's what Microsoft does with Windoze, they change the number in the "About Box" and print new artwork.


----------



## analogsurviver

1c3d0g said:


> Damn right. I bought an iDSD Micro and honestly my onboard Realtek chip truly sounds much better.
> 
> 
> 
> ...


 
 Which computer has so well implemented Realtek chip inside ?


----------



## bigshot

krutsch said:


> I am starting to wonder if some of these are paid bloggers / posters.


 
  
 You might be right. But I would never attribute to commercialism that which can be attributed to just being crazy and contrary.


----------



## Don Hills

stand said:


> Not to worry, they grow like mushrooms, more will pop up.


 
  
 They don't so so well in sunlight. Keep exposing them to the bright light of truth...


----------



## limpidglitch

bigshot said:


> You might be right. But I would never attribute to commercialism that which can be attributed to just being crazy and contrary.


 
  
 How's that line… "don't attribute to malice if stupidity explains it well enough"?


----------



## 1c3d0g

analogsurviver said:


> Which computer has so well implemented Realtek chip inside ?


 
  
 Lenovo Flex2 14". And I know it's ridiculous for the onboard chip to sound better, so perhaps it's just this pairing that's not a match. I'm hoping the TH900 will change that. If not, then iFi has a whole lot of explaining to do.


----------



## analogsurviver

1c3d0g said:


> Lenovo Flex2 14". And I know it's ridiculous for the onboard chip to sound better, so perhaps it's just this pairing that's not a match. I'm hoping the TH900 will change that. If not, then iFi has a whole lot of explaining to do.


 
 What type of files are you using? PCM, DXD, DSD ? Which software/player?
  
 Do you run micro off USB power supply or off its internal battery ? Although ifi says they made whole lot to combat the crappy USB power ( noise and whatnot), make sure this is not an isssue - by at least once running micro off batteries. As you probably know, it reqiuires (reasonably charged ) micro being turned on prior connecting to the USB. What is the current rating of your USB port ? - it helps if it is rated for more ( USB 3.0 ). 
  
 Good luck with your setup!


----------



## stv014

matte82 said:


> http://www.cranesong.com/jitter_1.html


 
  
 That page looks familiar, I think it was already discussed here a while ago, and problems were found with the samples, notably analyzing them revealed much higher than the originally claimed amounts of jitter. Perhaps it was fixed since then, but those samples might not be representative of jitter in typical real world equipment. Note also that the reference sample has inverted phase relative to the processed ones; this may or may not be audible, but it should ideally be corrected in an ABX test.


----------



## audionewbi

*Rant:*
  
 I am all for competition and open market but I think in order for us to progress in this hobby a set of regulation is required. No I am not saying the artist are to be stripped from their right to produce albums that they feel is right for their mood but my focus is those companies which is trying to make some profit out of this new hires movement. I am not going to open the 16 vs 24 bit debate again as it has been done instead my hope to shine light on those companies who sale a albums which are only upsampled to be 24 bit. 

 Site like NativeDSD try to at least ensure the files they are selling is recorded natively in the format their site stands for and if they find that any album is not produce according to their site standard they are to be removed from it why can't we have the same request from those HD music sellers? Upsamling and recording natively are two different thing. I can upsample anything to 24 bit depth, it does not make it truly a HD recorded album.
  
 I think the hires is slowly damaging the music industry instead of helping it. We haven't still fully figured how to ensure redbook music is playback properly and we already moved to a standard that arguably can degrade sound instead of helping it.


----------



## stv014

1c3d0g said:


> Lenovo Flex2 14". And I know it's ridiculous for the onboard chip to sound better, so perhaps it's just this pairing that's not a match.


 
  
 Did you use the headphone output of the laptop ? That could have made a difference, since onboard audio often has rather high output impedance. Of course, there are also the usual issues like level matching. For those interested in hearing (or not) the effects of passing audio through a Realtek DAC and then recording it, this older thread has some samples from two different chips (ALC887 desktop and ALC270 laptop).


----------



## lamode

stv014 said:


> That page looks familiar, I think it was already discussed here a while ago, and problems were found with the samples, notably analyzing them revealed much higher than the originally claimed amounts of jitter. Perhaps it was fixed since then, but those samples might not be representative of jitter in typical real world equipment. Note also that the reference sample has inverted phase relative to the processed ones; this may or may not be audible, but it should ideally be corrected in an ABX test.


 
  
 The very concept of that "demonstration" is flawed.
 I posted a proper published scientific study a while back which showed at what point jitter became audible and the threshold was 1000x times higher then typical jitter levels in DACs.


----------



## StanD

lamode said:


> The very concept of that "demonstration" is flawed.
> I posted a proper published scientific study a while back which showed at what point jitter became audible and the threshold was 1000x times higher then typical jitter levels in DACs.


 
 Now you're talking I always ask the religious ones with a case of the jitters and they never answer. You have a link to the study, that would be interesting reading.


----------



## lamode

stand said:


> Now you're talking I always ask the religious ones with a case of the jitters and they never answer. You have a link to the study, that would be interesting reading.


 
  
 https://www.jstage.jst.go.jp/article/ast/26/1/26_1_50/_pdf
  
 I think I should add some of these to my signature. There are so many people still sucked in by the jitter scam, even most reviewers


----------



## StanD

lamode said:


> https://www.jstage.jst.go.jp/article/ast/26/1/26_1_50/_pdf
> 
> I think I should add some of these to my signature. There are so many people still sucked in by the jitter scam, even most reviewers


 
 Thanks, I gave it a quick look, well written. I downloaded it for further reading.
 I don't think the evangelists know what a pico second is. I think that ignorance is an enabler for the powers of imaginative hearing.


----------



## castleofargh

that's always the origin of problems. the 16vs24 wouldn't have half the debates if everybody tried to hear what -96db sounds like compared to 0db.
jitter is marvelous because of how many people repeat whatever they heard but have actually no idea what it would sound like if they heard some. no idea where it may occur, no idea that the delays can fluctuate and the consequences of each type, no idea what a nanosecond is, and of course no idea how the jitter will affect the resolution of the signal for each frequency.
I have wasted a lot of time learning about jitter, trying to get samples,understand what I could... the only positive in all that has been to show once more that affecting really high frequencies almost never matters in practice. my own hearing, the spectrogram of most songs, and the FR of most headphones had already showed me the light, but one more bullet can't hurt.


----------



## Krutsch

castleofargh said:


> that's always the origin of problems. the 16vs24 wouldn't have half the debates if everybody tried to hear what -96db sounds like compared to 0db.
> *jitter is marvelous because of how many people repeat whatever they heard but have actually no idea what it would sound like *if they heard some. no idea where it may occur, no idea that the delays can fluctuate and the consequences of each type, no idea what a nanosecond is, and of course no idea how the jitter will affect the resolution of the signal for each frequency.
> I have wasted a lot of time learning about jitter, trying to get samples,understand what I could... the only positive in all that has been to show once more that affecting really high frequencies almost never matters in practice. my own hearing, the spectrogram of most songs, and the FR of most headphones had already showed me the light, but one more bullet can't hurt.


 
  
 I think the reviewers (e,.g,. Stereophile) like jitter because it is something that is measurable, ignoring whether or not it's audible. I've noticed that many Stereophile reviews separate the purely subjectivist portion from the measured results portion; that is, the lead reviewer talks about what music he listened to and how it sounded, compared to other pieces in the category (in very subjective terms, like "more air between the instruments" stuff), followed by Atkinson presenting graphs showing what was measured (carefully ignoring the "...is it audible?" question, unless the measured values are relatively high, in which case he will comment on that - see link at the end).
  
 In the case of DACs, it's inevitably jitter, maybe along with voltage measurements from the analog outs. It *does* add an air of credibility to the subjectivist portion of the review and makes for great salesmanship (your opinion may vary).
  
 I read JA's treatise on jitter last night, which I found informative, regardless of the audible effects of jitter: http://www.stereophile.com/content/case-jitters


----------



## Matte82

lamode said:


> stand said:
> 
> 
> > Now you're talking I always ask the religious ones with a case of the jitters and they never answer. You have a link to the study, that would be interesting reading.
> ...




Cool read. Seems like a solid test. However I am coming at it this discussion from a different direction. In the page I linked to, he says that jitter was increased beyond normal circumstances to exaggerate the results. I agree that jitter is mostly imperceptible. But what happens when you start to mix 60-80 tracks that all have jitter? Not just a 2 channel final mix of a complete song. But many tracks of the individual instruments. That would be a test I would be more interested in. It very well could produce the same results. I'm not saying it won't. Just saying that's how I'm approaching this conversation. 

And like I said earlier. Unless I'm mixing a project that is desired in another format for some reason, my final mixes are always delivered in 44.1k 16-bit. IMO for the end user there is no reason for anything higher. But I do think there is benefit to recording at 24-bit. And the only reason I use 48k most of the time is due to the digital desk (allen and heath iLive) I use runs at 48k. I've done some test sessions at 96k, but don't see the benefit and do notice the extra cpu resources required for the higher sampling rate. Not to mention the disk space required.


----------



## lamode

matte82 said:


> Cool read. Seems like a solid test. However I am coming at it this discussion from a different direction. In the page I linked to, he says that jitter was increased beyond normal circumstances to exaggerate the results. I agree that jitter is mostly imperceptible. But what happens when you start to mix 60-80 tracks that all have jitter? Not just a 2 channel final mix of a complete song. But many tracks of the individual instruments. That would be a test I would be more interested in. It very well could produce the same results. I'm not saying it won't. Just saying that's how I'm approaching this conversation.
> 
> And like I said earlier. Unless I'm mixing a project that is desired in another format for some reason, my final mixes are always delivered in 44.1k 16-bit. IMO for the end user there is no reason for anything higher. But I do think there is benefit to recording at 24-bit. And the only reason I use 48k most of the time is due to the digital desk (allen and heath iLive) I use runs at 48k. I've done some test sessions at 96k, but don't see the benefit and do notice the extra cpu resources required for the higher sampling rate. Not to mention the disk space required.


 
  
 Digital mixing is completely jitter free as it all happens in the digital domain.


----------



## Matte82

But what about the jitter introduced in the a/d conversion of the tracks? That's what I'm referring to.


----------



## lamode

matte82 said:


> But what about the jitter introduced in the a/d conversion of the tracks? That's what I'm referring to.


 

 Even if each track did not use the same clock, it would not increase the amount of jitter distortion anyway.


----------



## sonitus mirus

krutsch said:


> I think the reviewers (e,.g,. Stereophile) like jitter because it is something that is measurable, ignoring whether or not it's audible. I've noticed that many Stereophile reviews separate the purely subjectivist portion from the measured results portion; that is, the lead reviewer talks about what music he listened to and how it sounded, compared to other pieces in the category (in very subjective terms, like "more air between the instruments" stuff), followed by Atkinson presenting graphs showing what was measured (carefully ignoring the "...is it audible?" question, unless the measured values are relatively high, in which case he will comment on that - see link at the end).
> 
> In the case of DACs, it's inevitably jitter, maybe along with voltage measurements from the analog outs. It *does* add an air of credibility to the subjectivist portion of the review and makes for great salesmanship (your opinion may vary).
> 
> I read JA's treatise on jitter last night, which I found informative, regardless of the audible effects of jitter: http://www.stereophile.com/content/case-jitters


 
  
 It is not only reviewers, but manufacturers often cling to jitter as a means of attempting to show that their products are superior.  Jitter is measurable and relatively simple to provide comparisons that even a layman such as myself can interpret.  It is easy to show that something measures better with regards to jitter, and muddling with or omitting the context of audibility only serves to promote their goals.


----------



## U2Bono269

I've been reading as many of the 16bit vs 24bit threads I could find around here, because I'm interested in high resolution audio and whether or not it's worthwhile. I do have 2 albums in 24/96, though I don't yet have the equipment needed for them. I figured this would be the best thread to ask this in.
  
 I've seen the argument that there's no difference in practice between a 16bit and 24bit file, assuming the same mastering. A common argument seems to be that those who do hear differences between 16bit and 24bit, and between lossless and 320mp3, are hearing this difference due to mastering and other factors. This makes a lot of sense and I understand it.
  
 However, doesn't the fact that a high-resolution file may use different mastering justify it's existence? It would seem to me that if a high-resolution file offers a different master, it's automatically a worthwhile endeavor. Am I missing something, or oversimplifying?


----------



## Matte82

u2bono269 said:


> I've been reading as many of the 16bit vs 24bit threads I could find around here, because I'm interested in high resolution audio and whether or not it's worthwhile. I do have 2 albums in 24/96, though I don't yet have the equipment needed for them. I figured this would be the best thread to ask this in.
> 
> I've seen the argument that there's no difference in practice between a 16bit and 24bit file, assuming the same mastering. A common argument seems to be that those who do hear differences between 16bit and 24bit, and between lossless and 320mp3, are hearing this difference due to mastering and other factors. This makes a lot of sense and I understand it.
> 
> However, doesn't the fact that a high-resolution file may use different mastering justify it's existence? It would seem to me that if a high-resolution file offers a different master, it's automatically a worthwhile endeavor. Am I missing something, or oversimplifying?




So far as I know there is no reason to master differently for different formats. As long as we're talking audio only. There are finally some pretty strict loudness guidelines emerging in the video world. So I guess maybe a "made for Blu-ray" concert video might drop the average level a few dB and have a bit more headroom and punch, compared to a the CD version. I'm still don't see how it would justify needing to be 24-bit.


----------



## U2Bono269

Then I'm confused. I've seen similar sentiments in numerous threads on this topic: that perceived differences between mp3/lossless or 16/24 are due to comparing different masters. Logically, it doesn't seem likely that an album would be made into mp3, redbook, and 24bit versions without some sort of adjustment to take the format into account, right?


----------



## Krutsch

u2bono269 said:


> I've been reading as many of the 16bit vs 24bit threads I could find around here, because I'm interested in high resolution audio and whether or not it's worthwhile. I do have 2 albums in 24/96, though I don't yet have the equipment needed for them. I figured this would be the best thread to ask this in.
> 
> I've seen the argument that there's no difference in practice between a 16bit and 24bit file, assuming the same mastering. A common argument seems to be that those who do hear differences between 16bit and 24bit, and between lossless and 320mp3, are hearing this difference due to mastering and other factors. This makes a lot of sense and I understand it.
> 
> However, doesn't the fact that a high-resolution file may use different mastering justify it's existence? *It would seem to me that if a high-resolution file offers a different master, it's automatically a worthwhile endeavor. Am I missing something, or oversimplifying?*


 
  
 I download from time-to-time from HDTracks when I read reports of a particularly good mastering effort, or I want a lossless copy and am too impatient to wait for the CD from Amazon. Then I use Audiofile Engineering's Sample Manager w/ Izotope 64 to down-sample and down-convert to 44.1 / 16 or 48 / 16 (whichever is a power of two reduction from the original). I do this so my music files are compatible with my Android phone and my Sonos system.
  
 Yes, I do keep the originals and, yes, I do play them back on my laptop when I listen to music from there - why not. But, from my reading of the math/audio engineering involved, I am completely convinced I can't tell the difference between high-res audio and Redbook, even though I've never bothered with a formal ABX (anecdotal experiments produce similar results for me - guesswork).


----------



## RRod

u2bono269 said:


> Then I'm confused. I've seen similar sentiments in numerous threads on this topic: that perceived differences between mp3/lossless or 16/24 are due to comparing different masters. Logically, it doesn't seem likely that an album would be made into mp3, redbook, and 24bit versions without some sort of adjustment to take the format into account, right?


 
  
 There doesn't need to be an adjustment other than down-converting to 16/44.1 and then doing the compression of your choosing. Unless the difference between the 24bit and 16bit files is audible, what are you losing? The nature of the source material itself, along with the mixing/mastering, will determine how low you can go before hearing differences. As I pointed out recently, the 24/96 CSN album I just got off Pono is, to my ears, audibly indistinguishable from a down-conversion to 12/32 (yes that's 12 bits, 32k samples/s) in ABX testing. A look at the spectrograms tells us why: high background noise and significant roll-off in the treble above 15k. One might ask exactly how good this "mastering" is, considering the CD of the same album by Audio Fidelity scores better on the DR meter.


----------



## U2Bono269

But I'm not talking about simply downgrading a 24 bit file. I haven't done so myself, but I imagine the two would be indistinguishable. You mention Audio Fidelity as being a better master of the same album. Isn't the point of stores like Pono/MFSL/Audio Fidelity to provide different masters? If I bought a Pono-mastered copy of Harvest, for example, might it sound better than my FLAC CD rip because it's mastered to sound better? Assuming that's true, I would compare my 16bit CD version to my 24bit Pono version and conclude that the 24bit version sounds better. So in this hypothetical situation, the 24bit copy is superior sounding, but not because of its sampling rate or bit depth.
  
 I'm just reading all of these testimonials and mind-boggling numbers and wondering if I'm asking the right question when it comes to "hi-res" audio. A prevalent argument is that 24bit audio is indistinguishable from 16bit on a basis of pure blind comparison. I'm not disputing that. I tend to agree with it. This would lead to the conclusion that it is indeed a waste of money. But if these 24bit albums are engineered to sound better regardless of their specs, and just happen to come in 24/96 or whatever audio, then it would seem to me that it's money well spent if you're getting a better sounding product. Of course, there are some 24bit albums that are not superior sounding, and that's probably normal.
  
 FWIW, the 2 HDtracks albums I bought were because I don't buy CDs anymore and I wanted lossless files. So I bought them and downsampled to 16/44 for my ipod. I still have the originals. I'm not sure if I'll ever be able to listen to them, and if I do, I'm not sure if they'll sound better to my ears.
  
 I have done ABX testing with mp3 and FLAC files. I've found I can't reliably distinguish between the two when it comes to a lot modern records, like the Foo Fighters, for example. But I could do a pretty good job with older recordings.


----------



## RRod

u2bono269 said:


> But I'm not talking about simply downgrading a 24 bit file. I haven't done so myself, but I imagine the two would be indistinguishable. You mention Audio Fidelity as being a better master of the same album. Isn't the point of stores like Pono/MFSL/Audio Fidelity to provide different masters? If I bought a Pono-mastered copy of Harvest, for example, might it sound better than my FLAC CD rip because it's mastered to sound better? Assuming that's true, I would compare my 16bit CD version to my 24bit Pono version and conclude that the 24bit version sounds better. So in this hypothetical situation, the 24bit copy is superior sounding, but not because of its sampling rate or bit depth.
> 
> I'm just reading all of these testimonials and mind-boggling numbers and wondering if I'm asking the right question when it comes to "hi-res" audio. A prevalent argument is that 24bit audio is indistinguishable from 16bit on a basis of pure blind comparison. I'm not disputing that. I tend to agree with it. This would lead to the conclusion that it is indeed a waste of money. But if these 24bit albums are engineered to sound better regardless of their specs, and just happen to come in 24/96 or whatever audio, then it would seem to me that it's money well spent if you're getting a better sounding product. Of course, there are some 24bit albums that are not superior sounding, and that's probably normal.
> 
> ...


 
  
 The issue with mastering is one of information. Take my Pono example again: all it says is "remastered", and gives no info on exactly which remastering it is. Judging from the DR measurements, it's the same as the one on HDTracks, which does give info on the mastering (and in fact has a rational for the lower DR measurements in one of the reviews). But there are other HDTrack albums with not much info on the actual mastering.
  
 Here's what a "long-time engineer" for CS&N says about the HD release:
 "_Re-discovered is what I have to say happened that evening. The musical thrills CSN pulls off in these three releases along with the true beauty revealed in the HD files is nothing short of "clearly" wonderful to say the least. Please turn down the lights, sit back and enjoy these albums all over again!"_ 
  
Note the language doesn't immediately imply that HD is the reason for the "true beauty", but I would think many people would take it that way. So then the next question is, how much of the $18 price tag is for the new mix, and how much is for the moar bits and samples that aren't needed on the listener's end?


----------



## jugate

bigshot said:


> Why do they put "New and Improved!" on boxes of laundry detergent when it's the same soap as ever inside the box?


 
  
 ok, marketing. I see.
  
 But if i have a z906 (logitech) that dont have decoder for DD/DTS HD and play a movie with that codec audio... the system sound exactly that if i have conected the analog imputs and decoding with software?


----------



## Greenears

u2bono269 said:


> I've been reading as many of the 16bit vs 24bit threads I could find around here, because I'm interested in high resolution audio and whether or not it's worthwhile. I do have 2 albums in 24/96, though I don't yet have the equipment needed for them. I figured this would be the best thread to ask this in.
> 
> I've seen the argument that there's no difference in practice between a 16bit and 24bit file, assuming the same mastering. A common argument seems to be that those who do hear differences between 16bit and 24bit, and between lossless and 320mp3, are hearing this difference due to mastering and other factors. This makes a lot of sense and I understand it.
> 
> However, doesn't the fact that a high-resolution file may use different mastering justify it's existence? It would seem to me that if a high-resolution file offers a different master, it's automatically a worthwhile endeavor. Am I missing something, or oversimplifying?


 
  
 >A common argument seems to be that those who do hear differences between 16bit and 24bit, and between lossless and 320mp3, are hearing this difference due to mastering and other factors
  
 It seems you have missed a key point of this thread.  I'm sure it's back there 50 or 100 pages back (insert fading maniacal laugh)...
  
 There is not much agreed on this thread, but it's safe to say (more or less) that most agree (or at least don't dispute) that this can be easily controlled.  All the ABX testers convert the 24 bit files to 16 bit themselves to completely control for any mastering differences.  You can do it in Sox or @stv014's wonderful quantize.c/exe program.  Even more fun is converting 24 to 14 or 12 bit.  Try it.
  
 We know and accept there are mastering differences, but generally I think if 16 and 24 are offered side to side on the same site, I think it's the same mastering.  At least I've never heard anyone claim contrary.  The threads where mastering came up the 16 and 24 were obtained from different places.  One poster challenged me with both files, so I ABX'd 24, 16 and 24 converted to 16.  Couldn't tell the 3 apart, although that doesn't prove the 16 and 24216 were identical, just inaudibly different on my system.


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## Roly1650

greenears said:


> >A common argument seems to be that those who do hear differences between 16bit and 24bit, and between lossless and 320mp3, are hearing this difference due to mastering and other factors
> 
> It seems you have missed a key point of this thread.  I'm sure it's back there 50 or 100 pages back (insert fading maniacal laugh)...
> 
> ...



It's easier and faster to just do a null or difference extraction test. Tells you all you need to know about the difference between two files, sample rates, bit depths etc., several ways to do it and the softwares free. ABX is tedious and time consuming and might I venture, unnecessary when digital files are involved. For example, I used it to determine there was zero difference, (audible or not), between a supposed 24/96 file I downloaded from HDTracks and the RBCD I already had.


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## Greenears

u2bono269 said:


> ....
> 
> I have done ABX testing with mp3 and FLAC files. I've found I can't reliably distinguish between the two when it comes to a lot modern records, like the Foo Fighters, for example. But I could do a pretty good job with older recordings.


 
  
 Try the ABX again with MP3 320 LAME encoded vs FLAC. Seems to be consensus this can't pass ABX.  The older recordings may have been encoded with poor MP3 encoders.


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## bigshot

Try AAC. That is even a little better than LAME.


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## bigshot

Just take the HD tracks and Pono songs and bump them down to AAC 256 VBR or LAME 320. They'll sound exactly the same and take up a lot less space. You're welcome!


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## U2Bono269

I want lossless audio. Whether or not it's "necessary" over mp3 or aac or whatever is irrelevant because it's what I want. I tend to think that a file that has more information in it is more accurate than one with less. I don't care if I can hear it or not. That is my preference.
  
 I'm also really not interested in endless ABX testing and fretting over technical information. I want to listen to the music. So anytime a user here quotes test results, or graphs, or frequencies, or any of a billion other things, it misses the point for me.
  
 And "controlling for master differences" is simply not the point of my question. I want to find the best sounding masters/transfers/versions (whatever you want to call it) of my favorite albums, regardless of whether they're ripped in FLAC off a CD or downloaded from a hi-res website. But it seems that there's not really an answer to my question.


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## OddE

u2bono269 said:


> And "controlling for master differences" is simply not the point of my question. I want to find the best sounding masters/transfers/versions (whatever you want to call it) of my favorite albums, regardless of whether they're ripped in FLAC off a CD or downloaded from a hi-res website. But it seems that there's not really an answer to my question.


 
  
 -In that case, I believe the best course of action is simply to purchase several different masters and pick the one you like best. With time and experience, hopefully you will find that some sites in general offer masters which are more to your liking than others, and will be able to focus your efforts?
  
 After all, once we start discussing other aspects than pure sound quality (which can be objectively quantified), we're heading off into subjective opinion territory, and YMMV.


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## U2Bono269

Yes, I think that would have to be my approach. So much of the 16vs24 debate seems to focus on the sound science part of it, while purposefully discounting mastering differences. It seems that the mastering variations are perhaps not as prevalent or obvious as I had thought based on my reading. I guess it really does come down to trial and error.


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## RRod

u2bono269 said:


> Yes, I think that would have to be my approach. So much of the 16vs24 debate seems to focus on the sound science part of it, while purposefully discounting mastering differences. It seems that the mastering variations are perhaps not as prevalent or obvious as I had thought based on my reading. I guess it really does come down to trial and error.


 
  
 That's the focus because a) it's the sound science forum and b) there's no reason for Redbook releases to have any worse mastering than their hi-res source material. That they often do is a consequence of things like the loudness war that have nothing to do with the format.


----------



## OddE

u2bono269 said:


> Yes, I think that would have to be my approach. So much of the 16vs24 debate seems to focus on the sound science part of it, while purposefully discounting mastering differences. It seems that the mastering variations are perhaps not as prevalent or obvious as I had thought based on my reading. I guess it really does come down to trial and error.


 
  
 -That is because those are two completely different debates; the 16/24 debate - at least here in Sound Science - tends to focus on what either format is capable of and how that capability relates to the capabilities of the most important transducer of all - our ears.
  
 The music being distributed using the various formats, on the other hand, is a different issue altogether; if some publisher decides to release an album with, say, 40dB of dynamic range over a -55dBFS noise floor in 24 bits, while also sampling at 96 or 192kHz source material with no content whatsoever above 15kHz - that is an inherent problem with the publisher, not the format.
  
 Edit: Ninja'd by RRod; that's what you get for loading the thread, reading at a later time and then eventually getting around to writing an answer...


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## U2Bono269

Now that makes sense.
  
 And sorry, I didn't realize this was the Sound Science forum. I'm using Tapatalk and it's not very organized. I posted it in this thread because it had the most information. But for the most part, every other thread I read made the same claims, for the most part.


----------



## lamode

u2bono269 said:


> I've been reading as many of the 16bit vs 24bit threads I could find around here, because I'm interested in high resolution audio and whether or not it's worthwhile. I do have 2 albums in 24/96, though I don't yet have the equipment needed for them. I figured this would be the best thread to ask this in.
> 
> I've seen the argument that there's no difference in practice between a 16bit and 24bit file, assuming the same mastering. A common argument seems to be that those who do hear differences between 16bit and 24bit, and between lossless and 320mp3, are hearing this difference due to mastering and other factors. This makes a lot of sense and I understand it.
> 
> However, doesn't the fact that a high-resolution file may use different mastering justify it's existence? It would seem to me that if a high-resolution file offers a different master, it's automatically a worthwhile endeavor. Am I missing something, or oversimplifying?


 
  
 There is no technical reason to do it, but the audience for HD material is more discerning so perhaps they remaster some releases with less compression to keep audiophiles happy.


----------



## jugate

jugate said:


> ok, marketing. I see.
> 
> But if i have a z906 (logitech) that dont have decoder for DD/DTS HD and play a movie with that codec audio... the system sound exactly that if i have conected the analog imputs and decoding with software?


 
  
 somebody?


----------



## RRod

jugate said:


> somebody?


 
  
 How exactly do you have things set up? The HD surround formats have the ability to send out a non-HD "core" when connected with something like optical cable that can't handle the full bandwidth, but this all depends on the exact settings of your source and sound card.


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## bigshot

Blu-ray disks have multiple audio tracks in various formats. Generally, you set your player to default to the highest quality track your player is able to decode and don't worry about it any more. The only really significant difference in sound quality is 2 channel vs 5.1/7.1.


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## Greenears

u2bono269 said:


> I want lossless audio. Whether or not it's "necessary" over mp3 or aac or whatever is irrelevant because it's what I want. I tend to think that a file that has more information in it is more accurate than one with less. I don't care if I can hear it or not. That is my preference.
> 
> I'm also really not interested in endless ABX testing and fretting over technical information. I want to listen to the music. So anytime a user here quotes test results, or graphs, or frequencies, or any of a billion other things, it misses the point for me.
> 
> And "controlling for master differences" is simply not the point of my question. I want to find the best sounding masters/transfers/versions (whatever you want to call it) of my favorite albums, regardless of whether they're ripped in FLAC off a CD or downloaded from a hi-res website. *But it seems that there's not really an answer to my question.*


 
  
 Actually there might be an answer.  There seems to be some sites that run Foobar's DR (dynamic range) plugin on all the tracks in an album and post logs.  Now of course beware, one DR reading vs another does not guarantee quality.  But it does sometimes shed light on which "mastering" is downloadable from which source.  The nice thing is you can do this at home and try to match up to their database.
  
 This doesn't have much to do with 16 vs 24 format, they both yield the same results if they are derived from the same mastering.  I've seen it go both ways - I looked at a circa 2000 remaster of Dire Straights (Communique if I remember) in 24 bits on Pono store and I was warned in forums that that is a notoriously bad remaster (in the original 16 same in 24).  I was advised to stick with the original CD as the nicest sound.  Then I looked at AC/DC 2014 release (ok hardly audiophile haha) and was advised that in Europe it was mastered with DR8 and US DR6 (logs are posted).  Apparentlly the HD tracks 24 bit is the DR8 version and _may_ be the only legit place in US you can get that version (other than importing a plastic CD which is too much trouble). 
  
 Again examples chosen to show nothing to do with the 24b vs 16b debate - but a fun sidetrack.


----------



## bigshot

There is a contradiction between saying "I want lossless because it has more data and I don't care if compressed audio sounds exactly the same" and then saying "I don't want to be fretting over technical information, I just want to listen to music". If it sounds perfect to your ears, it *is* perfect. I agree that you shouldn't worry about technical information. All you should worry about is what your ears hear. MP3 LAME and AAC are capable of complete transparency. Just listen to the music and don't worry about file sizes.


----------



## dprimary

u2bono269 said:


> Then I'm confused. I've seen similar sentiments in numerous threads on this topic: that perceived differences between mp3/lossless or 16/24 are due to comparing different masters. Logically, it doesn't seem likely that an album would be made into mp3, redbook, and 24bit versions without some sort of adjustment to take the format into account, right?


 
 A MP3 or AAC release would have different masters, you make minor adjustments to maximize the quality of the final format. For example master for iTunes means the mastering engineer produced a separate master just for AAC encoding most likely listening to the original master through and encoder and processing it to sound as good as they can get it. So a mastered for iTunes track should not be exactly the same as the CD track. 
  
 A 16bit release and 24 bit release should have the same 24 bit master, when you dither a 24bit master down to 16 all that you are essentially doing is raising the noise floor. It think it was someone at Harman that said 24bit gives you foot room. 16 bit has a noise level -96dB while 24bit give about 24dB more so the noise is roughly -120dB down, which is limited by physics to get much better. An extremely silent studio will be about 20dB SPL you would playing the music at an ear splitting116dB SPL before you to start to hear the noise floor of the 16bit file. All you do for a 16bit master is dither down the 24bit master.


----------



## Don Hills

dprimary said:


> ... All you do for a 16bit master is dither down the 24bit master.


 
  
 I wish that's all they did.


----------



## dprimary

don hills said:


> I wish that's all they did.


 
 Unfortunately they will "remaster" old albums to sell "better high rez" (AKA louder and brighter) sound.


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## bigshot

dprimary said:


> A MP3 or AAC release would have different masters, you make minor adjustments to maximize the quality of the final format.


 
  
 Not if you roll your own. You end up with something that is indistinguishable from the original format.


----------



## analogsurviver

1c3d0g said:


> Damn right. I bought an iDSD Micro and honestly my onboard Realtek chip truly sounds much better.
> 
> 
> 
> ...


 
 Did you manage to get micro sounding better than the Realtek onboard chip ?
  
 I had quite "a-not-so-pleasent-ball-with-my-PC" before my nano started to sound as it should. PC related, registry leaning etc etc. And it takes a few days to configure whatever software you are using to your liking/preference - ALWAYS taking in account of the limitations of your hardware.
  
 It is only fair to any DAC to clearly state with which hardware and software ( and how configurd ...) it is being used - it can do nothing on its own.


----------



## Greenears

dprimary said:


> A MP3 or AAC release would have different masters, you make minor adjustments to maximize the quality of the final format. For example master for iTunes means ....


 
  
 I don't agree with that.  There s plenty of science on this thread that shows modern MP3 320 or AAC 256 the differences to lossless are inaudible so no need to remaster anything due to format.  The reason they are doing this push to remaster is to optimize for typical listening conditions of iTunes users (which I have to assume is mobile earbuds).  Which sucks if you are not typical.
  
 It would be nice if they start releasing with multiple masters for different conditions, put it as parameters in the EQ.


----------



## U2Bono269

greenears said:


> Actually there might be an answer.  There seems to be some sites that run Foobar's DR (dynamic range) plugin on all the tracks in an album and post logs.  Now of course beware, one DR reading vs another does not guarantee quality.  But it does sometimes shed light on which "mastering" is downloadable from which source.  The nice thing is you can do this at home and try to match up to their database.
> 
> This doesn't have much to do with 16 vs 24 format, they both yield the same results if they are derived from the same mastering.  I've seen it go both ways - I looked at a circa 2000 remaster of Dire Straights (Communique if I remember) in 24 bits on Pono store and I was warned in forums that that is a notoriously bad remaster (in the original 16 same in 24).  I was advised to stick with the original CD as the nicest sound.  Then I looked at AC/DC 2014 release (ok hardly audiophile haha) and was advised that in Europe it was mastered with DR8 and US DR6 (logs are posted).  Apparentlly the HD tracks 24 bit is the DR8 version and _may_ be the only legit place in US you can get that version (other than importing a plastic CD which is too much trouble).
> 
> Again examples chosen to show nothing to do with the 24b vs 16b debate - but a fun sidetrack.


 
  
 Thanks! That's kind of what I was looking to know, and is very helpful to me.
  
  


bigshot said:


> There is a contradiction between saying "I want lossless because it has more data and I don't care if compressed audio sounds exactly the same" and then saying "I don't want to be fretting over technical information, I just want to listen to music". If it sounds perfect to your ears, it *is* perfect. I agree that you shouldn't worry about technical information. All you should worry about is what your ears hear. MP3 LAME and AAC are capable of complete transparency. Just listen to the music and don't worry about file sizes.


 
  
 I think that lossless does sound better to me than compressed audio, so there's no contradiction there. My question was related to 24bit audio specifically, and how much of it is related to mastering or the conversion process. I'm still learning all of this stuff, so I'm probably not as eloquent as I'd like to be, so sorry if there's any confusion.
  
 I guess I was just wondering if what you guys do, i.e., making your own exact comparisons, is the same thing they do in a professional setting. Enough people claim to hear a difference that it makes me think that looking at it from a strictly scientific point of view is very limiting. After all, there is little-to-no transparency from labels or engineers, so how do we know for sure?
  
 This has been good though  Lots to think about and learn about.


----------



## Greenears

u2bono269 said:


> Enough people claim to hear a difference that it makes me think that looking at it from a strictly scientific point of view is very limiting. After all, there is little-to-no transparency from labels or engineers, so how do we know for sure?


 
  
 ABX testing in your future I see, young padawan.
  
 The reality is that only 0.1% of people claim to hear differences in anything the rest are oblivious.  The 0.1% are just over-represented on forums, self selecting.  But the interesting part is even those that claim a difference can't pass an ABX test.  When that event happens you get two reactions:  Disbelief in the test, or realization how strong the power of suggestion is.  Suggestion and imitation is very strong in primates, there is a reason the saying is "monkey see monkey do".  It provides many advantages for group survival but it does lead to people blindly buying 24 bit files because "they are better".
  
 Anyhow you don't have to believe me - ABX plugin with foobar is very easy it takes 5 minutes to set up.  Just make sure you convert your own file from CD redbook so you know there were no differences in the source.  Try out uncompressed vs MP3 192 it will open your mind.


----------



## U2Bono269

greenears said:


> ABX testing in your future I see, young padawan.
> 
> The reality is that only 0.1% of people claim to hear differences in anything the rest are oblivious.  The 0.1% are just over-represented on forums, self selecting.  But the interesting part is even those that claim a difference can't pass an ABX test.  When that event happens you get two reactions:  Disbelief in the test, or realization how strong the power of suggestion is.  Suggestion and imitation is very strong in primates, there is a reason the saying is "monkey see monkey do".  It provides many advantages for group survival but it does lead to people blindly buying 24 bit files because "they are better".
> 
> Anyhow you don't have to believe me - ABX plugin with foobar is very easy it takes 5 minutes to set up.  Just make sure you convert your own file from CD redbook so you know there were no differences in the source.  Try out uncompressed vs MP3 192 it will open your mind.


 
  
 Wow, this is getting kafkaesque. I've addressed all of those things in previous posts. Now, I know that I'm struggling to express myself due to lack of experience, but we seem to be speaking different languages. I came to this thread because it was the most active and it was reading this thread that sparked my interest. But I'm clearly in the wrong place.


----------



## bigshot

u2bono269 said:


> I think that lossless does sound better to me than compressed audio
> I guess I was just wondering if what you guys do, i.e., making your own exact comparisons, is the same thing they do in a professional setting.


 
  
 Yes. I do line level matched, direct A/B switched comparisons. I've done this comparing lossless and compressed audio and I've determined that there is a bitrate where lossy becomes audibly identical to lossless. If you are interested in finding out for yourself, I have put together a listening test of three different compression codecs at three different bitrates along with the lossless original. I'd be happy to send it to you. You listen and send me back a ranking of the ten samples. Then I'll let you know which sample was which. I bet you can't tell a difference in any of them. I've handed this test out to dozens of people and no one has been able to yet.


----------



## U2Bono269

bigshot said:


> Yes. I do line level matched, direct A/B switched comparisons. I've done this comparing lossless and compressed audio and I've determined that there is a bitrate where lossy becomes audibly identical to lossless. If you are interested in finding out for yourself, I have put together a listening test of three different compression codecs at three different bitrates along with the lossless original. I'd be happy to send it to you. You listen and send me back a ranking of the ten samples. Then I'll let you know which sample was which. I bet you can't tell a difference in any of them. I've handed this test out to dozens of people and no one has been able to yet.


 

 Thanks, but no. I have no interest in downsampling my music. I will keep my FLACs and ALACs...hard drive space is dirt cheap. Please stop bludgeoning me with this. Even if the difference I hear is purely a placebo, it increases my enjoyment. Like I said, my question was about mastering and/or processing differences in creating the files we see in places like Pono and HDTracks, and the relation to the bit depth.
  
 The takeaway for me is that if I know a 24bit hi-res version of an album is a different master, or in some way processed differently (difficult to determine, I know), then it's a worthwhile purchase for me. If it's the same as the CD version I have or could buy, it is not worthwhile.


----------



## RRod

u2bono269 said:


> I think that lossless does sound better to me than compressed audio, so there's no contradiction there. My question was related to 24bit audio specifically, and how much of it is related to mastering or the conversion process. I'm still learning all of this stuff, so I'm probably not as eloquent as I'd like to be, so sorry if there's any confusion.
> 
> I guess I was just wondering if what you guys do, i.e., making your own exact comparisons, is the same thing they do in a professional setting. Enough people claim to hear a difference that it makes me think that looking at it from a strictly scientific point of view is very limiting. After all, there is little-to-no transparency from labels or engineers, so how do we know for sure?
> 
> This has been good though  Lots to think about and learn about.


 
  
 The issue is that people who claim to "hear a difference" never do any test blind, which is the first rule in being objective about things involving human choice. If I just bought a $2000 amp, you bet my mind will want it to sound better than my $99 entry-level Magni. But that isn't necessarily the truth as far as my ears are concerned. And so it is with 24bit recordings. There is no track I have found that shows either analytical or audible proof that 24bits is superior to 16bits for playback. And it only takes one track… just one. The pro-24bit-delivery crowd need only find one music track where 24bits delivers a perceptible and beneficial* difference in sound quality at the user end; where is it?
  
 *I say beneficial because it is trivial to devise a music file that uses 24bits, but if I have to adjust the volume while I'm listening to it then it's not a beneficial experience.


----------



## bigshot

u2bono269 said:


> Even if the difference I hear is purely a placebo, it increases my enjoyment. Like I said, my question was about mastering and/or processing differences in creating the files we see in places like Pono and HDTracks, and the relation to the bit depth.


 
  
 Well, if the difference between compressed and lossless is placebo, you can bet the difference between lossless and high bitrate/sampling rate file is going to be placebo too. Transparent is transparent.
  
 When it comes to the quality of mastering, that has absolutely no relationship to the format. Some LPs sound better than CDs, some stuff at the iTunes store in AAC sounds better than at HD tracks. Some SACDs sound downright lousy. Mastering is entirely separate from the format. It all depends on the taste and experience of the engineer doing the mastering. "Remastered" doesn't necessarily mean "better". Sometimes music is poorly remastered and sounded better in an older format. To find that out, you need to talk to music collectors, not audiophiles.


----------



## U2Bono269

rrod said:


> The issue is that people who claim to "hear a difference" never do any test blind, which is the first rule in being objective about things involving human choice. If I just bought a $2000 amp, you bet my mind will want it to sound better than my $99 entry-level Magni. But that isn't necessarily the truth as far as my ears are concerned. And so it is with 24bit recordings. There is no track I have found that shows either analytical or audible proof that 24bits is superior to 16bits for playback. And it only takes one track… just one. The pro-24bit-delivery crowd need only find one music track where 24bits delivers a perceptible and beneficial* difference in sound quality at the user end; where is it?
> 
> *I say beneficial because it is trivial to devise a music file that uses 24bits, but if I have to adjust the volume while I'm listening to it then it's not a beneficial experience.


 
  
 And if you've read my previous posts here, I haven't disagreed with that sentiment. I've also stated that I have no capability at the moment to listen to 24bit tracks. I DO believe that in a strict, perfect comparison, 24bit and 16bit are indistinguishable. I have come to this conclusion based on what I've read in this thread and others, and also using what I know of human hearing and the creation of the Redbook standard. You will get no argument from me on that, though I'm still interested to listen for myself, of course. Like I said, I have 2 HDTracks albums...I bought U2's Achtung Baby album because I'm a U2-collector, and I have Damien Rice's My Favourite Fantasy because I didn't want the CD. I also recently bought a used iBasso DX50 as an upgrade to my iPod. So I will listen to those tracks when I receive it.
  
 Here's the scenario that I'm thinking of when I ask these questions...
  
 I know that My Favourite Fantasy will sound the same as my 16bit version, because I downsampled it from the 24bit. Again, no argument from me on that one. But I'm not 100% sure what's in store for me with Achtung Baby. I have the CD version of the same Achtung Baby release (the Deluxe Edition) to compare it to. Based on what I've read, I would assume they come from the same master.
  
 So, assuming they're the same master, if I listen to, say, Zoo Station from the CD FLAC, and the Zoo Station from the HDTracks FLAC, they will sound the same, right? I expect them to sound the same. I didn't buy those albums thinking "oh man, this is going to sound so much better!"
  
 But what if my expectations are wrong? What if I listen to the HDTracks version and I think it sounds different? Scientifically, what might account for that?


----------



## bigshot

u2bono269 said:


> So, assuming they're the same master, if I listen to, say, Zoo Station from the CD FLAC, and the Zoo Station from the HDTracks FLAC, they will sound the same, right?


 
  
 Well, if they are the same mastering, they will have the exact same sound quality to human ears, but whether you hear them as being the same depends on things like expectation bias and placebo.


----------



## U2Bono269

bigshot said:


> Well, if they are the same mastering, they will have the exact same sound quality to human ears, but whether you hear them as being the same depends on things like expectation bias and placebo.


 
  
 My expectation bias is that they would sound the same. However, isn't it true that for proper ABX testing, the 16bit file must be made from 24bit file? Does the fact that they are each unique spoil the results?


----------



## Greenears

u2bono269 said:


> Wow, this is getting kafkaesque. I've addressed all of those things in previous posts. Now, I know that I'm struggling to express myself due to lack of experience, but we seem to be speaking different languages. I came to this thread because it was the most active and it was reading this thread that sparked my interest. But I'm clearly in the wrong place.


 
  
 Why would a simple program you run on windows that takes less than 2 minutes to install fully with broadband be "Kafkaesque"? You were the one that said you cold hear the difference between lossy and lossless, so why not just give it a whirl instead of spending so much time reading and writing posts on forums?  You can tell us you don't like it after you tried it.  I know I know you're part of the "I don't need to" crowd.  Then what's your question exactly?


----------



## RRod

u2bono269 said:


> And if you've read my previous posts here, I haven't disagreed with that sentiment. I've also stated that I have no capability at the moment to listen to 24bit tracks. I DO believe that in a strict, perfect comparison, 24bit and 16bit are indistinguishable. I have come to this conclusion based on what I've read in this thread and others, and also using what I know of human hearing and the creation of the Redbook standard. You will get no argument from me on that, though I'm still interested to listen for myself, of course. Like I said, I have 2 HDTracks albums...I bought U2's Achtung Baby album because I'm a U2-collector, and I have Damien Rice's My Favourite Fantasy because I didn't want the CD. I also recently bought a used iBasso DX50 as an upgrade to my iPod. So I will listen to those tracks when I receive it.
> 
> Here's the scenario that I'm thinking of when I ask these questions...
> 
> ...


 
  
 Well then you get into the business of comparing masters. The good first thing to try is to put the two versions into something like Audio DiffMaker and see what it comes up with for a difference. This will let you eek out things like differences equalization and channel balances. For instance, here's the spectrogram of the difference file that I get from bass-boosting one of my tracks:

  
 If I listen to this file, I indeed hear a bass-heavy version of the track.


----------



## U2Bono269

greenears said:


> Why would a simple program you run on windows that takes less than 2 minutes to install fully with broadband be "Kafkaesque"? You were the one that said you cold hear the difference between lossy and lossless, so why not just give it a whirl instead of spending so much time reading and writing posts on forums?  You can tell us you don't like it after you tried it.  I know I know you're part of the "I don't need to" crowd.  Then what's your question exactly?


 
  
 Because I stated in a previous post that I have done ABX testing between mp3s and FLACs, yet multiple posters keep suggesting I do it as if I never said it to begin with...


----------



## bigshot

If you get the 16 and 24 from two different sources, you can't be sure the mastering is the same. That's why you make the 16 from the 24 yourself to test. You want to remove irrelevant variables and focus on what you are testing for. If you are testing just for the difference between 16 and 24 bit, you just want to test for that and not have other factors muddy the waters.


----------



## bigshot

u2bono269 said:


> Because I stated in a previous post that I have done ABX testing between mp3s and FLACs, yet multiple posters keep suggesting I do it as if I never said it to begin with...


 

 What codec? What bitrate? Did you make the compressed files yourself or did you compare a CD to an iTunes download? Did you line level match? Did you have direct A/B switching? Did you compare the samples blind? You have to test carefully to get accurate results.


----------



## U2Bono269

rrod said:


> Well then you get into the business of comparing masters. The good first thing to try is to put the two versions into something like Audio DiffMaker and see what it comes up with for a difference. This will let you eek out things like differences equalization and channel balances. For instance, here's the spectrogram of the difference file that I get from bass-boosting one of my tracks:
> 
> 
> If I listen to this file, I indeed hear a bass-heavy version of the track.


 
  
 I get that, which brings me back to my original (hypothetical) question...is it possible for a 24bit version of the same release, as in my Achtung Baby example, to sound different for any reason? If the comparison is not made from the same source/format file, is the sound different? Based on your answer above, it seems possible? Has anyone explicitly tested this??


----------



## U2Bono269

bigshot said:


> What codec? What bitrate? Did you make the compressed files yourself or did you compare a CD to an iTunes download? Did you line level match? Did you have direct A/B switching? Did you compare the samples blind? You have to test carefully to get accurate results.


 

 I did it using the Foobar plugin with a redbook file I compressed myself at 320kbps, as suggested by the other user.


----------



## bigshot

320 LAME or Fraunhofer?


----------



## U2Bono269

bigshot said:


> If you get the 16 and 24 from two different sources, you can't be sure the mastering is the same. That's why you make the 16 from the 24 yourself to test. You want to remove irrelevant variables and focus on what you are testing for. If you are testing just for the difference between 16 and 24 bit, you just want to test for that and not have other factors muddy the waters.


 
  
 Making the files to test myself isn't what I'm curious about. I'm curious about the mastering between the 2 formats. I'm curious about the factors that muddy the waters. Because if I have a CD version, and there's an HD version that does sound different because of the mastering, despite the fact that they are similar sources, I'd be interested in hearing that too. I am interested in the variables. Is that crazy?


----------



## U2Bono269

bigshot said:


> 320 LAME or Fraunhofer?


 
  
 LAME. But this is getting away from what I'd like to learn more about.


----------



## bigshot

Not at all... Unfortunately, just because it's on HD tracks, it doesn't mean that it is improved. It could be worse than a CD release. It's different with every record, so you have to ask other record collectors who have heard the various masterings which is the best.


----------



## RRod

u2bono269 said:


> I get that, which brings me back to my original (hypothetical) question...is it possible for a 24bit version of the same release, as in my Achtung Baby example, to sound different for any reason? If the comparison is not made from the same source/format file, is the sound different? Based on your answer above, it seems possible? Has anyone explicitly tested this??


 
  
 Of course it can, if the HD site in question purposefully releases a different master for the 16bit version. I have heard of, for instance, SACDs where the mastering differed between the SACD and Redbook layer, beyond what a simple DSD->PCM conversion would yield; see here for some possible examples.


----------



## U2Bono269

rrod said:


> Of course it can, if the HD site in question purposefully releases a different master for the 16bit version. I have heard of, for instance, SACDs where the mastering differed between the SACD and Redbook layer, beyond what a simple DSD->PCM conversion would yield; see here for some possible examples.


 

 What if the master is the same, and one file comes from the SACD layer, and the other comes from the CD layer...will they sound the same either in theory or in practice?
  
 And to go further, if the master is the same on a separate CD and a separate SACD, will they sound the same in theory or in practice?


----------



## RRod

u2bono269 said:


> What if the master is the same, and one file comes from the SACD layer, and the other comes from the CD layer...will they sound the same either in theory or in practice?
> 
> And to go further, if the master is the same on a separate CD and a separate SACD, will they sound the same in theory or in practice?


 
  
 Well note that you can't really master in DSD; you have to convert to PCM to be able to do any of the typical digital manipulations. But we still fall back into the format wars, as SACD proponents would say that the second you go to PCM you lose some special "magic" of DSD. In practice, if all you did was convert the DSD to Redbook competently, you couldn't hear a difference in blind testing.


----------



## U2Bono269

rrod said:


> Well note that you can't really master in DSD; you have to convert to PCM to be able to do any of the typical digital manipulations. But we still fall back into the format wars, as SACD proponents would say that the second you go to PCM you lose some special "magic" of DSD. In practice, if all you did was convert the DSD to Redbook competently, you couldn't hear a difference in blind testing.


 

 Thank you. Has anyone done an ABX in this sort of scenario?


----------



## RRod

u2bono269 said:


> Thank you. Has anyone done an ABX in this sort of scenario?


 
  
 Well this is the common reference, but it has its own critics (as you might imagine).


----------



## bigshot

u2bono269 said:


> What if the master is the same, and one file comes from the SACD layer, and the other comes from the CD layer...will they sound the same either in theory or in practice?


 
  
 In practice, the only SACDs I have that have identical mastering on both layers are by PentaTone. Most SACDs have deliberately hobbled redbook layers. This is particularly true of rock legacy titles. I've found that about a quarter of the SACDs I've bought sound better (usually because of a good 5.1 mix), a quarter sound worse, and half of them sound exactly the same as the CD. Not a very good batting average. No surprise the format is dying.


----------



## dprimary

greenears said:


> I don't agree with that.  There s plenty of science on this thread that shows modern MP3 320 or AAC 256 the differences to lossless are inaudible so no need to remaster anything due to format.  The reason they are doing this push to remaster is to optimize for typical listening conditions of iTunes users (which I have to assume is mobile earbuds).  Which sucks if you are not typical.
> 
> It would be nice if they start releasing with multiple masters for different conditions, put it as parameters in the EQ.


 

 When you release an album you create a production master for every format you have at some point. Of course for a downloadable format the file is the end of it. Put a copy on the server and you are done. Amazon for example seems to be MP3 256 pretty good but audible. AAC 256 for many things is inaudible for others you can tell but is not easy. None of is night and day like some claim.
  
 So you start out with a stereo 24/44.1 mix master (I will leave out higher sample rates to keep it simpler) the mastering engineer will edit and clean up the beginning and end of each track, put the tracks in order of the album adjust the spacing of each track, some tracks might crossfade. Then each track is adjusted for level, eq, dynamic range, possible noise removal. If you intending it to be listens to as an album time between tracks and levels are more important for it to flow as intended. The peak is brought up to just below 0 dBfs this likely took at least a day and a few thousand dollars of time. This is the final master. It will sound as good as it possibly can. Often the producer, engineer and artists  are there for the whole process. 
  
 From this 24bit 44.1 master you will create production masters
  
 Production masters that require no additional adjustments.
  
 WAV 24/44.1
 AIFF 24/44.1
 FLAC 24/44.1
 ALAC 24.44.1
  
 These are pretty much a straight forward conversion, still you don't want to sell 10,000 defective FLAC files before anyone catches it. So these are converted and checked for problems.
  
 The CD will need to to dithered down to 16bit and have track pointers and such added to it to make a rebook master which will be sent to the pressing plant.
  
 Production masters that can require additional adjustment.
  
 From the final master or 16 bit master depending on the encoder
  
 MP3 128
 MP3 256
 MP3 320
  
 AAC 128
 AAC 256
 AAC 320
  
 Ringtones
  
  
 Production masters that require additional adjustments.
  
 LP's
 12" singles
 7" single
  
 each will require a lacquer to be cut, each is processed differently.
  
 And the lowly cassette
  
 With singles on a ten song album you already have 33 production masters. More if you count each downloadable track as an individual master
  
 I was told a recent pop album had over 500 masters. I have to feel sorry who ever had to do that.


----------



## Greenears

u2bono269 said:


> Because I stated in a previous post that I have done ABX testing between mp3s and FLACs, yet multiple posters keep suggesting I do it as if I never said it to begin with...


 

 Great! So I guess you came to the same conclusion everyone else has:  MP3-320 LAME encoded will fail an ABX against lossless.
  
 So what were we arguing about again?


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## U2Bono269

Listen, I came here to learn and discuss. I don't appreciate the attitude. To my ears lossless sounds better. That's what matters to me so please stop bugging me about it.




Sent from my iPad using Tapatalk


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## Greenears

Sounds like you failed the ABX and refuse to believe the writing in the log file.  You just don't want to say it that way.   Welcome to the club.


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## U2Bono269

Why do you care so much? I didn't know I had to agree with you in order to participate. 

I think I've gotten what I can out of this thread and it's time for me to move on to other topics. Thanks to those who helped me! 


Sent from my iPad using Tapatalk


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## lamode

dprimary said:


> A MP3 or AAC release would have different masters,


 
   
 Why do you think so? I would be surprised if they bothered to remaster for MP3 at all compared to CD. What would be the point?
  
 Quote:


> 16 bit has a noise level -96dB while 24bit give about 24dB more so the noise is roughly -120dB down, which is limited by physics to get much better. An extremely silent studio will be about 20dB SPL you would playing the music at an ear splitting116dB SPL before you to start to hear the noise floor of the 16bit file.


 
  
 24dB has a noise level around -144dB, which is a 48dB difference.


----------



## bigshot

u2bono269 said:


> Why do you care so much? I didn't know I had to agree with you in order to participate.


 
  
 It's not that we have to agree... it's that if you can prove that you can successfully and consistently discern a difference between say MP3 LAME 320 VBR or AAC 256 VBR, you will have done something no other human being has been able to do before under controlled testing. That is VERY interesting to those of us who are interested in compressed audio codecs and how to use them. Either your hearing is beyond normal human's ability, or perhaps your comparison wasn't very well controlled and expectation bias crept in making your test results invalid. We're asking the questions to determine which is the case.


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## RRod

lamode said:


> 24dB has a noise level around -144dB, which is a 48dB difference.


 
  
 I think he means that the actual effective bits out of an ADC/DAC never gets to 24 in practice. 20—21 is the best I've seen actually tested results for.


----------



## RRod

bigshot said:


> It's not that we have to agree... it's that if you can prove that you can successfully and consistently discern a difference between say MP3 LAME 320 VBR or AAC 256 VBR, you will have done something no other human being has been able to do before under controlled testing. That is VERY interesting to those of us who are interested in compressed audio codecs and how to use them. Either your hearing is beyond normal human's ability, or perhaps your comparison wasn't very well controlled and expectation bias crept in making your test results invalid. We're asking the questions to determine which is the case.


 
  
 I thought he already admitted he can't tell a blind difference. Isn't his main question whether 24bits is a "mark" signifying better mastering? Hopefully we've argued convincingly otherwise, but perhaps an example of a CD with superior mastering to an HD release might help answer his question better.


----------



## lamode

u2bono269 said:


> Thank you. Has anyone done an ABX in this sort of scenario?


 
  
 There are tests comparing a 16/44 AD-DA loop to a live feed, and no-one could hear the difference. Given that the live feed is the original, it is by definition better than any possible digitized version, regardless of resolution, this makes the question about HD audio moot.


----------



## lamode

u2bono269 said:


> To my ears lossless sounds better.


 
  
 If that were really true, you could easily pass an ABX test.


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## U2Bono269

To be fair I only did the mp3/flac test once. I think I got 8 out of 10. But when I listen to stuff I like the sound of the flac better. Sounds fuller to me. I can't tell 100% of the time. For example for fighters stuff sounds the same. But for the most part it sounds different to me. I never said I could tell 100% of the time. 

My question was indeed about the mastering of 24 bit files. I think it's been answered pretty well. I don't expect to hear any differences or better mastering with 24 bits. 


Sent from my iPad using Tapatalk


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## dprimary

> dprimary said:
> 
> 
> > A MP3 or AAC release would have different masters,
> ...


 
 Read the mastering for iTunes part of the page
 http://www.justmastering.com/article-masteredforitunes.php
  
 the point would be to deliver the best sound you can for each format. Believe it or not huge amounts of effort went into getting what quality they could into high speed duplicated cassettes in the 80's
 Even if I did not use a certified  mastered for iTunes mastering engineer I would still check the master myself before sending to apple.
  
 A -144dB signal to noise ratio converter cannot be made. Most good 24bit converter are around -120dB as I said before physics limits what we can build. The thermal noise of one resistor - around -135dB there is a bit more then one resistor in a DAC. An Audio Precision APx555 has a THD+N of -120dB which is about as good as it gets.


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## lamode

dprimary said:


> Read the mastering for iTunes part of the page
> http://www.justmastering.com/article-masteredforitunes.php


 
  
 There was zero information on that page, just marketing fluff.
  
 And there is a difference between transcoding and creating a new master.


----------



## Greenears

lamode said:


> There are tests comparing a 16/44 AD-DA loop to a live feed, and no-one could hear the difference. Given that the live feed is the original, it is by definition better than any possible digitized version, regardless of resolution, this makes the question about HD audio moot.



I an aware of tests with 16 bit loop off a recorded feed of various formats, but not live. That would indeed settle it. Link please to the full testing methodology.


----------



## bigshot

u2bono269 said:


> I don't expect to hear any differences or better mastering with 24 bits.


 
  
 You might hear an improvement, but it has nothing to do with the file being 24 bit. Formats don't matter. The quality of the recording, mixing and mastering does. That depends on the skill, budget and most of all taste of the engineers working on the record.


----------



## lamode

greenears said:


> I an aware of tests with 16 bit loop off a recorded feed of various formats, but not live. That would indeed settle it. Link please to the full testing methodology.


 
  
 I tried googling to find the study I saw a while back but could only find references to the study (without links), and not the study itself. If I find it I will post it (thought I have a feeling I already have  )


----------



## dprimary

lamode said:


> There was zero information on that page, just marketing fluff.
> 
> And there is a difference between transcoding and creating a new master.


 
  
 It gets into some of the requirements that apple has for iTunes. They do expect the mastering engineer to confirm it will encode properly.  
 Every mastering studio tells me the same thing iTunes is a separate master. If you send in a CD master it will likely clip and apple will reject it. Apple does provide all the tools to do this, anyone can download them.
  
 Different production masters is not new it was done this way for decades. Just for about 10 years the only format was CD so multiple production masters was forgotten about. You always have wanted to maximize the quality of every format. Ok some are in it for the glamor. Then they need to eat.


----------



## bigshot

When you transcode from Redbook to AAC, the volume bumps up a hair. So you normalize down to 90% and then transcode. It isn't an audible difference. Just a slightly lower volume level.


----------



## 1c3d0g

analogsurviver said:


> Did you manage to get micro sounding better than the Realtek onboard chip ?
> 
> I had quite "a-not-so-pleasent-ball-with-my-PC" before my nano started to sound as it should. PC related, registry leaning etc etc. And it takes a few days to configure whatever software you are using to your liking/preference - ALWAYS taking in account of the limitations of your hardware.
> 
> It is only fair to any DAC to clearly state with which hardware and software ( and how configurd ...) it is being used - it can do nothing on its own.


 
 I'm still waiting for my Fostex TH900 to arrive - it should be coming next week. I'll reserve final judgment when I can rule out the Sony MDR-XB1000 headphone as a possible inferior variable. Sometimes, some headphones and amps just don't have the right synergy.
  
 Oh, and as for the PC, it's freshly formatted notebook with the latest drivers & updates on everything. It's a real audible difference, not just a slight difference. Explosions sound better, you can hear footsteps right behind you etc. With the iDSD, it's much less pronounced and almost kind of lifeless. You'll be hearing from me when I've done proper testing.


----------



## jugate

rrod said:


> How exactly do you have things set up? The HD surround formats have the ability to send out a non-HD "core" when connected with something like optical cable that can't handle the full bandwidth, but this all depends on the exact settings of your source and sound card.


 
  
 SOUNDBLASTER Z. Decoder software by windDVD PRO. Conected to z906 logitech by analog 3.5mm cables.


----------



## jugate

dolby true hd = 5.1 flac? is a lossless codec?
  
 Quote:


bigshot said:


> Blu-ray disks have multiple audio tracks in various formats. Generally, you set your player to default to the highest quality track your player is able to decode and don't worry about it any more. The only really significant difference in sound quality is 2 channel vs 5.1/7.1.


 
  
 ok but can 24 bits sound WORSE that 16 bits? because dolby/dts HD are recorded in 48khz/24 bits.


----------



## U2Bono269

So, I wanted to check back in with my observations. I got my iBasso DX50 yesterday and I spent quite a bit of time with it in the past 24+ hours.
  
 My goal was to listen to U2's Achtung Baby Deluxe Edition in 2 different formats. I did my research, and it would seem the HDTracks (24-44.1) version and CD version come from the same master. The Edge stated that the album, when rereleased for this edition, was not remastered, but rather tweaked. There were no further comments from him. Based on what I've read here, it would stand to reason that the album was dithered to redbook from the 24bit master, right?
  
 They sounded the same, except for one tiny detail. It sounded to me that reverb in certain tracks sounded clearer and airier on the HDTracks version. I was not expecting that. I think I noticed only because I've listened to this album hundreds of times since I first bought it way back in 1997.
  
 Perhaps this means the 2 are indeed a separate master, or the redbook version may have been adjusted slightly. Or I'm imagining it.
  
 From a layman's view, I suspect that there may be benefits to the high resolution audio movement aside from the science of it. I don't know what they would be; I've just been idly thinking while shoveling snow.


----------



## lamode

jugate said:


> ok but can 24 bits sound WORSE that 16 bits? because dolby/dts HD are recorded in 48khz/24 bits.


 
  
 Dolby is really irrelevant to this argument.
  
 And no, why would 24 bits sound worse than 16 bit, if we have already established that you can't hear a different between 16 and 24 bit.


----------



## bigshot

u2bono269 said:


> My goal was to listen to U2's Achtung Baby Deluxe Edition in 2 different formats. I did my research, and it would seem the HDTracks (24-44.1) version and CD version come from the same master. The Edge stated that the album, when rereleased for this edition, was not remastered, but rather tweaked. There were no further comments from him. Based on what I've read here, it would stand to reason that the album was dithered to redbook from the 24bit master, right?


 
  
 Not necessarily. There's remastering, which often involves remixing and just normal mastering, where just the volume and compression levels are adjusted. The recording artist isn't generally consulted for the latter. The only way to know that two versions are identical is to make the redbook version yourself by bumping down the 24 bit copy.


----------



## U2Bono269

Then, it was not remastered, just mastered. In U2's case, The Edge was involved in the mastering/remastering of the Deluxe Editions they've put out since 2007. Albums like Boy, October, War, Unforgettable Fire and Joshua Tree were definitely remastered. Achtung Baby was not. But that's beside the point...both the CD and HDtracks version are from the same project. While I can't be 100% certain of that without having been there with them, evidence and logic suggest it to be the case.
  
 But like I have said, I am not interested in simply downsampling and comparing. As a listener, I feel like that's only part of the story. I can't say this enough...I want to know the differences between similar commercial releases between formats.
  
 Yes, if I take the HDTracks version and downsample it, they are identical. But in practice, it seems that if I buy the 2012 deluxe release of Achtung Baby on CD, and I buy the 2012 deluxe release of Achtung Baby from HDTracks, there is an admittedly minute difference. So somewhere in the process of converting that 24bit recording into a 16bit redbook, something changed.
  
 The consensus in this thread seems to be that high resolution tracks aren't worth it because on a purely scientific basis, there's no difference. Again, I agree with that. But theory and practice are often two different things. If labels are putting out high resolution albums that have even minute differences, it kind of negates that conclusion, doesn't it? If I want to hear that airier reverb while listening to Zoo Station, the only way for me to get it is to buy the HDTracks version.


----------



## RRod

u2bono269 said:


> The consensus in this thread seems to be that high resolution tracks aren't worth it because on a purely scientific basis, there's no difference. Again, I agree with that. But theory and practice are often two different things. If labels are putting out high resolution albums that have even minute differences, it kind of negates that conclusion, doesn't it? If I want to hear that airier reverb while listening to Zoo Station, the only way for me to get it is to buy the HDTracks version.


 
  
 I know my personal feeling is that they shouldn't be charging $15-$30 just for a bit more reverb or to make us pay for a good mastering that they should have put on CD in the first place.


----------



## bigshot

In the case of Led Zeppelin and Frank Sinatra, the vinyl records from the original release dates often trump any digital release, even though LPs are much lower fidelity than either CDs or 24bit. In the case of the Beatles, the first CD release was a straight transfer off the original master tapes, while the recent remasters for the stereo box are slightly compressed. So in that case, the old CD is better than the remastered CD. Donald Fagen's The Nightfly, one of the best sounding records ever made, sounds best on Japanese SACD, even though the record was recorded in the early days of digital at 16/44.1.
  
 The format is completely unrelated to sound quality.
  
 If you want to know which of the releases of the U2 records sound the best, the place you are going to find that out is in the Music forum, not Sound Science. Rekkid collectors can tell you which release sounds the best. All we can tell you here is whether a particular format has the *potential* for sounding better or not. And you've already gotten that answer- just about any format you buy an album in, from the iTunes store download to massive DSD files all have the same potential for sounding perfect.


----------



## U2Bono269

I'm not looking to find out which U2 releases sound best. I pretty much own them all, so I can figure it out myself (and have!). I just used it as an example. I'm trying to understand the correlation between the science of 24bit vs 16bit and the real life use of it. I appreciate everyone joining in the discussion with me. But while I understand more of the science now, I don't feel like I've come to a satisfying conclusion. But that's ok, it's just all the more reason to read and learn new things. I'm probably not asking the right questions to find what I'm looking for, because I most likely don't know what the right questions are. But I'm trying.


----------



## heart banger-97

Blind test them for yourself. Theory has shown us, experiment would test it.


----------



## James-uk

u2bono269 said:


> I'm not looking to find out which U2 releases sound best. I pretty much own them all, so I can figure it out myself (and have!). I just used it as an example. I'm trying to understand the correlation between the science of 24bit vs 16bit and the real life use of it. I appreciate everyone joining in the discussion with me. But while I understand more of the science now, I don't feel like I've come to a satisfying conclusion. But that's ok, it's just all the more reason to read and learn new things. I'm probably not asking the right questions to find what I'm looking for, because I most likely don't know what the right questions are. But I'm trying.



The conclusion is the format doesn't matter. end of conversation. The best master is subjective. You might prefer the 1985 original vinyl release and I might prefer the 2015 digital 24/196 remastered SACD blu Ray TM high res /super HD 4k super duper 3D real TM hi res version. Either way CD could perfectly emulate both.


----------



## Greenears

u2bono269 said:


> .....
> 
> Perhaps this means the 2 are indeed a separate master, or the redbook version may have been adjusted slightly. *Or I'm imagining it.*
> 
> From a layman's view, I suspect that there may be benefits to the high resolution audio movement aside from the science of it. I don't know what they would be; I've just been idly thinking while shoveling snow.


 
  
 So what people are trying to tell you, in a roundabout way, is that most posters on this long and winding thread seem to agree on only one thing: Masters matter, masters are audibly different, and are independent of format.  It's not really a Sound Science question to know what some producer or engineer did X number of years ago but there are many forums that I think do a good job with that.
  
 However, if you want to apply science to figure out if you are imagining the difference, you simply make 3 files: A - the CD version, B - the 24 bit version, C - the 24b version truncated down to 16.  Scroll back for all the tools needed to do this - they are free and easy to install, and many have done this including me.  You ABX test A vs B, and A vs C.  If you can't use that improved air you heard to pick out the difference in at least one pair, then you were imagining it.  It's quite common to imagine things.
  
 That's all there is to it.


----------



## jugate

lamode said:


> Dolby is really irrelevant to this argument.
> 
> And no, why would 24 bits sound worse than 16 bit, if we have already established that you can't hear a different between 16 and 24 bit.


 
  
 not because if dolby/dts use 24 bits, is for something
  
 simple question: can you differentiate mp3 vbr/320 from a flac?


----------



## bigshot

u2bono269 said:


> I'm not looking to find out which U2 releases sound best. I pretty much own them all, so I can figure it out myself (and have!). I just used it as an example. I'm trying to understand the correlation between the science of 24bit vs 16bit and the real life use of it.


 
  
 Oh! Sorry! Then the answer is simple. It doesn't make any difference at all. No human can hear the difference between 16 and 24. Read the first post in this thread and you're done.


----------



## kraken2109

jugate said:


> ok but can 24 bits sound WORSE that 16 bits? because dolby/dts HD are recorded in 48khz/24 bits.


 
 Dolby TrueHD and DTS Master Audio are both lossless formats.
 24 bit cannot sound worse than 16 bit.


----------



## stv014

kraken2109 said:


> 24 bit cannot sound worse than 16 bit.


 
  
 That is, unless it is a worse master than the (older and in some cases dynamically less compressed) 16 bit version.


----------



## Danz03

Whichever version sounds best is very subjective in a way. Someone like Brian Eno, who is a very experimental producer, did a lot of unconventional things for the production of U2's LPs, like fading in the intro of a song in mono and using a lot of lo fi techniques. The remasters could be technically better but not necessarily better artistically. An example would be the 25th anniversary surround mix of Queen's 'A Night At the Opera', of course technically, it's superior to the original mix from the 70s, but it doesn't mean everyone would prefer it to the original.



u2bono269 said:


> I'm not looking to find out which U2 releases sound best. I pretty much own them all, so I can figure it out myself (and have!). I just used it as an example. I'm trying to understand the correlation between the science of 24bit vs 16bit and the real life use of it. I appreciate everyone joining in the discussion with me. But while I understand more of the science now, I don't feel like I've come to a satisfying conclusion. But that's ok, it's just all the more reason to read and learn new things. I'm probably not asking the right questions to find what I'm looking for, because I most likely don't know what the right questions are. But I'm trying.


----------



## Greenears

@U2Bono269  >I don't feel like I've come to a satisfying conclusion
  
 Don' be unsatisfied.  I wrote a whole three page thing about 20 pages back on this thread, you can read it if you want.  But the essence is don't be unsatisfied.  We're in the golden age of audio.  The problem is that some minor aspects of the science are murky so it may be hard to reach a definitive conclusion unless someone spends the money to do a test of about 300 listeners on modern (2015) equipment. 
  
 However, it is very safe to say, that the absolute limits of human hearing are somewhere between 16-20 bits.  And 20 bits would be very loud, like front row of rock concert loud.  There is a reason to record things with more than 20 bits, which are technical problems of recording hardware and environment.   But once it is mixed and leveled you can drop the extra bits with absolutely no audible difference.
  
 The most important thing I wrote is that although some disagree there is anything audible beyond 16 bits, even if it turns out there is something it is very very subtle.  20 vs 16 is not 25% better, it is 0.0001% better, if at all.  My testing in fact showed that even 12 bits sounds really amazing with a good quality master and nearly transparent.  Bits are exponential not linear. 
  
 The problem I see now is that people are figuring on that 24 bit sounds the same as 16 bit, so the vendors are changing the master so that it sounds different to justify higher prices.  Whether better or worse sounding I don't know but it's got nothing to do with number of bits but of course everyone will get confused.  That is life.


----------



## bigshot

amen


----------



## U2Bono269

In terms of the audio itself, I feel very satisfied. In the past month or so since I've become more invested in increasing the quality of my audio equipment, my satisfaction has gone up siginificantly. Sometimes I feel like I'm listening to old albums for the first time, which is wonderful.
  
 I did a lot of reading, and it seemed to me that maybe high resolution audio might allow for different mastering than was previously possible. I know now that this is not the case. My frustration was that I asked about that, and the majority of answers simply told me that 16 bits and 24 bits were indistinguishable and if I heard a difference it was because the master was different and I should do testing etc etc. But that wasn't what I was asking about. It took a while for someone to actually answer my question, and now I know that a CD can convey the same thing an HD track can. That wasn't clear to me at the time. So yes, the answers I received here were and are somewhat unsatisfying. It's one thing to compare 24bit to 16bit in a perfect vacuum, but I'm not yet convinced there are no other factors. Maybe our listening technology just isn't there yet? Who knows.
  
 Maybe there are intangible differences or benefits that can't be quantified by our ears. There's a convenience store near me that uses one of the annoying high-frequency sounds at night to warn off teenage loiterers. I could hear it myself until I was about 27 or 28 years old (I'm 32 now). Now, when I go to the store, I don't hear the sound anymore, but if I stay there for longer than a few minutes, my ears start to feel full. I have sensitive ears, so if I'm sick or congested, my ears will actually hurt. So maybe having those extra bits and frequencies might be worth it at some point, particularly if technology is advanced to a point we can't really comprehend right now. Or maybe there are some people who are able to sense the vibrations of higher and lower frequencies and it contributes to their enjoyment of the music. Can that be proven? Probably not. Just something I think about during my long commute.
  
 There are benefits  to HD music for me, however. I collect and trade concert recordings, and in that community lossless is king. Lossy files, no matter how transparent, are strictly forbidden. So I have a desire to go for whatever is closest to the original studio recording. If there are studio-quality versions of my favorite albums, that's what I want for my collection. If it's a different master, then I definitely want it. Luckily for me, my FLAC collection is strictly curated, so it kind of limits how much of a money pit this hobby becomes. For many albums, Spotify's high quality 320 stream is the best for me. For that reason, places like Pono and HDTracks will have a share of my business in the future.


----------



## StanD

@UBono269 CD's are 16 bit so perhaps some of those replies are relevant. If 16 and 24 bit make no difference at the distributed product, then I would think your question was answered.


----------



## U2Bono269

stand said:


> @UBono269 CD's are 16 bit so perhaps some of those replies are relevant. If 16 and 24 bit make no difference at the distributed product, then I would think your question was answered.


 
  
 But that was my point at the time: there could be a difference between commercial products and tracks we downsample ourselves in the comforts of our own home. You or I can take a 24/192 file, downsample it to 16/44.1, and it'll sound the same. I wondered if the same thing was done at the commercial level. Do professionals downsample like we do, and then do something else to the sound to make it more CD-palatable? Do engineers do things with the extra bits that we don't? Do all 24bit albums I can buy on HDTracks have identical-sounding CDs I can buy in the wild? Those questions are much more specific than "is there a noticeable difference?"
  
 What I learned (I hope) was that without transparency in regards to mastering, we may never know that. I also learned that that's not related to the bits themselves, which I didn't know at the time, and has nothing to do with ABX testing.
  
 So yes, those questions were eventually answered, but not in a manner than satiated my curiosity. If anything they just gave me more questions. That's all I meant by that.


----------



## RRod

u2bono269 said:


> But that was my point at the time: there could be a difference between commercial products and tracks we downsample ourselves in the comforts of our own home. You or I can take a 24/192 file, downsample it to 16/44.1, and it'll sound the same. I wondered if the same thing was done at the commercial level. Do professionals downsample like we do, and then do something else to the sound to make it more CD-palatable? Do engineers do things with the extra bits that we don't? Do all 24bit albums I can buy on HDTracks have identical-sounding CDs I can buy in the wild? Those questions are much more specific than "is there a noticeable difference?"
> 
> What I learned (I hope) was that without transparency in regards to mastering, we may never know that. I also learned that that's not related to the bits themselves, which I didn't know at the time, and has nothing to do with ABX testing.
> 
> So yes, those questions were eventually answered, but not in a manner than satiated my curiosity. If anything they just gave me more questions. That's all I meant by that.


 
  
 Engineers CAN do extra things to any release that they didn't do to the initial CD release, but they don't HAVE to. I think that's a point people are trying to make. An engineer is more than welcome to make a great mix/master for the HDTracks download and make a trash, compressed, clipped, mono sounding master for the CD. That has nothing to do with 16 vs. 24 bits, but more about making $$ by leading people to believe that it does.


----------



## U2Bono269

rrod said:


> Engineers CAN do extra things to any release that they didn't do to the initial CD release, but they don't HAVE to. I think that's a point people are trying to make. An engineer is more than welcome to make a great mix/master for the HDTracks download and make a trash, compressed, clipped, mono sounding master for the CD. That has nothing to do with 16 vs. 24 bits, but more about making $$ by leading people to believe that it does.


 

 And I just said that I understand that now. I did not at the time that I came to this thread.


----------



## RRod

u2bono269 said:


> And I just said that I understand that now. I did not at the time that I came to this thread.


 
  
 So then what extra questions did coming to this understanding bring up?


----------



## U2Bono269

Well I have questions about the mastering process and such, but it was made clear to me that this is not the place for that. I was just explaining myself based on bigshot's post.


----------



## StanD

u2bono269 said:


> Well I have questions about the mastering process and such, but it was made clear to me that this is not the place for that. I was just explaining myself based on bigshot's post.


 
 There you go, ready to rock.


----------



## U2Bono269

stand said:


> There you go, ready to rock.


 

 Yeah, I found some really great articles.


----------



## reginalb

u2bono269 said:


> And I just said that I understand that now. I did not at the time that I came to this thread.


 
  
 I think that there is just a lot of frustration with people that have answered the same questions over and over again. Provided piles of links, there are several threads in this forum where the OP has a pile of lins, there are some right at the top when you arrive at the forum as well, answering all of the questions that get asked repeatedly. And many people from other parts of the forum come in to this section and get hostile about it. People, as a result, put their guard up. Glad you eventually found the information that you were looking for.


----------



## Avi

Maybe someone can collate those links and basic answers to form the nucleus for a sticky post on the topic?


----------



## Soundsgoodtome

You guys might want to check this out and the ongoing convo. More interestingly the article on Morning Phase in HDtracks..


----------



## gebo

greenears said:


> >
> 
> We know and accept there are mastering differences, *but generally I think if 16 and 24 are offered side to side on the same site, I think it's the same mastering.  At least I've never heard anyone claim contrary.*


 
  
 Is this the unanimous opinion of the experts on this forum?
 Has anybody done comparative measurements or the like?


----------



## StanD

gebo said:


> Is this the unanimous opinion of the experts on this forum?
> Has anybody done comparative measurements or the like?


 
 Let's be practical, who has the time and money to compare the complete offerings on all of these sites? Which site has enough resources to remaster everything on their site?


----------



## Greenears

gebo said:


> Is this the unanimous opinion of the experts on this forum?
> Has anybody done comparative measurements or the like?


 
 "Unanimous" and "forum" in the same sentence is an oxymoron.
  
 I was told by others (reasonably convincingly, with measurements) that AC/DC new Rock or Bust the 24 bit version is identical to the Europe/Australia CD release mastering, the 16 bit is the US CD which is measurably and audibly different.
  
 I have seen other claims (with less detail) that some 24 bit was simply upsampled 16 bit. 
  
 You can try it yourself with foobar dynamic range plugin.


----------



## U2Bono269

I hesitate to bring this up again, but I found an article that's a little confusing and I think this is still the best place to ask it.
  
 I found an article about Hesitation Marks, Nine Inch Nails last album. Here's the link:
  
 http://nineinchnails.tumblr.com/post/59587808317/hesitation-marks-was-mastered-in-two-different
  
 Basically, if you bought the CD, you could download an audiophile version (24bits, 48khz) for free.
  
 I interpret this article to say that it was impossible to have a modern "loud" album, and have good dynamic range at the same time. It seems they wanted an album that was both loud and had great dynamic range and couldn't do it on a CD. Is redbook not sufficient for this? If you want a loud album with high DR, this article indicates that you'd need more than a CD can address. Is that a correct assessment on my part? If so, is that due to the increased bits, or the increased range?
  
 Regardless of what you think of the loudness wars, I really do like NIN stuff mastered loudly. I gave the Audiophile version a listen and it does have the loudness I'd expect of a modern album, and I thought it sounded more detailed and rich than the CD version.


----------



## Skyyyeman

This is a recent "Application Note" by John Sau, chief engineer for Benchmark Media, makers of highly regared audiophile and pro audio digital equipment.  (I didn't check if anyone posted this earlier.)  This article addresses many of the earlier comments on this subject.
  
 http://benchmarkmedia.com/blogs/news/14949345-high-resolution-audio-bit-depth


----------



## RRod

u2bono269 said:


> I hesitate to bring this up again, but I found an article that's a little confusing and I think this is still the best place to ask it.
> 
> I found an article about Hesitation Marks, Nine Inch Nails last album. Here's the link:
> 
> ...


 
  
 Absolute loudness, that is, how loud your ears hear the actual sound out of your speakers/phones, depends on how high you set your amp. Let's say you set it to where the relatively loudest passage of an album is uncomfortably loud. Dynamic range is then how absolutely loud the quietest parts of your album sound. 16bits allows these quiets parts to still be quite soft even if the loudest parts are near intolerable, at least in any normal listening room.
  
 "Loudness war" albums have had their soft parts brought up, so that the dynamic range of the music is reduced, but again this isn't the "fault" of 16bits, it's the fault of the person making the decision to reduce the range in this way.


----------



## RRod

skyyyeman said:


> This is a recent "Application Note" by John Sau, chief engineer for Benchmark Media, makers of highly regared audiophile and pro audio digital equipment.  (I didn't check if anyone posted this earlier.)  This article addresses many of the earlier comments on this subject.
> 
> http://benchmarkmedia.com/blogs/news/14949345-high-resolution-audio-bit-depth


 
  
 That article stays rather honest despite mentioning company hardware; kudos to that! Still underplays a bit just how ear-splittingly loud 130dBspl is, though.


----------



## StanD

skyyyeman said:


> This is a recent "Application Note" by John Sau, chief engineer for Benchmark Media, makers of highly regared audiophile and pro audio digital equipment.  (I didn't check if anyone posted this earlier.)  This article addresses many of the earlier comments on this subject.
> 
> http://benchmarkmedia.com/blogs/news/14949345-high-resolution-audio-bit-depth


 
 This article even tries to help the 24 bit world. It states that the human perception threshold is 0 dB SPL, I believe it's really 4 dB SPL. It uses a 130 dB SPL figure for the threshold of pain, figures vary from 120 to 140 dB. I'd say that 120 dB is incredibly loud and will cause most people pain, other than half deaf guitarists that stand next to their speakers at full blast. Even with this 14 dB added edge, 16 Bits still came up as the best practical bit depth for playback systems.


----------



## caml

u2bono269 said:


> I hesitate to bring this up again, but I found an article that's a little confusing and I think this is still the best place to ask it.
> 
> I found an article about Hesitation Marks, Nine Inch Nails last album. Here's the link:
> 
> ...


 
  
 That is not the way I understand the article. What I gather is that they released a normal version with more heavily compressed dynamics and the very low frequencies cut off, to sound good on any setup. And an audiophile version less heavily compressed and with all sub bass frequencies kept.
  
 The very low frequencies convey a lot of energy that 1) interferes with the process of dynamic compression (these very low frequencies will account for most of the energy of the sound during the dynamic compression process, in a way that is not proportional to our ears perception of these frequencies) 2) sub bass frequencies are not heard on an average sound system but can still alter the reproduction of the higher frequencies (in a bad way). Those are 2 reasons to release such a cut off version.
  
 They could have released the "audiophile" mastered version in 16 bits, that wouldn't have made a difference.
  
 I have the "audiophile" version... Just for fun I used a dynamic range meter on track #2 "Copy of A" : a whopping 4,7 dB dynamic range throughout the whole song... I think that could fit on a 16 bit CD alright


----------



## U2Bono269

Thank you. That makes sense. 


Sent from my iPad using Tapatalk


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## Skyyyeman

Here's an earlier (8/14) but similar article by John Siau, chief engineer of Benchmark, entitled "AUDIO MYTH - "24-BIT AUDIO HAS MORE RESOLUTION THAN 16-BIT AUDIO"
  
 http://benchmarkmedia.com/blogs/news/15121729-audio-myth-24-bit-audio-has-more-resolution-than-16-bit-audio


----------



## jugate

kraken2109 said:


> Dolby TrueHD and DTS Master Audio are both lossless formats.
> 24 bit cannot sound worse than 16 bit.


 
  
 ok thanks!


----------



## knucklehead

stand said:


> It states that the human perception threshold is 0 dB SPL, I believe it's really 4 dB SPL.


 
 I think it's just that it would be difficult to while living ... for very long ...   Things like blood flowing in your head and random air motion hitting your eardrums is about 4dB AFIK. Looking for a reference, I just saw breathing listed at 10dB.


----------



## Soundsgoodtome

I apologize if this questions seems elementary. I swear I'm not trolling and this is an honest question. I understand the idea behind 16/44 being "all the human hearing needs" in terms of SPL, studio work aside for lower noise floors,etc for 24-bit, but we're talking about bits and data-rate here to capture and replay electricity.

 Would a 24bit depth and higher frequency rate have more *accuracy* in capturing an analog wave even if just in theory and without applying DAC implentations etc.? This is probably a facepalm question but I'm having one of those Saturdays after a Friday night out and the question lingers like a cloud following me around.


----------



## limpidglitch

Semantics maybe, but I'd argue increased bit depth gives increased accuracy, in that the quantized level at any single point in time will likely be closer to the 'real' level.
 Sampling rate is only a matter of band width, so a bit harder to relate to 'accuracy'.
 Dolphins can communicate using sounds in excess of 100kHz. In a sense when we hear them 'speak', it's not really an accurate representation. While the capture might be more accurate sampling at >300kHz, what we hear will still be as inaccurate as it would have been sampling at 44.1kHz.
  
 In the end I'd say yes to both, but as soon as that accuracy surpasses the resolving power of our own sensory system, it's mostly of academic interest.


----------



## Greenears

It's only more accurate in a sound-proof room.  But we do not live in sound proof room, so the question is whether the improved "accuracy" is detectable in the presence of noise.  And the answer is probably not unless it is rock concert loud.
  
 To be clear detectable here means not just your ears, but electrically.  IE if you have a perfectly accurate microphone if 24 bit is transduced to sound waves in presence of noise, and recorded back, 16 vs 24 may not be electrically detectable either.
  
 There is a difference between "accuracy" in an electrical wire and after it is converted to sound pressure waves.


----------



## Soundsgoodtome

limpidglitch said:


> Semantics maybe, but I'd argue increased bit depth gives increased accuracy, in that the quantized level at any single point in time will likely be closer to the 'real' level.
> Sampling rate is only a matter of band width, so a bit harder to relate to 'accuracy'.
> Dolphins can communicate using sounds in excess of 100kHz. In a sense when we hear them 'speak', it's not really an accurate representation. While the capture might be more accurate sampling at >300kHz, the sound we hear is still as inaccurate as it would have been sampling at 44.1kHz.
> 
> In the end I'd say yes to both, but as soon as that accuracy surpasses the resolving power of our own sensory system, it's mostly of academic interest.


 
 Makes a lot of sense! If only we knew more about how human brains work and their ties to other sensory, maybe we can achieve a level where Hi-Res audio files do matter 
	

	
	
		
		

		
			





 We need that dolphin jawbone radar!


----------



## StanD

If I could stop the blood flowing through my ears, how many dB would my DR increase by? I don't think I should breath, it adds to the noise level. Then I've got that damn thumping heart. Oops, the air is moving.
 How crazy are we willing to get to go nowhere? Even what has been stated many times, the circuitry of and around a 24 bit DAC cannot achieve 24 bit DR.
*So what has been learned in this thread about 16/24 bit?* 24 bit or higher is good for processing/crunching numbers and DSP work, not necessary for the final product that we listen to. The difference in the *actual* DR/resolution between 16 and 24 bit hardware is so slight as not to be perceptible, even if we hold our breath. Then there's the part about the quality of masters which is not in the scope of 16/24 bit practical use but instead might be under the heading of GIGO (Garbage In, Garbage Out).


----------



## Soundsgoodtome

This might not be or might be the proper thread but..How does DSD encoding differ? The 1 bit but high sample rate?


----------



## Greenears

soundsgoodtome said:


> This might not be or might be the proper thread but..How does DSD encoding differ? The 1 bit but high sample rate?


 
  
 You're right! It's the wrong thread! 
  
 In short, DSD is inferior and I wish everyone would immediately stop buying any SACD/DSD/DVDA so it would just go away.  But that's just me...
  
 DSD is a 1 bit technology.  1 bit is ancient history.  All leading major ADC/DAC today are multi-segment, which approximates to 2.5 bit.  It's better for technical reasons of noise shaping.  So when people say 24-bit they mean 2.5 bit.  The DAC has 2 physical inputs: 24 bit, and 1 bit DSD.  They both convert through the same engine to 2.5 bits.
  
 So you can see it is redundantly redundant and should just go away.


----------



## Soundsgoodtome

Maybe I'm not technical in terms here but that didn't explain it well enough for me to grasp. ^

Why are companies now seemingly implementing dsd decoding to newer units? A come-back to make something new again and hence newer "improved" models for us to spend on?


----------



## Roly1650

greenears said:


> You're right! It's the wrong thread!
> 
> In short, DSD is inferior and I wish everyone would immediately stop buying any SACD/DSD/DVDA so it would just go away.  But that's just me...
> 
> ...



DVDA is multi bit PCM.

en.m.wikipedia.org/wiki/DVD-AUDIO


----------



## threephi

soundsgoodtome said:


> I apologize if this questions seems elementary. I swear I'm not trolling and this is an honest question. I understand the idea behind 16/44 being "all the human hearing needs" in terms of SPL, studio work aside for lower noise floors,etc for 24-bit, but we're talking about bits and data-rate here to capture and replay electricity.
> 
> Would a 24bit depth and higher frequency rate have more *accuracy* in capturing an analog wave even if just in theory and without applying DAC implentations etc.? This is probably a facepalm question but I'm having one of those Saturdays after a Friday night out and the question lingers like a cloud following me around.


 
 This has been pretty well answered already but I want to add one more thought for emphasis.  As others have pointed out, the improvements in the raw data of the audio signal that comes from increasing bit depth and sample rate above standard-res 16/44.1 are entirely in parts of the signal that are outside the range of human hearing under normal listening conditions.  To put that another way, increasing the bit depth/sample rate above 16/44.1 does not improve sounds within that 16/44.1 data space meaning higher-res is *not* more accurate in the parts we can hear.


----------



## RRod

soundsgoodtome said:


> Maybe I'm not technical in terms here but that didn't explain it well enough for me to grasp. ^
> 
> Why are companies now seemingly implementing dsd decoding to newer units? A come-back to make something new again and hence newer "improved" models for us to spend on?


 
  
 You have to understand noise shaping to talk DSD, as it depends on this technique to move noise outside of the part of the sound spectrum we can hear. Greenears point about DACs is that many of them transform both PCM and DSD into a "common" form (actually much like what DSD uses itself) so that the processing can be handled the same.
  
 Companies are implementing DSD on DAC units because many DAC chips support it (for reasons above), and why not add a feature that will make you unit a selling point to more audiophiles? Online purchasing has allowed DSD to survive, in part by enticing number-hungry audiophiles with things like DSD128/256/512, and so this small amount of musical material available gives a small audience that wants DACs that can handle these new DSD formats.


----------



## Soundsgoodtome

threephi said:


> This has been pretty well answered already but I want to add one more thought for emphasis.  As others have pointed out, the improvements in the raw data of the audio signal that comes from increasing bit depth and sample rate above standard-res 16/44.1 are entirely in parts of the signal that are outside the range of human hearing under normal listening conditions.  To put that another way, increasing the bit depth/sample rate above 16/44.1 does not improve sounds within that 16/44.1 data space meaning higher-res is *not* more accurate in the parts we can hear.


 
  


rrod said:


> You have to understand noise shaping to talk DSD, as it depends on this technique to move noise outside of the part of the sound spectrum we can hear. Greenears point about DACs is that many of them transform both PCM and DSD into a "common" form (actually much like what DSD uses itself) so that the processing can be handled the same.
> 
> Companies are implementing DSD on DAC units because many DAC chips support it (for reasons above), and why not add a feature that will make you unit a selling point to more audiophiles? Online purchasing has allowed DSD to survive, in part by enticing number-hungry audiophiles with things like DSD128/256/512, and so this small amount of musical material available gives a small audience that wants DACs that can handle these new DSD formats.


 


 Thanks gents, good info added to my noggin all in all. So...DSD does not sound better than Redbook is what I'm getting out of the last few posts.


----------



## bigshot

soundsgoodtome said:


> DSD does not sound better than Redbook is what I'm getting out of the last few posts.


 
  
 Redbook and high rate lossy are the same too. MP3 LAME 320, AAC 256 VBR.... same same.


----------



## Soundsgoodtome

Lol I can abx mp3 from flac with accuracy.





bigshot said:


> Redbook and high rate lossy are the same too. MP3 LAME 320, AAC 256 VBR.... same same.


----------



## StanD

soundsgoodtome said:


> Lol I can abx mp3 from flac with accuracy.


 
 I can do that as well, at the time of purchase.


----------



## kraken2109

soundsgoodtome said:


> Lol I can abx mp3 from flac with accuracy.


 
 Can you prove it? A lot of people make this claim but most can't.


----------



## OddE

kraken2109 said:


> Can you prove it? A lot of people make this claim but most can't.




-He didn't specify the bit rate of the mp3.

Heck, even I can tell the difference between flac and a 64kbps mp3 of the same recording...


----------



## Soundsgoodtome

320kbps ogg Vorbis (better than mp3 coding I'm told) vs cd flac. People make the claim sure but I can honestly say I heard it on my HE560 rig, the latter is better and it's more noticeable in the treble. 

How does one prove it with definitive proof over a forum? Just because people say it without proof doesn't automatically mean they're lying, you are entitled to your opinion of course and ymmv yada yada.



kraken2109 said:


> Can you prove it? A lot of people make this claim but most can't.







odde said:


> -He didn't specify the bit rate of the mp3.
> 
> Heck, even I can tell the difference between flac and a 64kbps mp3 of the same recording...


----------



## Head Injury

soundsgoodtome said:


> 320kbps ogg Vorbis (better than mp3 coding I'm told) vs cd flac. People make the claim sure but I can honestly say I heard it on my HE560 rig, the latter is better and it's more noticeable in the treble.
> 
> How does one prove it with definitive proof over a forum? Just because people say it without proof doesn't automatically mean their lying, you are entitled to your opinion of course and ymmv yada yada.


 

 Download foobar2000 if you haven't yet, find and download the ABX Comparator plugin, do 10 or ideally 20 trials comparing the OGG to FLAC, and post your results. For lack of a more controlled test, I'll take your word for it if you pass this with confidence.


----------



## Soundsgoodtome

If I don't get it right I'll post it as well, never used abx but to compare Redbook flac vs hi -res. I have compared 320 Spotify offline to cd rips. I've got all the software but do need to convert some FLAC files to ogg Vorbis. Probably do the abx tomorrow in the a.m.


----------



## U2Bono269

What percentage do you all consider to be passing, or sufficient? Do you have to get it right 100% of the time? Or just a majority?


----------



## RRod

u2bono269 said:


> What percentage do you all consider to be passing, or sufficient? Do you have to get it right 100% of the time? Or just a majority?


 
  
 It all depends on what you want your false positive rate to be. If you are fine with 5% of tests* yielding a false positive, then 9/10 is considered enough. Here are the necessary correct trials (for 5% false-positive rate) shamelessly stolen from Wikipedia:

Number of trials​10​11​12​13​14​15​16​17​18​19​20​21​22​23​24​25​*Minimum number correct*991010111212131314151516161718
  

  
 *Note that a "test" means the entire set of trials you have decided to do beforehand


----------



## Soundsgoodtome

I would think a 66-75% accuracy would be considered sufficient.


----------



## U2Bono269

rrod said:


> It all depends on what you want your false positive rate to be. If you are fine with 5% of tests* yielding a false positive, then 9/10 is considered enough. Here are the necessary correct trials (for 5% false-positive rate) shamelessly stolen from Wikipedia:
> 
> Number of trials​10​11​12​13​14​15​16​17​18​19​20​21​22​23​24​25​*Minimum number correct*991010111212131314151516161718
> 
> ...


 

 I didn't particularly care about this before, but I think I'm going to do this again with more variety. I did it once before, and I ran 2 trials of 10. I scored a 9 and an 8, but I didn't save the logs. I assumed that meant I passed. This was between a FLAC and a 256 AAC.
  
 I'll do some tests with 16 and 24 bit too, because why not. I've got free time this week.


----------



## RRod

u2bono269 said:


> I didn't particularly care about this before, but I think I'm going to do this again with more variety. I did it once before, and I ran 2 trials of 10. I scored a 9 and an 8, but I didn't save the logs. I assumed that meant I passed. This was between a FLAC and a 256 AAC.
> 
> I'll do some tests with 16 and 24 bit too, because why not. I've got free time this week.


 
  
 The 9 is a pass; the 8 isn't (at the 5% level), but there's a lot more to doing a correct ABX test (level matching, etc) that can be automated by software. What are you going to do these tests in?


----------



## U2Bono269

rrod said:


> The 9 is a pass; the 8 isn't (at the 5% level), but there's a lot more to doing a correct ABX test (level matching, etc) that can be automated by software. What are you going to do these tests in?


 

 Foobar with the ABX comparator plugin. Doesn't that automatically level match?


----------



## RRod

u2bono269 said:


> Foobar with the ABX comparator plugin. Doesn't that automatically level match?


 
  
 Yep that should do.


----------



## U2Bono269

That's what I used last time. Very easy.


----------



## Soundsgoodtome

So now that we're derailed I'll ask a question for further derailment. The 9/10 accuracy test and such but wouldn't the music playing also play a rile? I'd presume taking this test would mean using music with quality recordings and perhaps nothing too simple musically.


----------



## bigshot

Complexity of the music isn't the issue. There are just certain kinds of sounds that are more prone to artifact. Transparency is the lack of momentary artifacts. It isn't an overall sound quality. It's that the higher the rate, the fewer artifacts there are, until the artifacts disappear entirely.


----------



## Soundsgoodtome

Ok, that makes sense. Well I'm certainly looking forward to my results. Maybe if the outcome isn't what i think it is I'll be able to free some space on my portable for lossy.


----------



## bigshot

I did the test and determined that my whole library could be lossy.


----------



## sonitus mirus

I think that anyone that honestly believes they can easily hear a difference between well-encoded lossy formats and lossless should be able to tell in a few seconds with a proper ABX 100% of the time, especially when using a type of music that the tester claims is obvious to them.  Why would the tester ever get it wrong?  It's supposed to be obvious, and the listener can easily tell a difference.


----------



## Soundsgoodtome

^I disagree. You make it sound like it should be a glaring difference in sq, I doubt that. You may even need to have several loops of playing the different samples to differentiate the two files being compared.

For the majority of my listening its about enjoyment, critical listens are for testing and usually only when I get new gear or hear something amiss. Not to say high fidelity isn't appreciated (why would head-fi exist). But I haven't ditched Spotify for Tidal nor do i plan to but i can definitely hear the difference there. Could be a matter of Spotify player being a factor. It'll be interesting to test cd ripped ogg vorbis and different since both files are going to be Foobar. I'll entertain having lossy but since I've got massive terabytes for storage, why not keep source files in the purest of forms?


----------



## sonitus mirus

soundsgoodtome said:


> ^I disagree. You make it sound like it should be a glaring difference in sq, I doubt that. You may even need to have several loops of playing the different samples to differentiate the two files being compared.
> 
> For the majority of my listening its about enjoyment, critical listens are for testing and usually only when I get new gear or hear something amiss. Not to say high fidelity isn't appreciated (why would head-fi exist). But I haven't ditched Spotify for Tidal nor do i plan to but *i can definitely hear the difference there*. Could be a matter of Spotify player being a factor. It'll be interesting to test cd ripped ogg vorbis and different since both files are going to be Foobar. I'll entertain having lossy but since I've got massive terabytes for storage, why not keep source files in the purest of forms?


 
  
 You claim that you can definitely hear a difference between what I assume is Spotify's 320kbps Vorbis and Tidal's FLAC.  The word "definitely", to me, seems like it should be every time and nothing less.  This is an easy test to ABX in Foobar.  Go for it.  Let's see some definitive results.


----------



## Soundsgoodtome

sonitus mirus said:


> You claim that you can definitely hear a difference between what I assume is Spotify's 320kbps Vorbis and Tidal's FLAC.  The word "definitely", to me, seems like it should be every time and nothing less.  This is an easy test to ABX in Foobar.  Go for it.  Let's see some definitive results.





Ok, it's certainly my intentions. 

I can hear a definite difference when doing a manual a/b between Spotify and FLAC lossless from a cd rip of the same song and album. I did this by pausing/switching/pausing of tracks with both players open and Foobar on Directsound. Never tried Tidal's service except for their rigged abx test on their site.

So the other gent here that got 9/10 and 8/10 in his abx test, what do you think of that? Think he's lying? Maybe there was some other factor perhaps, say unknown original source FLACs not from his own cd collection.


----------



## bigshot

Take a CD, rip. Use that rip to make a LAME 320 MP3. Do a blind level matched test. Get back to us. (Don't assume two different sources for the same song are the same mastering. They probably aren't.)


----------



## Soundsgoodtome

bigshot said:


> Take a CD, rip. Use that rip to make a LAME 320 MP3. Do a blind level matched test. Get back to us. (Don't assume two different sources for the same song are the same mastering. They probably aren't.)


http://www.head-fi.org/t/655879/setting-up-an-abx-test-simple-guide-to-ripping-tagging-transcoding#post_9268096

This is my plan, although I want to use ogg Vorbis for my lossy. Haven't read it yet but I plan on using the guide above that was PMed to me.

I plan on converting a few tracks from:
Hilary Hahn Plays Bach
Discoveries by Gustavo Dudamel
Too Bright by Perfume Genius 
There's A Time by Doug Macleod
Rumours
maybe some other cds


----------



## RRod

soundsgoodtome said:


> So the other gent here that got 9/10 and 8/10 in his abx test, what do you think of that? Think he's lying? Maybe there was some other factor perhaps, say unknown original source FLACs not from his own cd collection.


 
  
 Could be legit, could have been different sources, could have heard clipping from the mp3/aac compression… There's more to these tests than the final sample ratio, but getting that all worked out on a forum is a messy business that usually pisses someone off ^_^


----------



## Soundsgoodtome

rrod said:


> There's more to these tests than the final sample ratio, but getting that all worked out on a forum is a messy business that usually pisses someone off ^_^




Never understood why there seems to be a high rate of people on head-fi that are so up tight. Like we're all suppose to know it all off the bat.


----------



## U2Bono269

sonitus mirus said:


> You claim that you can definitely hear a difference between what I assume is Spotify's 320kbps Vorbis and Tidal's FLAC.  The word "definitely", to me, seems like it should be every time and nothing less.  This is an easy test to ABX in Foobar.  Go for it.  Let's see some definitive results.


 

 I don't agree with the 100%. No one's perfect, and even experts can make a mistake. 9/10 (or equivalent) is a reasonable compromise with a margin for error. Doing anything 100% perfectly all the time is impossible for us as humans.
  
 The thing is, the differences aren't night-and-day obvious. You have to listen carefully and look for certain things. For me, I can hear the differences in cymbal crashes and delay effects the most. It helps for the recording to have nice dynamic range too...I can't tell the differences between lossy/lossless with "loud" recordings. There's just not enough space to hear the tiny details.


----------



## bigshot

u2bono269 said:


> The thing is, the differences aren't night-and-day obvious. You have to listen carefully and look for certain things.


 
  
 Try AAC 320 VBR.


----------



## U2Bono269

Ok, I will, as soon as I have the free time this week to do it.
  
 Regardless of the outcome, I'm not about to change my collection to aac files. Even if I fail miserably, I'm sticking with lossless. I have the storage space, so there's really no reason for me to switch. I'm quite comfortable with my archival process.
  
 And it will not stop me from picking up 24bit, high resolution copies of my favorite albums as time goes by.


----------



## castleofargh

u2bono269 said:


> Ok, I will, as soon as I have the free time this week to do it.
> 
> Regardless of the outcome, I'm not about to change my collection to aac files. Even if I fail miserably, I'm sticking with lossless. I have the storage space, so there's really no reason for me to switch. I'm quite comfortable with my archival process.
> 
> And it will not stop me from picking up 24bit, high resolution copies of my favorite albums as time goes by.



nobody told you to change anything in your way to listen and store music. tests are for your own information. after, you obviously do whatever you want with that information.
some want storage, some want battery life, some want something that works on any sources... we all have different priorities.


----------



## sonitus mirus

u2bono269 said:


> I don't agree with the 100%. No one's perfect, and even experts can make a mistake. 9/10 (or equivalent) is a reasonable compromise with a margin for error. Doing anything 100% perfectly all the time is impossible for us as humans.
> 
> The thing is, the differences aren't night-and-day obvious. You have to listen carefully and look for certain things. For me, I can hear the differences in cymbal crashes and delay effects the most. It helps for the recording to have nice dynamic range too...I can't tell the differences between lossy/lossless with "loud" recordings. There's just not enough space to hear the tiny details.


 
  
 Yes, but I'm talking about folks that claim it is easy and that there is really no debate on the matter.  In these cases, why would anyone need the right music, the right part in that music, and careful analysis that results in them still missing on occasions?  If the tester is unable to hear a difference while quickly changing from one version to another, the chance of identifying a difference while playing either file in a normal situation is even more unlikely.  I suppose I don't think very subtle differences that few can identify would qualify as easy or obvious.  
  
 When I read these claims, I have a difficult time believing that they have ever done a proper ABX or that their test was somehow flawed.


----------



## U2Bono269

sonitus mirus said:


> Yes, but I'm talking about folks that claim it is easy and that there is really no debate on the matter.  In these cases, why would anyone need the right music, the right part in that music, and careful analysis that results in them still missing on occasions?  If the tester is unable to hear a difference while quickly changing from one version to another, the chance of identifying a difference while playing either file in a normal situation is even more unlikely.  I suppose I don't think very subtle differences that few can identify would qualify as easy or obvious.
> 
> When I read these claims, I have a difficult time believing that they have ever done a proper ABX or that their test was somehow flawed.


 

 Fair point. I don't think it's easy or obvious. The importance of it depends on the application. For me, if I'm listening to music while working, or cleaning, or whatever, then I don't need FLAC files. I'm not listening closely. But sometimes I like to sit in a comfortable chair and do nothing but listen to an album in great detail, then I want to hear the subtle differences. Often, the albums I listen to critically are my favorite ones, and I've heard them hundreds of times. So I'm familiar with the details and can appreciate the differences. In those cases it becomes easier to discern a sharper cymbal crash, or really lush reverb on the vocals. That's a big deal for me.


----------



## bigshot

It doesn't matter what I'm doing, compression artifacts irritate me and stick out like a sore thumb. I want a codec to be audibly transparent, so I don't have to have two copies of every track. AAC 256 VBR is exactly the same as lossless and it is a much smaller file. It works for my home stereo, it works for portable. No compromises at all, except for ones that exist only in theory, not in the real world.


----------



## bigshot

u2bono269 said:


> Regardless of the outcome, I'm not about to change my collection to aac files. Even if I fail miserably, I'm sticking with lossless. I have the storage space, so there's really no reason for me to switch. I'm quite comfortable with my archival process. And it will not stop me from picking up 24bit, high resolution copies of my favorite albums as time goes by.


 
  
 So it really isn't a matter of sound quality, it's just that larger file sizes make you feel better about it.


----------



## U2Bono269

bigshot said:


> So it really isn't a matter of sound quality, it's just that larger file sizes make you feel better about it.


 
  
  
 It's what I'm used to, so it's easier for me to continue doing what I've been doing...don't fix what's not broken. I think it's important to preserve the recording in its most complete released form, which is lossless. AAC may sound the same, but it's not lossless.
  
 And I want to future-proof if at all possible. Who is to say that audio tech won't evolve to the point that it can actually do something worthwhile with the extra data a lossless file has, or the extra data in a high-res file?


----------



## bigshot

In order for you to need something better than audibly transparent, you would need to upgrade the technology of your ears, not your equipment.
  
 The way to improve sound quality is to focus on improving sound quality, not just increasing file sizes and listening to music with frequencies human ears can't hear.


----------



## U2Bono269

bigshot said:


> In order for you to need something better than audibly transparent, you would need to upgrade the technology of your ears, not your equipment.
> 
> The way to improve sound quality is to focus on improving sound quality, not just increasing file sizes and listening to music with frequencies human ears can't hear.


 

 Thanks for your input. I appreciate your concern for my personal music storage needs.


----------



## OrbitingCow

I'll honestly never even touch lossy material. I don't care what ABX you want to drive through my wrists. No, you cannot always tell the difference but on A LOT of stuff you sure as hell can. ABX tests are often a flawed methodology. You really need to sit down with an album and go all the way through with each. And then back to back with critical pieces.
  
 Anyway, my question is what is the deal with Phase problems in 24/96 vs 16/44.1? I hear there are more studies also being done on higher frequencies and if they affect your body rather than just ears. With the phase thing I have heard that people think there are more errors in this way with instruments placing and such with 44.1?
  
 Lastly, my reasoning for always taking the 24/96 or higher stuff is because storage for me is cheap, though I will take the CD quality for burns, and also that I have no idea what person did the downscaling and if they did it properly. With how music is treated today it's hard to trust many people fiddling with all the filters. I'd rather just take 24/96 for piece of mind.
  
 And up there: you honestly trying to tell me to go lossy lol? Hell no dude. Just because your collection is lossy don't mean I'm going to throw my 2TB collection of FLAC in a variety of resolutions and bit depths out the door. That is just insane. 256 VBR IS NOT LOSSLESS. Period.


----------



## U2Bono269

orbitingcow said:


> I'll honestly never even touch lossy material. I don't care what ABX you want to drive through my wrists. No, you cannot always tell the difference but on A LOT of stuff you sure as hell can. ABX tests are often a flawed methodology. You really need to sit down with an album and go all the way through with each. And then back to back with critical pieces.
> 
> Anyway, my question is what is the deal with Phase problems in 24/96 vs 16/44.1? I hear there are more studies also being done on higher frequencies and if they affect your body rather than just ears. With the phase thing I have heard that people think there are more errors in this way with instruments placing and such with 44.1?
> 
> Lastly, my reasoning for always taking the 24/96 or higher stuff is because storage for me is cheap, though I will take the CD quality for burns, and also that I have no idea what person did the downscaling and if they did it properly. With how music is treated today it's hard to trust many people fiddling with all the filters. I'd rather just take 24/96 for piece of mind.


 
 Do you have links to these studies? I've been wondering about that for a while.


----------



## bigshot

Transparent is transparent. Modern lossy codecs achieve audible transparency. You can use a file format that is bigger, but it won't sound better to human ears. I know you don't want to believe that, but it's true.
  
 Studies have shown that your brain might be receiving some sort of stimulus from ultra sonic frequencies (most likely discomfort), but they add absolutely nothing to perceived sound quality of music. Worthless as teats on a bull hog.


----------



## OrbitingCow

Well I have been following up on this debate again. Let me get what I have been reading. I have no idea about some of it and don't make any claims  as to the accuracy. As I said I get 24/96 to rest easy, and even 24/192 sometimes because really I don't care about space. I got the Grateful Dead is pure 24/192. Why some may ask? Because it is there and I have to worry about zero problems with mixing and mastering. It's definitive. Meaning never will be topped as far as I am concerned. One sec.....


----------



## OrbitingCow

bigshot said:


> Transparent is transparent. Modern lossy codecs achieve audible transparency. You can use a file format that is bigger, but it won't sound better to human ears. I know you don't want to believe that, but it's true.
> 
> Studies have shown that your brain might be receiving some sort of stimulus from ultra sonic frequencies (most likely discomfort), but they add absolutely nothing to perceived sound quality of music. Worthless as teats on a bull hog.


 

 I don't believe you. Why? Because now you are trying to tell me that there is science behind this audible transparency that 100% confirms it. I don't believe this.
  
 I do believe that a perfectly assigned downscaling from 24/96 to 16/44.1 is inaudible to my system and ears. But now you are trying to say that an algorithm has perfected itself to transparency on a lossy source. This is not a theorem. It is a theory bud. And these types of theories rarely take into account what we know about phase shifts and frequencies affecting the body and mind.
  
 ABX tests are a seriously flawed methodology. There is no getting around this. I would rather trust science in this case. And science says lossy is different and it is audible in many cases to many people. Period.
  
 When I collect I collect definitively, not on the suggestion that you probably won't hear a difference.


----------



## Greenears

u2bono269 said:


> Ok, I will, as soon as I have the free time this week to do it.
> 
> Regardless of the outcome, I'm not about to change my collection to aac files. Even if I fail miserably, I'm sticking with lossless. I have the storage space, so there's really no reason for me to switch. I'm quite comfortable with my archival process.
> 
> And it will not stop me from picking up 24bit, high resolution copies of my favorite albums as time goes by.


 
  
 I sense another ABX log posting coming!  I rub my hands with glee!
  
 Please post your 24 v 16.  Lossy v lossless is ok as well but I think that should be taken to another thread.  It's a fun argument point but it's not the same question as 24v16 and does not have the same impact for most people (unless most of your collection was never on CD to begin with then you may have a lot of lower bitrate lossy).  Besides, the thread title and OP is not about lossy.


----------



## OrbitingCow

Here was this debate I was reading which was pretty entertaining: http://www.macobserver.com/tmo/article/digital-music-16-bit-44-khz-explained
  
 One thing that should always be taken into account is that a high-res version of anything is almost ALWAYS going to better than its late 80s/early 90s counterpart at 44.1 if they even half-tried to make it decent. We didn't have technology back then to do the downscaling like we do now imo. I'm not saying it is 100% but I have heard some pretty resounding samples. STP's Core album to me sounds a lot better overall in high-res than it does with the old CD for whatever reasons. I'm sure the downscaling there had some issues.


----------



## Don Hills

bigshot said:


> So it really isn't a matter of sound quality, it's just that larger file sizes make you feel better about it.


 
  
 That's a valid reason for doing it.
 I enjoy driving my car more when I've just washed and waxed it, even though it still performs the same and I can't see the difference from inside.


----------



## OrbitingCow

Campbell and the Admin in the mac article up there get into an intense debate about some things. It is worth reading. They also link to some experimental studies on perception of sounds past the normal range and other useful tidbits.
  
 http://jn.physiology.org/content/83/6/3548
  
 "In conclusion, our findings that showed an increase in alpha-EEG potentials, activation of deep-seated brain structures, a correlation between alpha-EEG and rCBF in the thalamus, and a subjective preference toward FRS, give strong evidence supporting the existence of a previously unrecognized response to high-frequency sound beyond the audible range that might be distinct from more usual auditory phenomena. Additional support for this hypothesis could come from future noninvasive measurements of the biochemical markers in the brain such as monoamines or opioid peptides."
  
 http://www.wescottdesign.com/articles/Sampling/sampling.pdf
  
 But usually most think that a perfect conversation from 24/96 to 16/44.1 is fine. That doesn't mean you should trust those doing the converting! And that is why I prefer solid 24/96 or higher. Less human error.
  
 There is also a piece with Bob Ludwig saying it's in the entire piece of the High-res stuff. You need to listen to it as a whole to hear the differences. Botnick of the Doors also said something about feeling better. Make of that what you will.


----------



## RRod

orbitingcow said:


> Well I have been following up on this debate again. Let me get what I have been reading. I have no idea about some of it and don't make any claims  as to the accuracy. As I said I get 24/96 to rest easy, and even 24/192 sometimes because really I don't care about space. I got the Grateful Dead is pure 24/192. Why some may ask? Because it is there and I have to worry about zero problems with mixing and mastering. It's definitive. Meaning never will be topped as far as I am concerned. One sec.....


 
  
 There's still no guarantee that the effort given to a hi-res release is superior, in terms of mixing/mastering, to Redbook releases of the same material. It's still too far much of a crap-shoot to believe that sound quality is the overriding concern in the hi-res movement.
  


orbitingcow said:


> Here was this debate I was reading which was pretty entertaining: http://www.macobserver.com/tmo/article/digital-music-16-bit-44-khz-explained
> 
> One thing that should always be taken into account is that a high-res version of anything is almost ALWAYS going to better than its late 80s/early 90s counterpart at 44.1 if they even half-tried to make it decent. We didn't have technology back then to do the downscaling like we do now imo. I'm not saying it is 100% but I have heard some pretty resounding samples. STP's Core album to me sounds a lot better overall in high-res than it does with the old CD for whatever reasons. I'm sure the downscaling there had some issues.


 
  
 I have plenty of classical albums from that period that sound superb compared with the standards of today. Heck, even some of my *early* 80s releases sound great. It's not a technological issue, it's a matter again of care being put into the actual production process.


----------



## OrbitingCow

Yes, correct, but those old CDs also had terrible scans to begin with in most cases. The converters were not even close to what we have today. Sure, they can sound decent, so can normal CDs from that day. But again, I felt something different on that STP Core release in high-res and I'm sure it is because of the awful tech or human element used to do the downscaling back then. People didn't understand all the stuff about low pass filtering and all the other jazz like they do now, and nor were the DACs and ADCs near as accomplished. So going high-res does have the opportunity to bypass at least a little of the production process. Then again it could be messed up just as well.
  
 Also, I didn't comment on standards of today. I commented on the high-res version of an old CD being better, if it was properly made, because the converters were much worse and human element of error was much higher back then. It is not nearly as bad anymore though I have NO idea how accomplished most mastering and mixing engineers are. Again, another reason to go high-res if possible is to escape the human element, or at least have a chance at doing that.
  
 Here is one of the new threads where people are fighting each other; it is about the perceived audibility of digital errors or something: http://www.hydrogenaud.io/forums/index.php?showtopic=107124&st=1025
  
 Huge thread. And it turns out they may have wanted to sell high-res BS. Again, the best reason to get high-res stuff is because you have a higher chance of someone not ******* things up along the way. IMO anyway.


----------



## RRod

orbitingcow said:


> Yes, correct, but those old CDs also had terrible scans to begin with in most cases. The converters were not even close to what we have today. Sure, they can sound decent, so can normal CDs from that day. But again, I felt something different on that STP Core release in high-res and I'm sure it is because of the awful tech or human element used to do the downscaling back then. People didn't understand all the stuff about low pass filtering and all the other jazz like they do now, and nor were the DACs and ADCs near as accomplished. So going high-res does have the opportunity to bypass at least a little of the production process. Then again it could be messed up just as well.
> 
> Also, I didn't comment on standards of today. I commented on the high-res version of an old CD being better, if it was properly made, because the converters were much worse and human element of error was much higher back then. It is not nearly as bad anymore though I have NO idea how accomplished most mastering and mixing engineers are. Again, another reason to go high-res if possible is to escape the human element, or at least have a chance at doing that.
> 
> ...


 
  
 Technology today of course allows for more leeway in not screwing up a recording and processing, but then you have to consider that the whole Loudness War was due to *deliberate* screwing up of mastering, not an element of human error. So some of us just aren't too jazzed about being resold old albums that shouldn't have been messed up in the first place. Even then, stuff like this happens even in today's hi-res environment:
 http://www.digitalaudioreview.net/2015/02/disrespecting-artistry-becks-morning-phase-as-a-hi-res-download/


----------



## U2Bono269

bigshot said:


> Transparent is transparent. Modern lossy codecs achieve audible transparency. You can use a file format that is bigger, but it won't sound better to human ears. I know you don't want to believe that, but it's true.
> 
> Studies have shown that your brain might be receiving some sort of stimulus from ultra sonic frequencies (most likely discomfort), but they add absolutely nothing to perceived sound quality of music. Worthless as teats on a bull hog.


 

 And what is your harping on me about it going to do? I have stated my intention. It is my business and not yours. Leave it alone.


----------



## U2Bono269

orbitingcow said:


> Campbell and the Admin in the mac article up there get into an intense debate about some things. It is worth reading. They also link to some experimental studies on perception of sounds past the normal range and other useful tidbits.
> 
> http://jn.physiology.org/content/83/6/3548
> 
> ...


 

 Thanks! I'm going to read these later tonight.


----------



## U2Bono269

greenears said:


> I sense another ABX log posting coming!  I rub my hands with glee!
> 
> Please post your 24 v 16.  Lossy v lossless is ok as well but I think that should be taken to another thread.  It's a fun argument point but it's not the same question as 24v16 and does not have the same impact for most people (unless most of your collection was never on CD to begin with then you may have a lot of lower bitrate lossy).  Besides, the thread title and OP is not about lossy.


 

 I think they're cousins, at least. Lossy count as 16bit too, right?


----------



## Avi

bigshot said:


> AAC 256 VBR is exactly the same as lossless and it is a much smaller file. It works for my home stereo, it works for portable. No compromises at all, except for ones that exist only in theory, not in the real world.


 
 I envy your ears; under normal listening conditions I have trouble with 128 AAC. Then again, normal listening conditions means the kids are around and the furnace is runing (the furnace is right next to my study). Perhaps in the summer, when it's quieter


----------



## bigshot

avi said:


> I envy your ears; under normal listening conditions I have trouble with 128 AAC.


 
  
 I said AAC 256 VBR. That is quite different.


----------



## bigshot

orbitingcow said:


> Yes, correct, but those old CDs also had terrible scans to begin with in most cases.


 
  
 No.


----------



## OddE

avi said:


> I envy your ears; under normal listening conditions I have trouble with 128 AAC. _Then again, normal listening conditions means the kids are around and the furnace is runing (the furnace is right next to my study). Perhaps in the summer, when it's quieter _


 
  


bigshot said:


> I said AAC 256 VBR. That is quite different.


 
  
 -I think Avi's point was that under normal circumstances, he had a hard time discerning a difference between lossless and AAC 128 - much less AAC 256 VBR...


----------



## Avi

avi said:


> bigshot said:
> 
> 
> > AAC 256 VBR is exactly the same as lossless and it is a much smaller file. It works for my home stereo, it works for portable. No compromises at all, except for ones that exist only in theory, not in the real world.
> ...


 
  
  


bigshot said:


> avi said:
> 
> 
> > I envy your ears; under normal listening conditions I have trouble with 128 AAC.
> ...


 
  
  


odde said:


> avi said:
> 
> 
> > I envy your ears; under normal listening conditions I have trouble with 128 AAC. _Then again, normal listening conditions means the kids are around and the furnace is runing (the furnace is right next to my study). Perhaps in the summer, when it's quieter _
> ...


 
  
 OddE is correct (AAC 128 VBR -q63 using QAAC).


----------



## Opportunist

Hello, I am a new member but have followed this thread for a while and found it very interesting and inspiring. On the topic of tests to determine whether there are audible differences between 16 and 24 bit technologies, the following statement from a former Chief Technical Officer at Deutsche Grammophon's recording department (now a private company following an MBO and named Emil Berliner Studios - EBS) may be of some interest:
  

2002/2003Much ado about nothing: The experts and parts of the Hi-Fi/High End scene are at cross purposes over the new recording format DSD, on which the Super Audio CD is based, and possible advantages of this format in comparison to PCM, as it is used for CD and (in its high-resolution variety) for DVD-Audio, the rival format of SACD. Whereas the discussion is marred by the use of unsuitable comparisons and untenable marketing slogans, EBS really undertakes to compare the formats. They are the first (and perhaps the only) team worldwide to do so. During the recording of Mahler´s 2nd Symphony (Vienna Philharmonic Orchestra, Gilbert Kaplan, released on Deutsche Grammophon CD 474 380-2, SACD 477 594-2) in the Musikvereinssaal, Vienna, the whole recording sequence is carried out by using both PCM and DSD technology following the microphone. To exclude sound variations by different A/D converters, the team uses special converters capable of dealing with both formats. The result of the subsequent listening comparisons by double-blind test is as straight-forward as sobering: There is no difference whatsoever.


----------



## StanD

opportunist said:


> Hello, I am a new member but have followed this thread for a while and found it very interesting and inspiring. On the topic of tests to determine whether there are audible differences between 16 and 24 bit technologies, the following statement from a former Chief Technical Officer at Deutsche Grammophon's recording department (now a private company following an MBO and named Emil Berliner Studios - EBS) may be of some interest:
> 
> 
> 2002/2003Much ado about nothing: The experts and parts of the Hi-Fi/High End scene are at cross purposes over the new recording format DSD, on which the Super Audio CD is based, and possible advantages of this format in comparison to PCM, as it is used for CD and (in its high-resolution variety) for DVD-Audio, the rival format of SACD. Whereas the discussion is marred by the use of unsuitable comparisons and untenable marketing slogans, EBS really undertakes to compare the formats. They are the first (and perhaps the only) team worldwide to do so. During the recording of Mahler´s 2nd Symphony (Vienna Philharmonic Orchestra, Gilbert Kaplan, released on Deutsche Grammophon CD 474 380-2, SACD 477 594-2) in the Musikvereinssaal, Vienna, the whole recording sequence is carried out by using both PCM and DSD technology following the microphone. To exclude sound variations by different A/D converters, the team uses special converters capable of dealing with both formats. The result of the subsequent listening comparisons by double-blind test is as straight-forward as sobering: There is no difference whatsoever.


 
"Oh brother, someone is gonna *kiss the donkey." *Battleship


----------



## kraken2109

opportunist said:


> Hello, I am a new member but have followed this thread for a while and found it very interesting and inspiring. On the topic of tests to determine whether there are audible differences between 16 and 24 bit technologies, the following statement from a former Chief Technical Officer at Deutsche Grammophon's recording department (now a private company following an MBO and named Emil Berliner Studios - EBS) may be of some interest:
> 
> 
> 2002/2003Much ado about nothing: The experts and parts of the Hi-Fi/High End scene are at cross purposes over the new recording format DSD, on which the Super Audio CD is based, and possible advantages of this format in comparison to PCM, as it is used for CD and (in its high-resolution variety) for DVD-Audio, the rival format of SACD. Whereas the discussion is marred by the use of unsuitable comparisons and untenable marketing slogans, EBS really undertakes to compare the formats. They are the first (and perhaps the only) team worldwide to do so. During the recording of Mahler´s 2nd Symphony (Vienna Philharmonic Orchestra, Gilbert Kaplan, released on Deutsche Grammophon CD 474 380-2, SACD 477 594-2) in the Musikvereinssaal, Vienna, the whole recording sequence is carried out by using both PCM and DSD technology following the microphone. To exclude sound variations by different A/D converters, the team uses special converters capable of dealing with both formats. The result of the subsequent listening comparisons by double-blind test is as straight-forward as sobering: There is no difference whatsoever.


 

 That looks interesting, can you link the source?


----------



## Opportunist

kraken2109 said:


> That looks interesting, can you link the source?


 
  
 Here it is:
  
 http://www.emil-berliner-studios.com/en/chronik5.html
  
 As you will see, the quote comes from a chronicle of Deutsche Grammophon's long history, which I found very interesting (if a bit long-winded).


----------



## sonitus mirus

It seems relevant that the patent royalties that Sony and Philips were reaping from Compact Discs was expiring around the same time that DSD was being heavily pushed to the market.


----------



## Krutsch

sonitus mirus said:


> It seems relevant that the patent royalties that Sony and Philips were reaping from Compact Discs was expiring around the same time that DSD was being heavily pushed to the market.


 
  
 Sure... conspiracy theories aside, the real benefit of SACD is it's support for multi-channel audio.
  
 Of course, multi-channel is also available on other formats, as well (DVD, DVD-A, BD), but the Hybrid SACD can handle all use cases with a single disc and plays everywhere - which is why it has (kinda/sorta) survived in the classical music market.


----------



## OddE

krutsch said:


> Sure... conspiracy theories aside, the real benefit of SACD is it's support for multi-channel audio.


 
  
 -AOL. (Or, as they say today: +1.)


----------



## bigshot

Although it seems that DVD-A and BD-A are replacing it in that.
  
 It appears that multichannel audio is primarily a thing for home theater folks today. Audiophiles are slow to adopt it. But as TV, internet and stereos all merge into one single thing instead of three different things, more and more people will be able to play 5.1 music.


----------



## RRod

krutsch said:


> Sure... conspiracy theories aside, the real benefit of SACD is it's support for multi-channel audio.
> 
> Of course, multi-channel is also available on other formats, as well (DVD, DVD-A, BD), but the Hybrid SACD can handle all use cases with a single disc and plays everywhere - which is why it has (kinda/sorta) survived in the classical music market.


 
  
 Though there are finally some signs of BD gaining ground in the classical world, which means being able to make easy backups and avoiding SACD rigmarole.


----------



## Krutsch

bigshot said:


> Although it seems that *DVD-A and BD-A are replacing it *in that.
> 
> It appears that multichannel audio is primarily a thing for home theater folks today. Audiophiles are slow to adopt it. But as TV, internet and stereos all merge into one single thing instead of three different things, more and more people will be able to play 5.1 music.


 
  
 Yes... I am surprised to see a resurgence of DVD-Audio, which I thought was even more dead than SACD. My last two 5.1 discs, remastered classic rock titles and released in 2014 were both shipped as DVD-A.


----------



## HPiper

krutsch said:


> Yes... I am surprised to see a resurgence of DVD-Audio, which I thought was even more dead than SACD. My last two 5.1 discs, remastered classic rock titles and released in 2014 were both shipped as DVD-A.


 

 True and I found out (the hard way) that my SACD player can not play DVD-A disks so I am either going to have to make sure I don't buy any or get a better SACD player that will also play DVD-A and while I am at it, Blu-ray Audio as well. Very few companies make such a player, Oppo and Marantz do and I think Cambridge Audio makes one but they are kind of hard to find.


----------



## Greenears

What's wrong with multichannel FLAC?  Why do we need anything else?


----------



## Krutsch

hpiper said:


> True and I found out (the hard way) that *my SACD player can not play DVD-A disks* so I am either going to have to make sure I don't buy any or get a better SACD player that will also play DVD-A and while I am at it, Blu-ray Audio as well. Very few companies make such a player, Oppo and Marantz do and I think Cambridge Audio makes one but they are kind of hard to find.


 
  
 Oh, that's OK... DVD-Audio is easy to rip to multi-channel FLAC (see next comment); Google: DVD Audio Extractor.


greenears said:


> What's wrong with multichannel FLAC?  Why do we need anything else?


 
  
 Nothing, and I've ripped all of my personal DVD-A, BD and DVD Video discs to multi-channel FLAC as soon as they arrive from Amazon, et al. I connect my MacBook via HDMI to my AVR, fire-up Audirvana+ and play away.  But, sometimes the discs come with pretty cool art work that plays on the screen during playback and it's nice to just to pop in a disc, versus hauling my MacBook out.
  
 To be honest, I've been spinning more discs lately because of that last reason.
  
 This is why the whole 16 vs. 24 bit thing is annoying for me, because it's a distraction from things that add real value to the playback experience, like well-mastered multi-channel audio. When its done well, it's incredible.


----------



## RRod

krutsch said:


> Oh, that's OK... DVD-Audio is easy to rip to multi-channel FLAC (see next comment); Google: DVD Audio Extractor.
> 
> Nothing, and I've ripped all of my personal DVD-A, BD and DVD Video discs to multi-channel FLAC as soon as they arrive from Amazon, et al. I connect my MacBook via HDMI to my AVR, fire-up Audirvana+ and play away.  But, sometimes the discs come with pretty cool art work that plays on the screen during playback and it's nice to just to pop in a disc, versus hauling my MacBook out.
> 
> ...


 
  
 What are you using to rip DVD-A (hardware-wise)?


----------



## Krutsch

rrod said:


> What are you using to rip DVD-A (hardware-wise)?


 

 LG Super Multi Blue - it's very fast and rips are robust.


----------



## Soundsgoodtome

So I found out dbpoweramp needs to be purchased to convert flac to mp3 after trial expires so I needed more time to get the CDs out of storage and ripped to mp3 via EAC.

 Then this came in today, Steven Wilson's latest album *HAND. CANNOT. ERASE.* which includes a download in 16/44 both FLAC and MP3!


 And so after I get a pattern down for a proper abx technique i was able to get an accurate hearing of wether a/b=x or if a/b=y. I will admit it wasn't easy and took several loops of the same 5-15sec segments but here's my result with my secondary chain of: Foobar ABX <cheap usb cable> Vlink 24/96 <spdif rca cable> Audio-gd NFB11 >> Fidue A83 triple hybrid iem. My He560 at home is a better rig and I'd expect the same results.

 A bit blurry so 9/10 with 1.1% chance of guessing. ABX is not half as fun as actually listening to albums. Totally a snooze fest that requires concentration but yes I'll take my FLAC lossless anytime over lossy, even on my less than great Samsung Galaxy S3. YMMV



Tags: @kraken2109 @bigshot @sonitus mirus  @Music Alchemist @RRod @Stillhart  @AxelCloris @Ivabign


----------



## Stillhart

Yeah, but how is the album?  You mention "snoozefest" as if Wilson's solo stuff isn't already a snoozefest...
  
 (Only somewhat in jest... Porcupine Tree is one of my favorite bands, but I can't get into Wilson's solo stuff.)


----------



## castleofargh

are you sure the files are all from the same master and at the same loudness?
I know it sounds dumb, but they can put totally different masters for each file format. dunno if it's for rights troubles or just to fool people but they sometimes do it.


----------



## Soundsgoodtome

Haven't gotten into it, really looking forward to the 5.1 aspect of the album. Wilson is known to be one of the few that gets it "right"? Saw that it came with the digital download and the BD was only 4 bucks more so why not. My first surround album.

What's more important I think is that Wilson's albums gets a nod from the audiophiles. This album is no different in that aspect. As to wether I like the album or not, I'd be lying if I just didn't get into his stuff. Wouldn't call it boring, I was calling the act of abx-ing clips boring.

Reply to your post below Stillhart
Well color me boring, I like his solo albums. I did get into his solo stuff at the same time of Porcupine Tree though. Plus I only have Deadwing from PT..

My cheap Sony 5.1 bluray player at home isn't the most ideal but at least I'll be able to get a feel of a BD-A multi channel sound. Interesting you mention emulator, wonder how it sounds through that gaming headphone surround doohickey


----------



## Stillhart

soundsgoodtome said:


> Haven't gotten into it, really looking forward to the 5.1 aspect of the album. Wilson is known to be one of the few that gets it "right"? Saw that it came with the digital download and the BD was only 4 bucks more so why not. My first surround album.
> 
> What's not important I think is that Wilson's albums gets a nod from the audiophiles. This album is no different in that aspect. As to wether I like the album or not, I'd be lying if I just didn't get into his stuff. Wouldn't call it boring, I was calling the act of abx-ing clips boring.


 
  
 I have a couple of PT's 5.1 albums and they're SUPERB on a real 5.1 setup.  I've never tried them with virtual surround.  I'm really eager to get some of his remasters of the classic Yes albums.  He's quite good at mastering the PT stuff.
  
 And yes, I get that you were calling ABX boring.  But his solo stuff is also boring so it shouldn't be any different.


----------



## Soundsgoodtome

castleofargh said:


> are you sure the files are all from the same master and at the same loudness?
> I know it sounds dumb, but they can put totally different masters for each file format. dunno if it's for rights troubles or just to fool people but they sometimes do it.




With the low noise floor of where I'm at and having iems, as far as I can tell yes it is level matched. I have no solid proof but my ears which seems to be ok to hearing very slight differences would say yes. Doesn't the abx plugin work that out?


----------



## bigshot

I got angry at the last two 5.1 remixes I got by Steven Wilson. XTC's "Drums and Wires" sounded like a prog rock album by the time he got done with it. The new wave edge was totally shaved off. And Jethro Tull's "War Child" had mixes that left out important stuff like doubling vocals. In fact, the vocals were mixed exactly the same in every song. I like the old quad mix better. He is on strike two for me.
  
 (XTC's "Nonsuch" sucked big time too, but I blame that on the band, not the mix.)


----------



## Stillhart

bigshot said:


> I got angry at the last two 5.1 remixes I got by Steven Wilson. XTC's "Drums and Wires" sounded like a prog rock album by the time he got done with it. The new wave edge was totally shaved off. And Jethro Tull's "War Child" had mixes that left out important stuff like doubling vocals. In fact, the vocals were mixed exactly the same in every song. I like the old quad mix better. He is on strike two for me.
> 
> (XTC's "Nonsuch" sucked big time too, but I blame that on the band, not the mix.)


 
  
 The Yes albums are mastered pretty poorly as it stands.  I'm not sure how much he could mess them up.
  
@Soundsgoodtome , you really need to listen to "In Absentia" and "Stupid Dream" if you like "Deadwing".  "Lightbulb Sun" is also in the same vein tho it's not quite as polished as their later stuff.  "Arriving Somewhere But Not Here" and later are all good too, but I think the first three I mentioned are the solid core.
  
 Anyways, sorry for the OT posts.  I'll stop now.  lol
  
 On topic:  I'm kind of annoyed that there are people selling HD stuff for so much more than CD quality.  I was, quite frankly, shocked when I saw the price of the HD Led Zeppelin albums.  I don't know how they expect HD to catch on with pricing like that.  Even if it does sound better (which we know is arguable), it doesn't sound THAT much better.


----------



## Krutsch

stillhart said:


> The Yes albums are mastered pretty poorly as it stands.  I'm not sure how much he could mess them up.


 
  
 I thought the SW multi-channel mix of "Close to the Edge" was spectacular - not many albums I've listened to more times than that one and I just loved it.


----------



## Stillhart

krutsch said:


> I thought the SW multi-channel mix of "Close to the Edge" was spectacular - not many albums I've listened to more times than that one and I just loved it.


 
  
 "Close to the Edge" and "The Yes Album" are on my short list.  "Fragile" too if he did that one.  I'm just having trouble validating spending that much on these albums.  I've spent a lot of money on Yes over the years (including concerts and tees) so spending additional money on an album I already own on LP and CD is... tough.  But first-hand reviews from audiophiles saying it's "spectacular" go a long way towards helping my decision!  Thx!


----------



## bigshot

His remixes would be a lot better if he wasn't remixing Yes.


----------



## lamode

u2bono269 said:


> I hesitate to bring this up again, but I found an article that's a little confusing and I think this is still the best place to ask it.
> 
> I found an article about Hesitation Marks, Nine Inch Nails last album. Here's the link:
> 
> ...


 
  
 Sorry but you seem to be going around in circles.
  
 There is NO advantage to HD, ever (as a consumer format). Nothing, zip, nada, nichts, ничего, 沒什麼...


----------



## castleofargh

no audible advantage for the end user at least.


----------



## lamode

orbitingcow said:


> ABX tests are a seriously flawed methodology.


 
  
 You keep saying that, without any evidence. What was it you were saying about science again?


----------



## lamode

orbitingcow said:


> Here was this debate I was reading which was pretty entertaining: http://www.macobserver.com/tmo/article/digital-music-16-bit-44-khz-explained
> 
> One thing that should always be taken into account is that a high-res version of anything is almost ALWAYS going to better than its late 80s/early 90s counterpart at 44.1 if they even half-tried to make it decent. We didn't have technology back then to do the downscaling like we do now imo. I'm not saying it is 100% but I have heard some pretty resounding samples. STP's Core album to me sounds a lot better overall in high-res than it does with the old CD for whatever reasons. I'm sure the downscaling there had some issues.


 
  
 You have mentioned 'downscaling' a few times, but again this is basically inaudible. Try comparing a native 24 bit recording with a truncated 16 bit version in an ABX test and see how you go. You won't hear a difference. Yes, we have some more advanced dithering techniques now compared to in the 80s but all that does is drive down the noise floor even lower.
  
 As for 24/96 to 16/44 conversion, that wasn't an issue in the 80s anyway.


----------



## lamode

opportunist said:


> 2002/2003During the recording of Mahler´s 2nd Symphony (Vienna Philharmonic Orchestra, Gilbert Kaplan, released on Deutsche Grammophon CD 474 380-2, SACD 477 594-2) in the Musikvereinssaal, Vienna, the whole recording sequence is carried out by using both PCM and DSD technology following the microphone. To exclude sound variations by different A/D converters, the team uses special converters capable of dealing with both formats. The result of the subsequent listening comparisons by double-blind test is as straight-forward as sobering: There is no difference whatsoever.


 
  
 Which many of us have been saying for a long time.
  
 However, there are technical disadvantages to DSD (such as incompatibility with DSP) which makes me wish this format would die ASAP.


----------



## kraken2109

I find Porcupine Tree and Steven Wilson's solo stuff are all produced really well. I haven't listened to his remasters of other bands though. The SW and PT surround mixes are great though.


----------



## analogsurviver

lamode said:


> Which many of us have been saying for a long time.
> 
> However, there are technical disadvantages to DSD (such as incompatibility with DSP) which makes me wish this format would die ASAP.


 
 I like DSD particularly for the fact that it can not be mangled with PCM>DSP every time some computer geek would like to sell yet another DSP software of one kind or another.
  
 IF you personally can fathom the thought of making a recording and playing it back without any computer (except the recorder)  within miles, then you might, just perhaps might start hearing the advantages of DSD over PCM. With live mike feed as a reference. And not remasters of masters done by third and fourth party, at and during which you most definitely were never present.
  
 If you have to PCM>ABX>DSP everything before it ever reaches your ears, that is the same as saying decent (not necessary expensive) restaurants should die ASAP because they do not conform to the "standards" of McDonald's. Which is to say the lowest possible common denominator.
  
 I do try to not use the word "you" as much as possible. In this post, it is very much intentional and adressed to you personally - because you are doing anything possible to tar and feather DSD, most probably before you heard it done right and properly demoed vs properly done PCM.
  
 I am not familiar with the Mahler/Kaplan recording on DG - because DG is NOT the record label generally known for sound quality. Its forte are good to excellent musical performances of mainly core classical repertoire. But I do have Kaplan's first effort recorded by Tony Faulkner.  There is a reason WHY the recordings from DG reissued on vinyl are almost invisible in number compared to Decca, RCA, Mercury, Westminster, etc - and fast forward to present day, DG sound has not changed much - in some cases, it has become even worse. Utilizing the advantages of which DSD is capable of does require more understanding of the recording process than the usual DG issues are displaying. Having said this, PCM vs DSD test, if carried out by DG in their usual ways, bears very little weight for me. By the time signal hit the recorders, either PCM or DSD, it was all over.


----------



## RRod

analogsurviver said:


> I like DSD particularly for the fact that it can not be mangled with PCM>DSP every time some computer geek would like to sell yet another DSP software of one kind or another.
> 
> IF you personally can fathom the thought of making a recording and playing it back without any computer (except the recorder)  within miles, then you might, just perhaps might start hearing the advantages of DSD over PCM. With live mike feed as a reference. And not remasters of masters done by third and fourth party, at and during which you most definitely were never present.


 
  
 [redacted] Oh nm, it's you analog


----------



## analogsurviver

rrod said:


> [redacted] Oh nm, it's you analog


 
 Even if I nm (redacted), RRod please try to go trough the entire post and, if required, read in between the lines 
	

	
	
		
		

		
		
	


	




.


----------



## RRod

analogsurviver said:


> Even if I nm (redacted), RRod please try to go trough the entire post and, if required, read in between the lines
> 
> 
> 
> ...


 
  
 More I didn't want to rehash things when I know well your position. I agree with you on the DG sound, btw, though there's too much good material to avoid.


----------



## analogsurviver

rrod said:


> More I didn't want to rehash things when I know well your position. I agree with you on the DG sound, btw, though there's too much good material to avoid.


 
 No problem - I share the same opinion.
  
 I never meant to imply to avoid DG altogether - music comes first. I only wanted to stress the point it simply is not a benchmark in SQ. Under the current umbrella of Universal, on average it is the worst of the three. No contest.
  
 Perhaps the best, in terms of SQ, from the current Universal catalog, are the last Philips recordings done in analog. It was a memorable swan song !


----------



## sonitus mirus

soundsgoodtome said:


> So I found out dbpoweramp needs to be purchased to convert flac to mp3 after trial expires so I needed more time to get the CDs out of storage and ripped to mp3 via EAC.
> 
> Then this came in today, Steven Wilson's latest album *HAND. CANNOT. ERASE.* which includes a download in 16/44 both FLAC and MP3!
> 
> ...


 
  
 So you weren't able to convert the file and ensure that it was encoded properly, you simply downloaded 2 separate files?  You don't even have the necessary tools to convert a file or rip a CD to a properly encoded lossy format?  Were the 2 files you tested even volume matched?  Were you allowing the spectrogram visualizer to run during the testing?  
  
 If it was me, I would certainly want to investigate a bit more thoroughly. I'm able to identify a difference between the Tidal music's FLAC and AAC 320 test every time 15/15 in under 3 minutes, but it was discovered that there was a subtle EQ being applied to one of the formats that seemed out of place and did not occur when I encoded the same music to the same formats with EAC or dBPoweramp.  Maybe the files are different due to an inferior encoding process?  
  
 Unless your only goal is to show others that you can hear a difference, there are a lot of issues to address with the methodology of your test.  If you really want to know for yourself if you can hear a difference, there is some more research required to be more certain.


----------



## Soundsgoodtome

Lol more hoops huh? I'll re-rip a CD in mp3 and recheck. I don't know how to run the spectogram during an abx so i wouldn't know. However a flac v mp3 abx from the same cd should make this irrelevant. 

But I'm guessing the mp3 and flac rip from Kscope is the same master, I can email them to verify.

There's a sound difference and to me that's all I need to know I'm sticking with Flac.


sonitus mirus said:


> So you weren't able to convert the file and ensure that it was encoded properly, you simply downloaded 2 separate files?  You don't even have the necessary tools to convert a file or rip a CD to a properly encoded lossy format?  Were the 2 files you tested even volume matched?  Were you allowing the spectrogram visualizer to run during the testing?
> 
> If it was me, I would certainly want to investigate a bit more thoroughly. I'm able to identify a difference between the Tidal music's FLAC and AAC 320 test every time 15/15 in under 3 minutes, but it was discovered that there was a subtle EQ being applied to one of the formats that seemed out of place and did not occur when I encoded the same music to the same formats with EAC or dBPoweramp.  Maybe the files are different due to an inferior encoding process?
> 
> Unless your only goal is to show others that you can hear a difference, there are a lot of issues to address with the methodology of your test.  If you really want to know for yourself if you can hear a difference, there is some more research required to be more certain.


----------



## RRod

rrod said:


> There's more to these tests than the final sample ratio, but getting that all worked out on a forum is a messy business that usually pisses someone off ^_^


 
  


soundsgoodtome said:


> Lol more hoops huh? I'll re-rip a CD in mp3 and recheck. I don't know how to run the spectogram during an abx so i wouldn't know. However a flac v mp3 abx from the same cd should make this irrelevant.
> 
> But I'm guessing the mp3 and flac rip from Kscope is the same master, I can email them to verify.


 
  
 See what I mean with my quote above


----------



## Soundsgoodtome

Yes. I'll say I'm just verifying for my own experience and everyone's mileage will vary along with tolerance of difference. If I were to simply enjoy music I think mp3 would be more than enough, however I want all the data audible or not in my files. Also, if we weren't spending so much on audio gear I'd go MP3 but since people pay hundreds for an extra detail in sound, why short yourself in the data? 





rrod said:


> See what I mean with my quote above


----------



## sonitus mirus

I'm not mad at anything or anyone.  It was just that your comments suggested that you had very little experience converting a file or CD to a lossy format or with the ABX tools.  There were simply too many unanswered questions from what I read to indicate that the test was properly conducted.   I included a recent example where I had no control over the test files created, and in the end it was discovered that the files were actually different where there was no rational reason for there to have been any difference.
  
 I'd be more curious about my results and would want to investigate further, but that is my personality.  I am a natural skeptic.
  
 I seriously considered purchasing this music so that I could analyze the downloads that you tested.  So, I certainly have issues.


----------



## Soundsgoodtome

It's a really good album. Buy the BD-A and use the one-time code provided (free) to get both mp3 and flac in a single zip file.

The CD might have the same codes too but I'd verify before buying.


----------



## lamode

analogsurviver said:


> I like DSD particularly for the fact that it can not be mangled with PCM>DSP every time some computer geek would like to sell yet another DSP software of one kind or another.
> 
> IF you personally can fathom the thought of making a recording and playing it back without any computer (except the recorder)  within miles, then you might, just perhaps might start hearing the advantages of DSD over PCM. With live mike feed as a reference. And not remasters of masters done by third and fourth party, at and during which you most definitely were never present.


 
  
 Sorry, but this seems like willful ignorance. I don't know of a single headphone or loudspeaker (even high end) which would not benefit from digital EQ. And yes, that EQ will bring you closer to the live feed you mentioned. That is just one of several reasons why I am only interested in a computer-based digital audio system.
  
 As for DSD releases, they are mostly captured and mixed using PCM anyway, and the final result is transcoded to DSD, a process which just introduces distortion and noise (not that it will be audible in the real world though). Still, it is always better to get the PCM release if there is a choice.


----------



## bigshot

soundsgoodtome said:


> Lol more hoops huh? I'll re-rip a CD in mp3 and recheck.


 
  
 AAC 320, not MP3


----------



## bigshot

soundsgoodtome said:


> If I were to simply enjoy music I think mp3 would be more than enough, however I want all the data audible or not in my files.


 
  
 That's fine, but you are saying you can hear what has always been inaudible for everyone else. If you just want nice big chunky file, save everything as WAV and fill up hard drives like crazy. If you don't care if lossy sounds exactly the same or not, there's no reason to take the test.


----------



## analogsurviver

lamode said:


> Sorry, but this seems like willful ignorance. I don't know of a single headphone or loudspeaker (even high end) which would not benefit from digital EQ. And yes, that EQ will bring you closer to the live feed you mentioned. That is just one of several reasons why I am only interested in a computer-based digital audio system.
> 
> As for DSD releases, they are mostly captured and mixed using PCM anyway, and the final result is transcoded to DSD, a process which just introduces distortion and noise (not that it will be audible in the real world though). Still, it is always better to get the PCM release if there is a choice.


 
 Sorry, if the headphone or speaker/room requires more than a decent analog parametric EQ, it should be optimized differently. There is a reason why manufacturers strive for inherently as flat response as possible. 
  
 I agree that recording PCM and then bounce it to DSD does not present the proper way to do it. Recording DSD and editing with the least invasive PCM tools only at and around the edit point yields approx 90% pure DSD that can be provided to the end user. That means those 90% are pure DSD that never was in any form of PCM. Provided musicians are really up to the task and allow minor errors in playing to be issued, that percentage can reach 100%.
  
 The inability to edit in pure DSD is to me an advantage; it is essentially direct to XYZ process, giving an honest account of the musicians' capabilities. It is the PCM fix/correct/master to 100 % note by note perfection we grew so accustomed with and spoiled by that is creating this problem in the first place. One could rely on the recording to be a true sound of musician(s) far better before PCM started to change things - sometimes to unrecognizability. 
  
 With today's tools (PCM, DSP ), my howling or piano banging may well win a Grammy - if I find a competently crazy enough recording engineer and/or computer geek and can afford his/hers fee.


----------



## lamode

analogsurviver said:


> Sorry, if the headphone or speaker/room requires more than a decent analog parametric EQ, it should be optimized differently.


 
  
 Why should I use analogue EQ when digital is so much better?
  


> > Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> >
> > Recording DSD and editing with the least invasive PCM tools only at and around the edit point yields approx 90% pure DSD that can be provided to the end user. That means those 90% are pure DSD that never was in any form of PCM. Provided musicians are really up to the task and allow minor errors in playing to be issued, that percentage can reach 100%.
> 
> ...





> > Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> > The inability to edit in pure DSD is to me an advantage;


 But impractical for almost all modern recordings, so basically irrelevant. Post tools can be abused but they are also very useful and advantageous in the right hands. Somewhat similar to Photoshop and photography.
  
 At the end of the day, artists want to mix, and use multiple takes, so PCM will be necessary and DSD becomes irrelevant. DSD is difficult to use and offers no advantage, only disadvantages.


----------



## analogsurviver

lamode said:


> Why should I use analogue EQ when digital is so much better?
> 
> 
> 
> ...


 
 Why do you think analog EQs offer equalizing WAY above 20 kHz - with center frequency as high as 50 kHz ?
  
 Why do you think I avoid more than two microphones and mixing - like a pleague ?
  
 It can be recorded in several takes and DSD changed to PCM for editing only around/at edit mark.
  
 Post tools never equal a well captured natural sound. "Photoshoped" music is precisely as unatural as photoshoped pictures - in real life, it does not exist.


----------



## bigshot

lamode said:


> Why should I use analogue EQ when digital is so much better?


 
  
 Analogue equalizers are less precise, with lots of spill between bands and tend to drift, so you have to keep adjusting them. Digital EQ is the way to go.


----------



## RRod

analogsurviver said:


> It can be recorded in several takes and DSD changed to PCM for editing only around/at edit mark.
> 
> Post tools never equal a well captured natural sound. "Photoshoped" music is precisely as unatural as photoshoped pictures - in real life, it does not exist.


 
  
 So you favor an "abstinence plus" approach? I still don't see why unmodified hi-res PCM would be inferior to DSD. It's not like one is *required* to use any DSP tools just because he recorded in PCM.


----------



## lamode

analogsurviver said:


> Why do you think analog EQs offer equalizing WAY above 20 kHz - with center frequency as high as 50 kHz ?


 

 Too get rid of ultrasonic noise! Glad you asked 
  

  
 (from "Introduction to Live Sound Reinforcement: The Science, the Art, and the Practice" By Teddy Boyce, p. 132)
  


> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> Why do you think I avoid more than two microphones and mixing - like a pleague ?


 
  
 You might avoid them, but >99% of recordings don't.
  


> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> It can be recorded in several takes and DSD changed to PCM for editing only around/at edit mark.


 
  
 ...which contradicts your earlier statement about the recordings being pure DSD. Converted DSD can only be worse than the PCM mix.
  


> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> Post tools never equal a well captured natural sound.


 
  
 Nonsense. If properly used, post tools can improve the sound (EQ, reverb, etc) and of course artists should be free to process sound in any way they see fit. It is THEIR vision.


----------



## headwhacker

> Originally Posted by *RRod *
> 
> 
> Quote:
> ...


 
  
 Exactly, the existense of post processing tools in PCM doesn't force anyone to use it.
  
@analogsurviver Why would music recorded in PCM be inferior to DSD if we take out post processing into equation?


----------



## analogsurviver

lamode said:


> Too get rid of ultrasonic noise! Glad you asked
> 
> 
> 
> ...


 
 I will only comment on the post tools today. 
  
 Musicians (remember, I am talking about acoustic music ) CAN NOT hear how a person in the audience does. 
 Even the conductor does NOT hear it anywhere like a listener even in the first row - which is still thought to be too close by most listeners. So a violin player will invariably say he/she is way too quiet in the "mix" - of course, with a violin say 20 cm from his/hers nose, anything else becomes background - and pretty much everybody playing a different instrument will tell you the same/similar. 
  
 This is the fundamental difference - I try to capture the sound as perceived by the LISTENER IN THE BEST POSSIBLE SEAT. And will try to get the best possible venue within the practical limitations. NO post mumbo jumbo.


----------



## lamode

analogsurviver said:


> I will only comment on the post tools today.


 
  
 Well you can ignore my points if you wish, but I hope you realize how incorrect and futile your stance is.


----------



## analogsurviver

rrod said:


> So you favor and "abstinence plus" approach? I still don't see why unmodified hi-res PCM would be inferior to DSD. It's not like one is *required* to use any DSP tools just because he recorded in PCM.


 
 I wrote many times recording DSD and PCM does not make sense with multimiking/mixing.
  
 If the minuscule advantages offered by the DSD over approximately same size file PCM is to be taken advantage of, one has to make sure any time related distortions in the signal prior it reaches the input of the recorder are kept to a minimum. 
  
 Once this criterion is met, DSD will show its superiority over PCM. Time errors with multimiking are so large compared to the advantages of DSD over PCM that they make these small advantages  effectively without any consequence. 
  
 That is why I say that transfering old(er) recordings with less than stellar miking as to showcase the DSD is simply wrong. It will only reveal the shortcomings of the original recording better - possibly making it  sounding worse.
  
 The acid test for any recording is what I have coined binaural natural. That is to say headworn mics. Properly done, this allows one to listen to the concert while recording it at the same time - and this is as close as you will ever get to real sound if this binaural recording is reproduced using a good set of earspeakers - preferably AKG K 1000.  Here DSD will outperform PCM - via listening to the recording made to two parallel recordings, one made to DSD recorder and other to PCM recorder.
  
 NO mic , mixing desk, DSP, etc limitations - just DSD vs PCM. There is no monitoring possible - at least for the "unartificial artificial head" - but second person could listen while third person switching between the level matched recorders AND live mike feed - ABCX if you wish. 
  
 I think this should also answer the question of the @headwhacker.


----------



## analogsurviver

lamode said:


> Well you can ignore my points if you wish, but I hope you realize how incorrect and futile your stance is.


 
 The first quote was for the live sound reinforcement - which in itself is yet another possibility for the sound engineer to create an artificial sound that does not exist in real life. Be it for rock band (acceptable) or for a symphonic orchestra playing outdoors or BIG "mixed purpose" hall also used for concerts (unacceptable) - which usually creates such poor sound I endure these only if absolutely unavoidable for one reason or another. I would NEVER pay to hear such a concert - period. The last one was a concert by Bocelli - which only strengthened this belief. 
  
 Now fast forward to real acoustic thing in a real acoustic space - go and listen to this - and the same musicians outdoors with mikes and speakers. The second will NEVER even approach the real thing - and the errors in the sound heard live are too great to warrant even the use of the CD redbook, let alone justifying the difference DSD to PCM.
  
 Real acoustic thing in a real acoustic space is quite another matter. And it is the reference we should be trying to capture. 
  
 Anything you do to the sound as captured by the carefully positioned stereo mike in post will be *detrimental *to the realistic SQ. In other genres it is acceptable, in acoustic music it creates something that does not exist in real life and sounds plain unnatural - but unfortunately it represents some 99% of all available recordings.
  
 In other words - recording a piano in a relatively small studio necessitating closer than optimum miking and adding reverb etc with sampling of acoustics of famous halls will never be the equal of the piano recorded in that famous concert hall. Due to economical constraints, it is often being done - but that does not mean it is good, let alone the best. Regardless how much DSP is involved.


----------



## Greenears

Hello - anyone out there got anything to say about 24b vs 16b? Yes that is PCM, usually embodied as FLAC.


----------



## castleofargh

staying on topic is so 2014.  
the photoshop comparison works well for once. when did the ability to do more become a bad thing? sure some will abuse everything, but that's mostly what noobs do. when pro abuse stuff they suck or were told to.

analogsurv if you really are into minimalistic interventions, why not go all the way and only do binaural records? that way at least you have a real reason to touch nothing.


----------



## bigshot

If someone wants to comment on the music production process, it helps to have actually worked in it on a professional level. Just sayin'...


----------



## limpidglitch

analogsurviver said:


> I try to capture the sound as perceived by the LISTENER IN THE BEST POSSIBLE SEAT.


 

 Which makes sense if you plan for your capture to be played back on headphones. As Castle mentions, why not just go full binaural?


----------



## analogsurviver

limpidglitch said:


> Which makes sense if you plan for your capture to be played back on headphones. As Castle mentions, why not just go full binaural?


 
 There are MANY reasons why not going binaural only. Although it is the best way to go - nothing else comes even close in realism. Specially if headphones are supported by the addition of subwoofer in order to provide tactile bass.
  
 There are persons who would rather stop listening to reproduced music if  the only option would be binaural over headphones. With these people, sales potential of binaural is exactly nil. It is speakers - or nothing.
  
 There are people, mostly performers, who prefer somewhat "enhanced" version of miking meant for loudspeakers. And would choose it over truth but nothing but the truth any day in a week.
  
 I will not beat around the bush - binaural will really knock your socks off with one "headphone" only  - AKG K 1000. Discontinued and ever harder to get - around 13-14 thousands of pairs ever produced.
  
 IF there is enough time and the main recording meant to be listened over speakers does not get compromised, I will simultaneously record binaural. But it is, in financial terms, strictly hobby on my part so far. With one single exception. It means mastering has to be done twice - which drives the costs beyond the capabilities and willingness of most customers.
  
 I have to eat - clear enough ?


----------



## analogsurviver

bigshot said:


> If someone wants to comment on the music production process, it helps to have actually worked in it on a professional level. Just sayin'...


 
 IIRC, you said who in the world would put 10 amplifiers in a chain.
  
 Now, please go and see any schematics for a mixing desk. As a starter.
  
 Then repeat that for a recorder.
  
 These two above are more than enough to prevent you to ever hear the true sound of the source.
  
 IIRC, you said you prefer the dynamic range limited to 40-50 dB. That necessitates the use of a hardware compressor or some software in DAW. And are both way worse than just an additional amplifier stage in the chain.
  
 All of the above will mangle the signal enough for the CD redbook to be "transparent" .
  
 And I did not touch the subject of the microphones at all - and I never will.
  
 Because 99% of the pro gear not only is useless, it is detrimental to the SQ. 
  
 That is why I will never go ta a "professional level sound school" - as it has, over the years, evolved in better photoshop than Photoshop itself.  Creating ever more gear, glossing over the inherent deficiences. Thank you, but NO - thank you.
  
 Somewhat inspirational : http://www.stereophile.com/content/keith-o-johnson-reference-recordings


----------



## analogsurviver

castleofargh said:


> staying on topic is so 2014.
> the photoshop comparison works well for once. when did the ability to do more become a bad thing? sure some will abuse everything, but that's mostly what noobs do. when pro abuse stuff they suck or were told to.
> 
> analogsurv if you really are into minimalistic interventions, why not go all the way and only do binaural records? that way at least you have a real reason to touch nothing.


 
 The ability to do more _*falsification *_was never good in my book. Pros with the capability WILL abuse stuff - if they are told so. They have to eat, too. Far more dangerous is the temptation "whether I should TRY effectthiseffectthat" on the next project - why that "effectthiseffectthat" go to waste - if you already had to pay for it in order to get what you wanted in the first place and was not available on any model short of the TOTL - graced with every bell and whistle, imaginable or not.
  
 In the vinyl world, Mobile Fidelity Sound Lab ( MFSL for short ) is hailed as consistently churning out the records sounding better than originals. At least one of the engineers involved admitted he/they have been told to use EQ - more bass, more sizzle - because the owner of the label at the time knew such records would sell better. 
  
 And they did - but that does not make them true to the source.
  
 One big advantage of the MFSL pressings during a certain period was the use of the best ever pressing plant - JVC Japan. That vinyl, originally developed for the CD-4 quadrophony ( and the only one good enough to withstand multiple playings without shaving off the high frequency carrier ),  still blows anything else out of the water - now some 20-30 years ago.
  
 JVC did also do quite well with binaural - both on amateur and pro level.
  
 http://www.discogs.com/No-Artist-Adventure-In-Binaural/release/2747277
  
 http://www.discogs.com/No-Artist-Adventure-In-Binaural-Vol-2/release/3
  
 It was premature, it was not not "finished" - but they definitely knew what they were doing. The least which can be said they deserve some credit for trying to improve things - it went way above ooching and aaaching about falling on the "safe bet" of say Telefunken U-47 ...


----------



## YtseJamer

PUSHING THE AURAL ENVELOPE WITH HIGH-RES AUDIO EVANGELIST STEVEN WILSON

http://www.digitaltrends.com/features/interview-steven-wilson-on-high-res-hand-cannot-erase/


----------



## Krutsch

ytsejamer said:


> PUSHING THE AURAL ENVELOPE WITH HIGH-RES AUDIO EVANGELIST STEVEN WILSON
> 
> http://www.digitaltrends.com/features/interview-steven-wilson-on-high-res-hand-cannot-erase/


 
  
 Wow. Well, I love Steven Wilson for his own music and his 5.1 mixes (e.g. Tears For Fears is amazing).
  
 But statements like these:
  


> One thing that shouldn’t be understated is that *there is a psychological aspect to this.* *If you know you are listening to high-resolution files, there’s something quite comforting about it, you know?* That’s also important. There’s information in those tracks that’s missing when you listen to a CD. *Whether you can hear it or not, it is quite comforting to know that it is there.*
> 
> I know we’ve talked about this before, but I think it’s worth saying again that *all of this high-resolution stuff is pointless if the mastering sucks.* Bad mastering is more of a problem than things being released at CD resolution, or even MP3s. What’s nice about this move to 96/24 is the amount of *things that are coming out in flat transfers — no compression, and no mastering engineers ******* up the sound. That is a very, very good development in the history of music.*


 
  
 ...show his talent for marketing. I mean, these statements will be hard to objectively discuss and will be music to the ears of subjectivists.


----------



## Don Hills

krutsch said:


> ... I mean, these statements will be hard to objectively discuss and will be music to the ears of subjectivists.


 
  
 What he says makes sense to me. You may not hear a difference with increased resolution, but you know it is there. Likewise, I enjoy driving my car more when I've just washed and waxed it, even though the performance is unchanged and I can't see the difference from inside the car.
  
 And he's very much right about the mastering (and overall production) quality being much more important than the resolution.


----------



## bigshot

Steven Wilson's Jethro Tull remixes (War Child) and XTC (Drums and Wires) leave out important elements of the mix and alter the style of the music. I am a big fan of 5.1, but I have yet to hear a good Steven Wilson mix.


----------



## RRod

don hills said:


> What he says makes sense to me. You may not hear a difference with increased resolution, but you know it is there. Likewise, I enjoy driving my car more when I've just washed and waxed it, even though the performance is unchanged and I can't see the difference from inside the car.
> 
> And he's very much right about the mastering (and overall production) quality being much more important than the resolution.


 
  
 Yeah but when you sell your car do you sell it at twice the price because you waxed it?


----------



## analogsurviver

I remember the only reggae concert I attended live - not by the concert itself, not even by the artist, but by the statement the lead singer made during that concert in late 70s :
  
*Computers are killing the music*​  ​ I have to admit I was shocked and completely clueless as to what he meant - vinyl records were still in the shops, CD was something no one outside Philips/Sony labs held in his/her hands, it was still relatively good regarding the sound quality.
  
 And then came any number of synthesizers, CD, computers, DAWs .... - the possibility to do "more" :


----------



## analogsurviver

rrod said:


> Yeah but when you sell your car do you sell it at twice the price because you waxed it?


 
 One should not do that.
  
 Regarding the psychology of knowing to have done everything - or at least "enough" - an example from cycling :
  
 Any rider will tell you that it feels much different if he/she him/herself has cleaned and inspected the bike prior to a (n important) ride. True, even the pros have mechanics to do this job for them, it is their/YOUR head - a slight crack that developed on the frame during the last ride covered by even the slightest residue of a light shower during that last ride may well go unnoticed - and can cost you life during a high speed descent. Same goes for "slight" cuts in tyres, etc, etc.
  
 It definitely _*feels *_different seating on a bike that has been prepared immaculately. And it helps extracting that last n-th degree of an effort - which would not have been attempted if the bike was soiled from the previous ride(s). 
  
 An absolute fanatic regarding this was/is? Eddy Merckx. In his time, handlebar tape was fabric - and each day, his handlebars had to be graced by a new fresh tape.
  
 Which objectively does not contribute (unless torn/worn/greasy - unlikely after a single day without a crash ) to the performance in the slightest. Yet it helped him to win.


----------



## lamode

analogsurviver said:


> *Computers are killing the music*​  ​


 
  
 Couldn't disagree more. Almost all of my favourite artists (other than classical/jazz) rely on electronic (and computer based) instruments to some degree.
  
 The quality/price ratio of a modern DAW also puts serious sound quality in the hand of the masses, giving access to a lot more talented musicians who might otherwise miss out.
  
 Digital audio is the best thing that ever happened to music.
  
 And you keep willfully ignoring the fact that post production tools can be used well or poorly, that is not the fault of the tool but the fault of the engineer if they are used poorly, and there are many analogue equivalents of these tools too. Compression, for example, is NOT a 'digital problem'.


----------



## analogsurviver

lamode said:


> Couldn't disagree more. Almost all of my favourite artists (other than classical/jazz) rely on electronic (and computer based) instruments to some degree.
> 
> The quality/price ratio of a modern DAW also puts serious sound quality in the hand of the masses, giving access to a lot more talented musicians who might otherwise miss out.
> 
> ...


 
 I can sympathize with you regarding the "dirty" musicians ( anyone who has to plug into any electrical device in order for us to hear him/her in the first place ).
  
 I agree that digital has allowed the access to recording music to the masses - which is a good thing.
  
 Post production tools can be used well or poorly, yet I resort to them only if no other course is available. In acoustic music, if correct from the playing/performance view, they are not necessary - at all. For those musicians that rely on electronic > computer based instruments, it is an unavoidable necessary evil.
  
 Compression, at least what the recording industry has been throwing at us in the last 20 or so years, IS predominantely a "digital" problem - just go trough the Bob Katz video. It "democrately affordable" allowed for loudness wars - culminating in one evening of our composer association, dedicated to new electroacoustic works. Except for the work I helped to record for a friend composer, ALL of the rest had the dynamic range of 3,2,1, (yours) dB - and sounded interesting exactly the same as a cold cup of tea. Noise in duration of so and so much time, with the minimum "modulation" imaginable. Although "my" composition has been intentionally compressed in post production,  it  still clearly allowed to hear the interplay of what started as natural sound - and its electronically manipulated "echoes", if you will. 
  
 One musical group that is borderline between "clean" and "dirty" is Oregon. They always have their own sound engineer with them - and HE is the key but invisible member - at least what concerns live shows. And they are perfectly aware of this fact...


----------



## bfreedma

bigshot said:


> Steven Wilson's Jethro Tull remixes (War Child) and XTC (Drums and Wires) leave out important elements of the mix and alter the style of the music. I am a big fan of 5.1, but I have yet to hear a good Steven Wilson mix.


 
  
 You should try some of his mixes for his own bands or Yes.  They are, IMO, the best 5.1 out there, particularly his solo work.


----------



## Krutsch

bigshot said:


> Steven Wilson's Jethro Tull remixes (War Child) and XTC (Drums and Wires) leave out important elements of the mix and alter the style of the music. I am a big fan of 5.1, but I have yet to hear a good Steven Wilson mix.


 
  
 OK, well, I have Steve Wilson's mix of Jethro Tull's "Benefit" and I like it, but I am not familiar enough with the original to know if something has deviated from the artist's original vision.
  
 I am, however, very familiar with the Yes albums he's done and these are nice 5.1 mixes, IMO.
  
 But, I think you stated earlier you aren't a fan of Yes, so ignoring Wilson's own work (like "Raven..." or "Storm Corrosion", also great as 5.1 mixes), let me recommend Tears For Fears "Songs From The Big Chair". When I first heard the 5.1 mix of that, I just played it a second time from beginning to end - very engaging surround mix of an album I know extremely well.
  
 You should try some of his other stuff, if you like 5.1, because there just isn't that much 5.1 audio out there (outside of classical works) and Wilson is really trying to expand the audience for 5.1.


----------



## U2Bono269

analogsurviver said:


> One should not do that.
> 
> Regarding the psychology of knowing to have done everything - or at least "enough" - an example from cycling :
> 
> ...


 

 I agree with all of this. Tiny things can change our perception of how we hear or see. And what ultimately matters is our perception. I do find it nice to think that my digital files are bit-perfect representations of my CDs. And it's also nice to think that the couple of HD albums I have are (in theory at least) representations of what my favorite bands recorded in the studio.
  
 Yes, it's annoying and wrong that the cost of high-res music is so high. But I do think it might one day be worth it with the right technology. I know someone (bigshot maybe??) argued that our ears can't hear it so any new technologies would be worthless. Still, there are things we can't see with our eyes because we lack the resolution, but there are numerous instruments like microscopes and such that allow us to see things we couldn't normally see. It may sound farfetched, but we don't know what kind of audio technology will exist next year, let alone 10 or 20 years from now. If such a thing turned out true, it would be most likely be nothing like we imagine and would be something completely out of the box.
  
 I believe that _at this time_ we are unable to hear a difference between 16bit and 24bit files. And in the meantime, if it increases your enjoyment of music for any reason at all (and you can afford it), then go for it. It's worthwhile if someone thinks it's worthwhile.


----------



## sonitus mirus

u2bono269 said:


> I believe that _at this time_ we are unable to hear a difference between 16bit and 24bit files. And in the meantime, if it increases your enjoyment of music for any reason at all (and you can afford it), then go for it. It's worthwhile if someone thinks it's worthwhile.


 
  
 I'm guessing from the use of italics that you assume we may somehow develop superior hearing abilities that would change this view?  
  
 If the reason is purely psychological, it is probably not worthwhile to anyone except the person making money on a sale.


----------



## bigshot

Inaudible is inaudible.


----------



## Krutsch

http://theghostinthemp3.com/theghostinthemp3.html
  
 Interesting read and I am eager to hear the samples, when I can take the time.


----------



## U2Bono269

sonitus mirus said:


> I'm guessing from the use of italics that you assume we may somehow develop superior hearing abilities that would change this view?
> 
> If the reason is purely psychological, it is probably not worthwhile to anyone except the person making money on a sale.


 

 Did you actually read what I said, or just cherry pick the part you wanted to make fun of?


----------



## sonitus mirus

u2bono269 said:


> Did you actually read what I said, or just cherry pick the part you wanted to make fun of?


 
  
 I completely disagree with your assessment that perception is all that matters when it comes to music and the equipment used to play it.  My perception is all over the map, depending on a myriad of factors.  The one thing I want to be certain about is the sound quality.  I'm not concerned with how much the equipment costs or how big the files might be.  My perception will change.  Why not only listen to music when you suck on a peppermint candy if it makes the music sound better on Tuesdays?  Perception is unreliable and ultimately one of the last things that matters.
  
 I don't believe a microscope and our vision is an appropriate analogy.   A microscope or telescope is not improving your sight.  It is the equivalent of increasing or decreasing an otherwise inaudible frequency to something within the human hearing range.
  
 A transparent medium that you can see through compared to a transparent audio signal would be more appropriate.  Transparent is transparent.  Invisible is invisible.  "Inaudible is inaudible." (-bigshot)


----------



## U2Bono269

bigshot said:


> Inaudible is inaudible.


 

 I understand your point, but at the same time, ultraviolet light is invisible to our eyes. So invisible is invisible, right? Yet, technology exists that allows artists to make beautiful works of art through ultraviolet photography. My only point is that if that exists, it's possible that something similar could happen with the bits of audio that we can't hear. You're free to disagree, but are you so narrow-minded that you can't consider the possibility?


----------



## bigshot

I think we understand what you're saying... The difference isn't audible, but larger file sizes make you feel more secure. You are willing to give up drive space and spend more money for that feeling of security. That's a perfectly good explanation. We accept it. It's just that chunky file sizes don't do anything at all for us. As long as a format is convenient and is audibly transparent, we don't worry or obsess over missing out on some file size.


----------



## U2Bono269

sonitus mirus said:


> I completely disagree with your assessment that perception is all that matters when it comes to music and the equipment used to play it.  My perception is all over the map, depending on a myriad of factors.  The one thing I want to be certain about is the sound quality.  I'm not concerned with how much the equipment costs or how big the files might be.  My perception will change.  Why not only listen to music when you suck on a peppermint candy if it makes the music sound better on Tuesdays?  Perception is unreliable and ultimately one of the last things that matters.
> 
> I don't believe a microscope and our vision is an appropriate analogy.   A microscope or telescope is not improving your sight.  It is the equivalent of increasing or decreasing an otherwise inaudible frequency to something within the human hearing range.
> 
> A transparent medium that you can see through compared to a transparent audio signal would be more appropriate.  Transparent is transparent.  Invisible is invisible.  "Inaudible is inaudible." (-bigshot)


 

 If I think my music sounds better for whatever reason, then it does...to me. Bottom line. If I think my music sounds better with peppermint candy, then guess what? It does...to me.
  
 Though for the record, I don't think it does in this case, I'm just agreeing with the other person's point.
  
 I don't think it's at all unreasonable to believe that technology could one day exist to take advantage of otherwise inaudible frequencies in any of a number of different and currently unthinkable ways.


----------



## castleofargh

u2bono269 said:


> bigshot said:
> 
> 
> > Inaudible is inaudible.
> ...


 

 you're talking about DSPs. when I take an IR picture, I can't see IR. so I use a trick moving the signal into the visible range. it's kind of like changing the pitch of the song, some ultrasounds will slow down and become trebles. but I fail to see how that is a technology improvement, or how it is supposed to mater in music.
 it's not about being open minded, it's about looking at what will and will not change. when we already fail to distinguish 16/44 compared to pretty much anything above, what do you expect to hear with further resolution increase? more of the exact same perceived sound.


----------



## sonitus mirus

Just different opinions, nothing wrong with that, though I'm sure you think that my opinion is wrong. 
	

	
	
		
		

		
		
	


	



  
 Since we hear with our brains, it would be neat if we could all somehow listen to music only in our own heads with perfect hearing and no worries about damaging our ears.  There has been some work done to bring sound to the deaf along these ideas, but I couldn't find a link in a quick search.  If this ever becomes a reality, we may all be listening without ears at all, just some implant in our jaw, and perhaps we will need much higher resolutions.


----------



## U2Bono269

castleofargh said:


> you're talking about DSPs. when I take an IR picture, I can't see IR. so I use a trick moving the signal into the visible range. it's kind of like changing the pitch of the song, some ultrasounds will slow down and become trebles. but I fail to see how that is a technology improvement, or how it is supposed to mater in music.
> it's not about being open minded, it's about looking at what will and will not change. when we already fail to distinguish 16/44 compared to pretty much anything above, what do you expect to hear with further resolution increase? more of the exact same perceived sound.


 

 You could be absolutely correct. I wouldn't expect to hear anything with a resolution increase. But my argument is theoretical, so in order for it work, sound engineers would need to do something different to take advantage of whatever this technology might be.
  
 Look, I know it sounds a little crazy and not really rooted in our current understanding of sound science. I'm a firm believer that there is no limit to the human imagination and there are endless possibilities. Maybe I read too much science fiction.


----------



## U2Bono269

sonitus mirus said:


> Just different opinions, nothing wrong with that, though I'm sure you think that my opinion is wrong.
> 
> 
> 
> ...


 

 Exactly! That, I agree with. Just because we can't do anything with it now doesn't mean we won't one day.
  
 One of my friends is deaf, and told me about something similar. He has a cochlear implant and sometimes complains of high frequencies getting picked up and drowning out other sounds that he wants to hear. He also said once that he wished he could hook his iphone directly into it and not have to hold the phone to his ear.
  
 There are just so many possibilities...why limit our thinking?


----------



## threephi

u2bono269 said:


> I understand your point, but at the same time, ultraviolet light is invisible to our eyes. So invisible is invisible, right? Yet, technology exists that allows artists to make beautiful works of art through ultraviolet photography. My only point is that if that exists, it's possible that something similar could happen with the bits of audio that we can't hear. You're free to disagree, but are you so narrow-minded that you can't consider the possibility?


 

 The way ultraviolet photography works is by converting ultraviolet frequencies down into the range that is already visible by humans.  There are a lot of examples of this in astronomy photos and other scientific photography as well as art.  100% perfectly analogous technologies do of course already exist for audio.  Some recordings of dolphin and whale songs are an example--the significant ultrasonic frequencies are converted down into our human audible range.  But none of these change the range of what humans can see or hear, what they do is alter the _information _to fit within that unchanged, unchanging natural biological range.
  


u2bono269 said:


> If I think my music sounds better for whatever reason, then it does...to me. Bottom line. If I think my music sounds better with peppermint candy, then guess what? It does...to me.
> 
> Though for the record, I don't think it does in this case, I'm just agreeing with the other person's point.
> 
> I don't think it's at all unreasonable to believe that technology could one day exist to take advantage of otherwise inaudible frequencies in any of a number of different and currently unthinkable ways.


 
 I agree with you completely on the first point, although I and others here clearly make a different choice.   If it pleases you, it's your right to store and play back music in whatever format you want, end of story, whether it is rational or not.  It's unusual to be able to maintain the placebo effect once the curtain has been lifted, but if you can, more power to you.

 To the second point, the only technology that would achieve what you are talking about would be some kind of cyborg/replicant artificial sensory organs which expanded the actual range of our senses.  Short of that, we can't see what we can't see, and we can't hear what we can't hear.  Down-converting frequencies, which is already very commonly done, is a work-around which doesn't change that range.


----------



## U2Bono269

threephi said:


> To the second point, the only technology that would achieve what you are talking about would be some kind of cyborg/replicant artificial sensory organs which expanded the actual range of our senses.  Short of that, we can't see what we can't see, and we can't hear what we can't hear.  Down-converting frequencies, which is already very commonly done, is a work-around which doesn't change that range.


 
  
 I tend to think that if we're ever able to take advantage of these frequencies, it will be in an entirely different manner than we would traditionally expect.
  
 I have a whopping 4 albums in high-resolution. I don't plan on picking any more up unless I know it's a really great master that I can't get elsewhere or a brand new master of one of a handful of albums I really, really love.


----------



## bigshot

u2bono269 said:


> But my argument is theoretical


 
  
 Entirely!


----------



## threephi

u2bono269 said:


> I tend to think that if we're ever able to take advantage of these frequencies, it will be in an entirely different manner than we would traditionally expect.


 
 I agree with you there, but at that point, we're no longer talking about _hearing _those frequencies, which is the question under consideration.  There are some really fascinating cross-sensory technologies that have emerged in the last few years, which some others touched on above.  One example is the BrainPort, which is a device for the blind that translates visual data into electrical stimulation on the surface of the tongue.  With only a little bit of acclimation, people intuitively translate that into spatial perception.  That tells me that the brain's sensory pathways are quite flexible, and can be reprogrammed under certain circumstances.  So yeah, it's conceivable that we might at some point develop a device which translated ultrasonic frequencies onto some other sensory input, which our brains could then be trained to interpret as some kind of extension of the experience of listening.  We still wouldn't be able to hear them though.


----------



## bigshot

The AES did a test where they asked all kinds of people to ID the "best sound quality in music". There was nothing to indicate that high frequencies made a lick of difference. In fact, they rolled off everything above 10kHz, and most people said that it sounded the same. Not surprising because 10-20kHz is just one little octave at the edges of human perception.


----------



## kraken2109

bigshot said:


> Steven Wilson's Jethro Tull remixes (War Child) and XTC (Drums and Wires) leave out important elements of the mix and alter the style of the music. I am a big fan of 5.1, but I have yet to hear a good Steven Wilson mix.


 
 Try some of his solo stuff or porcupine tree in 5.1. I like pretty much all of them, I've got in 5.1:

Stupid Dream (PT)
Deadwing (PT)
Anesthetize (PT tour DVD)
Grace for Drowning (SW)
Storm Corrosion (SW)
The Raven That Refused to Sing (SW)
  
 All the mixes make good use of 5.1 and sound better than the stereo versions IMO.


----------



## YtseJamer

New Steven Wilson is now available @ HDtracks

http://www.hdtracks.com/hand-cannot-erase-1


----------



## bfreedma

ytsejamer said:


> New Steven Wilson is now available @ HDtracks
> 
> http://www.hdtracks.com/hand-cannot-erase-1




I can't think of a reason to buy HCE from HDTracks. The BR disc is basically the same price, has the content in stereo, 5.1, and instrumental versions. All in 96/24. Also has alternate versions of the songs and a 30 minute "making of" video. And a FLAC download code.

HDtracks has only the 96/24 stereo content. I love the concept of HDTracks but their pricing is, IMO, ridiculous.


----------



## Krutsch

bfreedma said:


> I can't think of a reason to buy HCE from HDTracks. The BR disc is basically the same price, has the content in stereo, 5.1, and instrumental versions. All in 96/24. Also has alternate versions of the songs and a 30 minute "making of" video. And a FLAC download code.
> 
> HDtracks has only the 96/24 stereo content. I love the concept of HDTracks but their pricing is, IMO, ridiculous.


 

 +1 ... I ordered the BD for the reasons you mention. But for some, ripping the BD is a challenge and is inconvenient, hence the download code, which is a nice touch. Is the FLAC download 44.1/16 or 96/24?


----------



## bfreedma

krutsch said:


> +1 ... I ordered the BD for the reasons you mention. But for some, ripping the BD is a challenge and is inconvenient, hence the download code, which is a nice touch. Is the FLAC download 44.1/16 or 96/24?


 
  
 I don't know what format the FLAC is in.  I ordered the deluxe set with the book and other material and haven't looked to see if there was a download card in there somewhere yet.  If there is, I'll download and let you know.
  
 As ridiculous as it is to have to spend money to rip my own BRs, I broke down a while back and purchased AnyDVD HD, so ripping BRs is pretty simple.  Nothing like having to pay to use content you legally purchased.....


----------



## lamode

u2bono269 said:


> I understand your point, but at the same time, ultraviolet light is invisible to our eyes. So invisible is invisible, right? Yet, technology exists that allows artists to make beautiful works of art through ultraviolet photography. My only point is that if that exists, it's possible that something similar could happen with the bits of audio that we can't hear. You're free to disagree, but are you so narrow-minded that you can't consider the possibility?


 
  
 This post smacks of desperation. In spite of that, your UV example is spot on. It's like releasing photo prints which contain UV information even though the human eye can't see it. Completely pointless. Yes, you can convert UV images to visible images. That would be like slowing down ultrasonic recordings to make them audible. Completely irrelevant to hi-fi sound reproduction.


----------



## U2Bono269

lamode said:


> This post smacks of desperation. In spite of that, your UV example is spot on. It's like releasing photo prints which contain UV information even though the human eye can't see it. Completely pointless. Yes, you can convert UV images to visible images. That would be like slowing down ultrasonic recordings to make them audible. Completely irrelevant to hi-fi sound reproduction.


 

 I fail to see how I'm being desperate. I completely recognize that I can't hear the difference between the 2 formats. I've said it a dozen times. I just believe the possibility exists that one day someone may be able to do something interesting with it. That's it.
  
 I completely disagree that ultraviolet or infrared photo prints are useless. There are some incredibly beautiful works of art done using UV and IR cameras. Yes, they are the result of digital filtering (or trickery) but that doesn't change the fact that the end result can be a unique experience that we don't get through traditional still photography. Perhaps someone, somewhere, somewhen will be able to manipulate ultrasonic or subsonic frequencies into a unique experience we don't get through traditional music.


----------



## bigshot

An infrared photo is like taking gamelan music and pitching the super audible frequencies down into the audible range. But you don't need infrared in a photo of your mom or your dog. You don't need super audible frequencies in your copy of Dark Side of the Moon or Bob Dylan either.


----------



## RRod

bfreedma said:


> I can't think of a reason to buy HCE from HDTracks. The BR disc is basically the same price, has the content in stereo, 5.1, and instrumental versions. All in 96/24. Also has alternate versions of the songs and a 30 minute "making of" video. And a FLAC download code.
> 
> HDtracks has only the 96/24 stereo content. I love the concept of HDTracks but their pricing is, IMO, ridiculous.


 
  
 Yeah, the pricing makes me shake my head. I just ordered this disc:
 https://www.sonoluminus.com/p-369-toccatas-blu-ray-cd.aspx
  
 I got a new disc for $14 on the Amazon market. That gets me a Blu-ray with 2.0 and 5.1 @ 24/192 and 7.1 @ 24/96 AND a Redbook CD. Even at list price, that beats what you get from HD Tracks and Pono, let alone for $14.


----------



## U2Bono269

bigshot said:


> An infrared photo is like taking gamelan music and pitching the super audible frequencies down into the audible range. But you don't need infrared in a photo of your mom or your dog. You don't need super audible frequencies in your copy of Dark Side of the Moon or Bob Dylan either.


 

 Yes, you're right. It's a different thing entirely. Doesn't make it wrong though. Just different.
  
 I never said I would want infrared in a normal photo. It's a different application that uses invisible light.
  
 I also wouldn't want those frequencies in my Bob Dylan. That's missing the point. But it might be interesting to hear music that was made specifically with super-audible frequencies in mind.
  
 Is there any commercially available music that pitches super audible frequencies into an audible range? I tried to Google it but I'm not 100% sure exactly what to search for.


----------



## RRod

u2bono269 said:


> Yes, you're right. It's a different thing entirely. Doesn't make it wrong though. Just different.
> 
> I never said I would want infrared in a normal photo. It's a different application that uses invisible light.
> 
> ...


 
  
 Most of the hi-res content is going to be upper harmonics of the low-frequency content, so if you move the hi-res stuff down into the audible range there's a good chance it would just get masked and you wouldn't hear it anyway (especially since it's lower in volume unless you gain boost it). What you could hear would certainly clash with the original low-res content.


----------



## bigshot

u2bono269 said:


> Is there any commercially available music that pitches super audible frequencies into an audible range? I tried to Google it but I'm not 100% sure exactly what to search for.


 
  
 Here you go... https://www.youtube.com/watch?v=7-t_Bo54ftM


----------



## U2Bono269

bigshot said:


> Here you go... https://www.youtube.com/watch?v=7-t_Bo54ftM


 

 Clever. Whatever man.


----------



## U2Bono269

rrod said:


> Most of the hi-res content is going to be upper harmonics of the low-frequency content, so if you move the hi-res stuff down into the audible range there's a good chance it would just get masked and you wouldn't hear it anyway (especially since it's lower in volume unless you gain boost it). What you could hear would certainly clash with the original low-res content.


 

 That's interesting. I wonder if one could strip away the low-res content and manipulate the sounds that are left?


----------



## RRod

u2bono269 said:


> That's interesting. I wonder if one could strip away the low-res content and manipulate the sounds that are left?


 
  
 Of course, you can do anything really. But the material at the top isn't really "musical" as raw material: the frequency relationships sound all wrong.
  
 As an example, here's a clip of hi-res material, pitched down a few octaves:
 https://drive.google.com/file/d/0BwmVtb5IwniEemVTRXlzYVkyZTQ/view?usp=sharing


----------



## bigshot

u2bono269 said:


> Clever. Whatever man.


 
  
 Isn't that what you asked for? Frequencies above the range of human hearing are beyond the range of control by musicians. Musicians can't hear them. They are going to be just sounds, not music.


----------



## reginalb

Also, correct me if I'm wrong here, but couldn't you still use 16/44.1 after bringing that stuff down in to the audible spectrum? Once again, for final distribution you wouldn't need high res.


----------



## RRod

reginalb said:


> Also, correct me if I'm wrong here, but couldn't you still use 16/44.1 after bringing that stuff down in to the audible spectrum? Once again, for final distribution you wouldn't need high res.


 
  
 Yes, but I assume the argument was focused on making sure these sounds made it through the ADC at least, so they were available to use for whatever nefarious purpose (like selling people DSotM again).


----------



## bigshot

Considering that the only real sound information that exists in music above 20kHz is upper harmonics on cymbal crashes, I don't think pitched down super audible frequencies is going to sound any better than the trash man banging your garbage cans around on trash day. It's not like musicians are able to create music they can't even hear.


----------



## OddE

bigshot said:


> It's not like musicians are able to create music they can't even hear.




-Beethoven;  Ninth. 

(Yes, not quite the same thing - but an opportunity too good to pass up, anyway...


----------



## threephi

odde said:


> -Beethoven; Ninth.
> 
> (Yes, not quite the same thing - but an opportunity too good to pass up, anyway...


 
 I know what you did there, but I would maintain that Beethoven did hear his music, in his mind.  Ears are for amateurs


----------



## OddE

threephi said:


> I know what you did there, but I would maintain that Beethoven did hear his music, in his mind.  Ears are for amateurs




-And where, do you reckon, does that leave the Head-Fi community?


----------



## StanD

odde said:


> -And where, do you reckon, does that leave the Head-Fi community?


 
 Some may be Deaf and D.....?


----------



## RRod

odde said:


> -Beethoven; Ninth.
> 
> (Yes, not quite the same thing - but an opportunity too good to pass up, anyway...


 
  
 I know who I'd ask about deafness and hi-frequency content:


----------



## prot

A pretty cool article about highres from a mainstream journal. Much better than most I read in the so called audio press
http://www.wsj.com/articles/hi-res-audio-hijinx-why-only-some-albums-truly-rock-1425675329?KEYWORDS=high+resolution+audio


----------



## lamode

prot said:


> A pretty cool article about highres from a mainstream journal. Much better than most I read in the so called audio press
> http://www.wsj.com/articles/hi-res-audio-hijinx-why-only-some-albums-truly-rock-1425675329?KEYWORDS=high+resolution+audio


 
  
 It's easy to see how a n00b like that reporter could get sucked into the marketing BS... All he had to do was compare a 16/44 and a hi-res copy of the same master, and he would know that the differences are not due to the format.


----------



## StanD

lamode said:


> It's easy to see how a n00b like that reporter could get sucked into the marketing BS... All he had to do was compare a 16/44 and a hi-res copy of the same master, and he would know that the differences are not due to the format.


 
 He almost figured that out but fell and hit his head along the way.


----------



## threephi

From the article: 


> I relaxed into a seat in front of Magico’s $229,000 Q7 Mark II speakers. Mr. Wolf played a jazz track from the album “Mirror,” by saxophonist Charles Lloyd—first in CD quality, then in high-resolution—and spoke of the subtle difference, a richness that hi-res delivers. I believed I could hear it, too. But maybe I just wanted to.


 
 Rather telling that a high priest of hi-res felt he had to coach the reporter in real time in a non-blind comparison in order to hear the difference.


----------



## Soundsgoodtome

https://www.yahoo.com/music/jack-white-on-not-being-a-sound-bite-artist-113177424986.html?soc_src=mail&soc_trk=ma

 Jack White, anti-digital.


----------



## Avi

bigshot said:


> u2bono269 said:
> 
> 
> > Is there any commercially available music that pitches super audible frequencies into an audible range? I tried to Google it but I'm not 100% sure exactly what to search for.
> ...


 
 That was amazing; thank you, bigshot.


----------



## Soundsgoodtome

In response to the HDtracks thread: http://www.head-fi.org/t/756377/special-hdtracks-sale-plus-new-led-zeppelin/15#post_11401135


hdtracks said:


> Hello all,
> 
> Thank you very much for all your comments and feedback.  We value each and every one of our users and potential users opinions.  We are constantly striving to make the HDtracks service as great as it can be.  With that being said, we test each and every release in house and independently for resolution and bit depth and stand by all of our designations.  We try to be as transparent as possible in the about section of all of our albums in regards to recording, mastering, remastering etc.  For example with the Beck album,  from the very first day it was made live we commented on the recording and particular tracks "_Tracks 4, 5, 7, 10, 11 contain elements of 48k tracking, mastered in 96/24."  _
> 
> ...


----------



## Roly1650

soundsgoodtome said:


> In response to the HDtracks thread: http://www.head-fi.org/t/756377/special-hdtracks-sale-plus-new-led-zeppelin/15#post_11401135



Oh good, I'll have to check they've credited my card with all the charges they made for the hi res files that weren't or maybe they'll invite me to download the "proper" files for free.
Note to self, "don't hold your breath".
That statement they've made is total bovine excrement, I still see at least a dozen albums that can't possibly be what they say they are and that without looking too hard.


----------



## threephi

> At the very beginning of HDtracks *due to our naivete* we released a very few amount of albums that were not up to par with our hi-res designation.  *This was due to hi-res audio being a very new thing, miscommunication between us and labels, us not doing our due diligence and a few other reasons*.  Since then we have gone above and beyond to make sure we put out the very best product possible and test each and every file.  And in cases like the Beck album we give as much information as humanly possible to put forth all the information we have, so you the consumer can make an educated decision.


 
 Wow, so in other words, they started their business with zero expertise, knowledge, or care about what they were doing, and made it up on the fly.  And using the fact that "hi-res audio [was] a very new thing" as an excuse is another way of saying that *they couldn't tell the difference between hi-res and "normal" tracks*.  Doesn't give one much reason to think they're selling anything but snake oil.


----------



## bigshot

> As you can see on this thread alone their are people who are very happy with the sound of the Beck album, once again because music is subjective and HDtracks is not going to take that away.


 
  
 And more digital packing peanuts filling up the file size isn't going to make it any better either! Subjective impressions of the music are up to Beck, and Kanye has something to say about that!


----------



## Greenears

Such a wonderful marketing answer that I would be proud of. So many paragraphs written without actually stating that their 24 bit files are actually recorded and mastered in 24 bit. Such complicated information to get from such busy studios.... ah me....


----------



## fallen angel

Really sorry if this has been covered, but I have two questions. 
  
 If I understand correctly, the bit size (16/24) is relative to the db range the music has. Would it not be possible that 24bit can fit into the same db range but perhaps with more accuracy? Certainly, we don't want music playing at 150db, but perhaps that 60 or so db range can have more detail using 24 bit, something like over sourcing I guess?
  
 for resolution. I was actually thinking before I read it "I wonder if they use 44k to cover the 22kHz range in music - that could cover both wavelengths." Turns out it seems right. Though, I would wonder if perhaps 96k resolution may be better to pick up off phase sounds (if I said that correctly) and not so much for playback of higher frequencies that we can't possibly hear. Rather, higher resolution just the same as a picture, more detail within the hearing range we do have.
  
 If anything I'm thinking applies or makes sense, I would suppose that 24/96 is ideal. Though both arguments pretty much say the same thing, more detail within the range we do listen to and not so much used to extend beyond that range. I would think, being a digital format, it couldn't possibly merrit every single bit of range  either frequency or decibel output, just close enough that one would have a very hard time pointing it out. Would that make any sense or am I completely off base?


----------



## RRod

fallen angel said:


> Really sorry if this has been covered, but I have two questions.
> 
> If I understand correctly, the bit size (16/24) is relative to the db range the music has. Would it not be possible that 24bit can fit into the same db range but perhaps with more accuracy? Certainly, we don't want music playing at 150db, but perhaps that 60 or so db range can have more detail using 24 bit, something like over sourcing I guess?
> 
> ...


 
  
 As far as bit depth, it gets easier to understand what's going on if you view it as a rounding problem. You have some set of continuous voltages taken at snapshots in time that you need to round to some number the computer can understand. This means that your final sampled signals can be seen as:
 sample = actual + rounding
 Thus the frequency content of the final sample will be the frequency content of the un-rounded samples plus the frequency content of the rounding (quantization) error. So we need to understand what the frequency content of the rounding looks like and we're done. The general result is that for high amplitude signals, the rounding frequency content looks basically like noise, and is many dB down from the actual signal. As the signal amplitude gets lower, it starts to look more like harmonic distortion and it gets relatively louder compared to the sample. The number of bits reduces the rounding error: you can get closer to the exact value. This gives you a larger dynamic range before you start to see the distortion errors, and means that the noise is at a lower level in general relative to the signal.
  
 As far as higher sample rates for "off-phase" sound, is this the kind of thing you are talking about?
 http://forums.stevehoffman.tv/threads/time-resolution-of-red-book-45ns.85436/


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## castleofargh

fallen angel said:


> Really sorry if this has been covered, but I have two questions.
> 
> If I understand correctly, the bit size (16/24) is relative to the db range the music has. Would it not be possible that 24bit can fit into the same db range but perhaps with more accuracy? Certainly, we don't want music playing at 150db, but perhaps that 60 or so db range can have more detail using 24 bit, something like over sourcing I guess?
> 
> ...


 

 I'll try the dumb explanation as that's the only way I can think ^_^.
 let's say one sound is a wave, another sound is another wave, when a band is playing you record only one signal accumulating all the waves. you know like ripples on the surface of water that will add up or cancel each others so that the surface at one point is always only at one position. same for the album, the signal is always only at only 1 amplitude at a time, however how many instruments are recorded.
  when the music is recorded on 16bit instead of 24bit, you end up with each sample slightly different from the original, that difference to us is as if another instrument had been added to the playing band. it doesn't change how the band sounds, just like if you added one more guitar in the mix, the singer would still sing the same way, but the signal would be modified.
 that's the magic of audio, quantization errors don't change the music, they just add some noise to the music.
 now does it matter? well that noise on a 16bit track is 96DB quieter than the max signal, so even with some headroom on the record, you can expect the noise to be a good 90db below the music. remember that the quietest part of the song will rarely go 60db below the max loudness. so you pretty much end up with a slight hiss some 30db below the quietest sound recorded on the most dynamic album you own. that's how dramatic 16/44 really is ^_^.
  
 you don't improve music, with highres you only improve the silence really. on both resolutions the music itself will be perfectly reproduced down to the quietest sound.
 going to 24bit reduces the error value of the samples, so you end up with the band playing and now some noise at -144db(in theory at least) instead of -96. but it changes nothing to the sound of the band. that's the crazy cool thing about sound.
  
 and getting higher sample rate can yield the same kind of result, be it in time error or in noise(both are linked for waves). it is factually better, but could you actually hear the noise on a 16/44 track? if you could, then highres might be justified, if you couldn't(like everybody at normal listening level) you're only improving something you don't hear and most likely are paying premium price for that.


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## limpidglitch

fallen angel said:


> for resolution. I was actually thinking before I read it "I wonder if they use 44k to cover the 22kHz range in music - that could cover both wavelengths." Turns out it seems right. Though, I would wonder if perhaps 96k resolution may be better to pick up off phase sounds (if I said that correctly) and not so much for playback of higher frequencies that we can't possibly hear. Rather, higher resolution just the same as a picture, more detail within the hearing range we do have.


 
  
 Phase is essentially a matter of timing, and as PCM (whether 16/44.1, 24/96 or whatever) doesn't really have a temporal resolution (there are no discrete steps), as it has for amplitude, it isn't really a problem.
 You can sample at 16kHz (typical VoIP), there still isn't a limit to temporal resolution. The only limit is the number of cycles per second (Hz) the transmitted sound can contain, namely half the frequency rate.


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## fallen angel

Quantization and jitter.. makes a lot more sense now (followed that link, explained jitter perfectly)
  
 Now trying to understand the jitter. I would think at 44k, that would allow up to 22k jitter, 0.0022s (been a long time since I've done math). Supposedly we can hear 200ps of jitter? I can't figure how that math correlates.


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## Greenears

I've written at length about quantization and jitter somewhere between pages 50 and 100 of this thread   And so have others.  Search my posts.
  
 The bottom line is you can run in circles on all the mathematics all you want there is not enough information available to prove mathematically whether quantization error is audible at 16 bit.  Jitter is independent of 24 vs 16 bits, but it is a component of total quantization error.  Jitter is also one of the most common audiophile red herrings, just before (or after?) silver cables.
  
 So in the end you only have blind listening tests to go by.  And that's a different can of nice round plump juicy worms.


----------



## lamode

fallen angel said:


> Quantization and jitter.. makes a lot more sense now (followed that link, explained jitter perfectly)
> 
> Now trying to understand the jitter. I would think at 44k, that would allow up to 22k jitter, 0.0022s (been a long time since I've done math). Supposedly we can hear 200ps of jitter? I can't figure how that math correlates.


 
  
 As stated in my sig...
 Don't be fooled by the jitter scam. Real world level of jitters are inaudible. Study here: https://www.jstage.jst.go.jp/article/ast/26/1/26_1_50/_pdf


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## StanD

lamode said:


> As stated in my sig...
> Don't be fooled by the jitter scam. Real world level of jitters are inaudible. Study here: https://www.jstage.jst.go.jp/article/ast/26/1/26_1_50/_pdf


 
 Everytime I ask an "audiophile" that's doing the jitterbug, how many ps can a human perceive and how does it manifest in an observable audio artefact. I never get an answer, suddenly there is silence.


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## fallen angel

Kinda what I thought. I will fully admit I can't really tell you the difference between flac 16/44 and anything better, I always chalk it up to better mastering.


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## bigshot

stand said:


> Everytime I ask an "audiophile" that's doing the jitterbug, how many ps can a human perceive


 
  
 20 ns


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## StanD

bigshot said:


> 20 ns


 
 And what is the audible effect? I just love those guys chasing pico seconds and femto clocks.


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## fallen angel

Can a digital format even produce that? Crazy small measurement that is.


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## DiscoProJoe

Years ago when playing with the sound card settings on my PC, I couldn't notice the slightest difference between 24-bit, 48kHz and 16-bit, 44.1kHz, so I've always chosen the latter.  Since then, I read somewhere that the only practical purpose of 24-bit sound is when musicians record, master, mix, and remix music, because it allows a for larger margin of error without causing noise or distortion in the recording. But once the final product is ready for release, all you need is MP3 quality at 192 kbps, in my biased opinion!
  
 And oh,...I also read somewhere that humans can only notice the loudest 30 dB of sound that's present at any given moment, and usually can't notice distortion that's less than 3%. So for stereo equipment that isn't being used in a studio, my guess is that dynamic range ratings and distortion ratings haven't meant diddly squat since at least the 1960s when transistors become common.


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## DiscoProJoe

By contrast, I also wanted to add that signal-to-noise ratings _*are*_ important for personal stereo equipment, since they allow for more flexibility in how the user can set the amp gains and pre-amp gains without getting audible noise or hiss, particularly when the music is turned down or paused. But...worrying about dynamic range and distortion ratings since the advent of transistors? Meh....


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## castleofargh

discoprojoe said:


> Years ago when playing with the sound card settings on my PC, I couldn't notice the slightest difference between 24-bit, 48kHz and 16-bit, 44.1kHz, so I've always chosen the latter.  Since then, I read somewhere that the only practical purpose of 24-bit sound is when musicians record, master, mix, and remix music, because it allows a for larger margin of error without causing noise or distortion in the recording. But once the final product is ready for release, all you need is MP3 quality at 192 kbps, in my biased opinion!
> 
> And oh,...I also read somewhere that humans can only notice the loudest 30 dB of sound that's present at any given moment, and usually can't notice distortion that's less than 3%. So for stereo equipment that isn't being used in a studio, my guess is that dynamic range ratings and distortion ratings haven't meant diddly squat since at least the 1960s when transistors become common.


 

 setting your output to 24bit may let you mess around with some volume setting on the computer(for lazy ease of use instead of always getting to the amp knob).
 192k mp3 is a little low, I actually can abx that with several tracks, when I fail a lot with even some max vbr.
 the 30db thing is wrong out of context. masking may work for even lower values, but it depends on the frequencies of the loud signal and how close it is to the one that will be masked. and if it's a lower or higher freq. in a general way 60db would be a more prudent figure for most situations.


----------



## lamode

fallen angel said:


> Can a digital format even produce that? Crazy small measurement that is.




It is a property of digital hardware, nothing to do with the format


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## stv014

Additionally, the format does in fact allow for delays with such resolution, it is not limited to integer samples.
  
 20 ns peak to peak sinusoid jitter creates a pair of sidebands at -69.2 dBr if the input is a 11.025 kHz tone (the level is proportional to the frequency, it is f * jitter peak level * PI). That is -66.2 dB, or about 0.05% total non-harmonic distortion, which seems to be a reasonable estimate for the threshold.


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## bigshot

stand said:


> And what is the audible effect?


 
  
 High frequency distortion. But no consumer audio gets close to 20 ns of jitter. Usually, it's 100 times less than that.


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## fallen angel

Love the math! Makes sense, though I still can't point it out myself, cool to see what actually is going on.


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## StanD

bigshot said:


> High frequency distortion. But no consumer audio gets close to 20 ns of jitter. Usually, it's 100 times less than that.


 
 Yes most of it is measured in ps, two orders of magnitude lower or better. HF as in above the human FR?


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## bigshot

Well, since it's never really been an issue, it's hard to say exactly where it would start getting annoying. The distortion would start messing stuff up at the top and work its way down as the jitter got worse and worse. It might not be noticeable for quite a ways past 20 ns actually, since it would only be affecting stuff up at the edge of hearing. All this is just theory because in home audio, even the cheapest components, jitter is completely inaudible.


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## StanD

bigshot said:


> Well, since it's never really been an issue, it's hard to say exactly where it would start getting annoying. The distortion would start messing stuff up at the top and work its way down as the jitter got worse and worse. It might not be noticeable for quite a ways past 20 ns actually, since it would only be affecting stuff up at the edge of hearing. All this is just theory because in home audio, even the cheapest components, jitter is completely inaudible.


 
 That's pretty much the way I see it despite the lamentations of some golden eared audiophiles.


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## Don Hills

stand said:


> Everytime I ask an "audiophile" that's doing the jitterbug, how many ps can a human perceive and how does it manifest in an observable audio artefact. I never get an answer, suddenly there is silence.


 

 The audibility of the jitter is the same as the audibility of their response...


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## lamode

stand said:


> Everytime I ask an "audiophile" that's doing the jitterbug, how many ps can a human perceive and how does it manifest in an observable audio artefact. I never get an answer, suddenly there is silence.


 
  
 Nooo, they are answering you but you need golden ears to hear them


----------



## StanD

lamode said:


> Nooo, they are answering you but you need golden ears to hear them


 
 I never thought, I'm deaf, I jitter at the thought. Must be why I cant tell the difference to hirez flac rips. Perhaps if I get golden ear rings it would help.


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## YtseJamer

ytsejamer said:


> New Steven Wilson is now available @ HDtracks
> 
> http://www.hdtracks.com/hand-cannot-erase-1




FLAC 24/96 stereo download with 2 bonus tracks

https://www.burningshed.com/store/stevenwilson/product/144/6495/


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## Krutsch

ytsejamer said:


> FLAC 24/96 stereo download with 2 bonus tracks
> 
> https://www.burningshed.com/store/stevenwilson/product/144/6495/


 

 I went with the BD version... I like the little slideshow that plays through the album. I was wondering if these were real photos from the story of Joyce Vincent, but Wikipedia shows otherwise. Still, heartbreaking story and interesting album concept.


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## Rob Stewart

lamode said:


> There was zero information on that page, just marketing fluff.


 
  
 Greetings, thank you for your feedback. I agree with you, I could have put more details in the top section for the consumer market. Most of the article is intended to help those looking to publish on iTunes, but, I have made a few updates to improve the content. As others have mentioned in this thread, there is a separate quality control stage for Mastered for iTunes. The primary goal with MFiT mastering is to leave enough room for the AAC encoder to do its job accurately and arrive at an AAC version that is as close as possible to the original source. 
  
 I hope this helps. Thanks again!
 Rob Stewart


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## bigshot

Leaving enough room for the AAC encoder just means lowering the overall volume level a touch. AAC encoders tend to bump up the volume slightly and that can push a track normalized up to 100% into clipping.


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## sonitus mirus

rob stewart said:


> Greetings, thank you for your feedback. I agree with you, I could have put more details in the top section for the consumer market. Most of the article is intended to help those looking to publish on iTunes, but, I have made a few updates to improve the content. As others have mentioned in this thread, there is a separate quality control stage for Mastered for iTunes. The primary goal with MFiT mastering is to leave enough room for the AAC encoder to do its job accurately and arrive at an AAC version that is as close as possible to the original source.
> 
> I hope this helps. Thanks again!
> Rob Stewart


 
  
 I don't get the point of having to master something specifically for iTune's AAC.  I have inexpensive software that can convert a CD or a high resolution file into AAC and it remains transparent.  Other than making sure the lossy version does not clip, there isn't any reason to use a slightly different EQ, reduce the loudness of only the peaks, or use some alternate dynamic processing that might change the sound characteristics. 
  
 Conspiracy theories aside, I'm concerned that mastering engineers will offer separate versions of their work when this is completely unnecessary.  I don't want a library of music available in high resolution that is slightly different than what is offered on iTunes or any other format that is currently used for any of the streaming music services.  
  
 Tidal music service already appears to behave in this manner.  I can hear a difference between a Tidal FLAC and a Tidal AAC 320 song; however, if I own the CD and rip this file to both FLAC and AAC 320, I can't hear a difference between the Tidal FLAC, my FLAC, or my AAC 320.


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## castleofargh

aren't the different masters just a way to lower the rights they have to pay to a studio by mastering the track themselves? the artist in many situations got money in advance to make an album, and most of the money made after release goes to the the guys who advertised for the album and own the rights (or am I wrong about all this?)
 so I would imagine a famous band recording in a very famous studio, mastered by a famous guy. if you remaster the original tape, you avoid the right for the mastering and save money maybe? or simply make a new version so the legal rights for the song are reconducted for X many years?
 anyway as you see my imagination goes with money, never with sound quality.


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## jcx

Recording Studio and final production release Mastering are often completely different hired companies today - the people paying them are the ones that own the copyrights, get to tell the Mastering engineers how much dynamic compression to use


----------



## Greenears

rob stewart said:


> Greetings, thank you for your feedback. I agree with you, I could have put more details in the top section for the consumer market. Most of the article is intended to help those looking to publish on iTunes, but, I have made a few updates to improve the content. As others have mentioned in this thread, there is a separate quality control stage for Mastered for iTunes. The primary goal with MFiT mastering is to leave enough room for the AAC encoder to do its job accurately and arrive at an AAC version that is as close as possible to the original source.
> 
> I hope this helps. Thanks again!
> Rob Stewart


 
  
 Reading between the lines of this non-statement statement you don't list your secondary or tertiary goals, or give any technical information.
  
 So the average reasonably technically savvy reader of this long and winding thread might instead conclude that what is really going on is that rights holders are adding a bit of bass EQ and dynamic range compression to make tracks on iTunes "hotter" since they know most listeners are on earbuds or low quality speakers.  Same thing has been done on radio for years.  So MFiT means you are trying to dial that back so that the AAC encoder doesn't clip.
  
 We can conclude this because hundreds of people on this forum have shown that any garden variety AAC256 encoder is capable of generating a file that is nigh on impossible to distinguish from the original CD in a blind ABX listening test.  So really MFiT is not needed for the reason you describe, unless the "original source" is not actually the same as the "original CD". 
  
 You could easily refute my supposition in one clear sentence, but you won't....


----------



## bigshot

Stuff gets remastered worse and worse because some hairbrained exec listens to a CD in his crappy car stereo on the freeway with the windows open and decides it needs more "oomph" and more "in your face". They send the tracks to a mastering engineer who shrugs his shoulders and adds another layer of compression on top of the four other times it's been sent down because it needed more "oomph" and more "in your face".
  
 Never attribute to malice that which can be explained by the stupidity of execs.


----------



## Rob Stewart

greenears said:


> You could easily refute my supposition in one clear sentence, but you won't....




You are correct. I won't. I am not here to argue. 

Cheers to you.

Rob Stewart


----------



## limpidglitch

greenears said:


> Reading between the lines of this non-statement statement you don't list your secondary or tertiary goals, or give any technical information.
> 
> So the average reasonably technically savvy reader of this long and winding thread might instead conclude that what is really going on is that rights holders are adding a bit of bass EQ and dynamic range compression to make tracks on iTunes "hotter" since they know most listeners are on earbuds or low quality speakers.  Same thing has been done on radio for years.  So MFiT means you are trying to dial that back so that the AAC encoder doesn't clip.
> 
> ...


 
  
 I don't know who this mr. Stewart is, but there seems to be some confusion as to what 'Mastered for iTunes' actually is.
  
 It is a purely technical requirement set by Apple, and is made out of two parts.
 One: the file sent to Apple should be of the highest resolution possible. If you have a 24/96 master, send that, and don't down-sample it.
 Two: That file shall have its levels set so that it won't clip as it is converted to AAC. This includes inter sample clipping.

 That is all. Really simple, I can't understand that this hasn't been done all along.
  
 They've also made a little software package to help in the process, and it works very well. The RoundTripAAC AU plug-in even has a simple ABX tester included.
  

  
  
 As you can see from the screen grab, it is clipping rather badly, even though the source file peaks at -0.3dBFS. I eventually had to attenuate it 3dB to make it come through clean. I also converted that lossless (CD) file to AAC 256 cvbr, with no attenuation, using iTunes, and the number of clipped samples matched what the plug-in predicted.
 This phenomenon of lossy compression leading to clipping isn't a new thing, and isn't restricted to AAC. When I converted that same file to LAME V1, it ended up even worse. Lastly I checked the iTunes Store version of the same song (The Trip by Still Corners, in case someone wondered), and lo and behold, no clipping! Seems like the system works.
  
 (other than this minor difference in amplitude, and of course some (inaudible) artefacts, the two commercial releases seems to be identical, as they should be)


----------



## dprimary

castleofargh said:


> aren't the different masters just a way to lower the rights they have to pay to a studio by mastering the track themselves? the artist in many situations got money in advance to make an album, and most of the money made after release goes to the the guys who advertised for the album and own the rights (or am I wrong about all this?)
> so I would imagine a famous band recording in a very famous studio, mastered by a famous guy. if you remaster the original tape, you avoid the right for the mastering and save money maybe? or simply make a new version so the legal rights for the song are reconducted for X many years?
> anyway as you see my imagination goes with money, never with sound quality.


 

 Remastering will not revert the rights back to the artist. Releasing "enhanced versions"  is how movie studios extend copyright beyond the already ridiculously long copyright terms we already have. The record companies are doing the same thing.


----------



## lamode

rob stewart said:


> The primary goal with MFiT mastering is to leave enough room for the AAC encoder to do its job accurately and arrive at an AAC version that is as close as possible to the original source.


 
  
 This doesn't seem to require a different master at all, but a simple attenuation as part of the conversion (in the event of inter-sample clipping).


----------



## castleofargh

lamode said:


> rob stewart said:
> 
> 
> > The primary goal with MFiT mastering is to leave enough room for the AAC encoder to do its job accurately and arrive at an AAC version that is as close as possible to the original source.
> ...


 

 yeah but "reencoded quieter to avoid conversion clipping"  feels like quality loss. when "remastered" makes the eyes shine ^_^.

 oh blessed sarcasm, only you can make me smile in those sad moments.


----------



## limpidglitch

lamode said:


> This doesn't seem to require a different master at all, but a simple attenuation as part of the conversion (in the event of inter-sample clipping).


 
  
 Technically that would be a new master. It's not all about EQ and compression, you know.
 Of course you could send that same attenuated file to the CD plant, but that wouldn't normally fly with the interests of maximum loudness. So two masters are made, only differing by a few dB.


----------



## lamode

limpidglitch said:


> Technically that would be a new master. It's not all about EQ and compression, you know.
> Of course you could send that same attenuated file to the CD plant, but that wouldn't normally fly with the interests of maximum loudness. So two masters are made, only differing by a few dB.


 
  
 The louder master should never have been made either, so you are not comparing apples with apples. A properly mastered 16/44 will convert just fine to AAC. Any "mastered for iTunes" label is nothing more than marketing nonsense.


----------



## limpidglitch

Sure, but the issue then is with the loudness war, and not Apple. To Apple this is a just a precaution, they don't want to sell damaged goods.
 If anything, Apple is actively against the whole loudness hysteria. At least going by how they manage playback levels through their radio service using perceived loudness, à la BS1770.


----------



## bigshot

lamode said:


> The louder master should never have been made either, so you are not comparing apples with apples. A properly mastered 16/44 will convert just fine to AAC. Any "mastered for iTunes" label is nothing more than marketing nonsense.


 

 Encoding using iTunes, I've had tracks that sounded just fine on CD clip when I did a straight conversion to AAC. I just lowered the level overall a few dB before converting and it encoded fine. AAC tends to boost the level a bit as it compresses.


----------



## Greenears

limpidglitch said:


> Sure, but the issue then is with the loudness war, and not Apple. To Apple this is a just a precaution, they don't want to sell damaged goods.
> 
> If anything, Apple is actively against the whole loudness hysteria. At least going by how they manage playback levels through their radio service using perceived loudness, à la BS1770.




I don't have a problem with the concept. But then why doesn't Mr. Stewart just provide some solid technical information. It seems like we are all just left guessing.


----------



## limpidglitch

He's a business man, förstås, selling a service.


----------



## Greenears

limpidglitch said:


> He's a business man, förstås, selling a service.


 Well there are more than one way to sell a service.


----------



## efeuvete

I think I have something to say here. This:
 Before any noise proccesing (Your text is very clear on this issue) the system must "take" samples of an analogic audio signal; And it is clear that the error in taking samples in 24 bits is 256 (2^8) times lesser than in 16 bits. In other words, 16 bits has a "minimun fixed" error of about 15,6 parts in a million and 24 bits only about 0,06 parts in a million.
 So, the point to me is: Can my ears notice a 15,6 part in a million error in every sample of every sounding sound wave?
 I woud say that, listening closely and carefully, yes, they can (And maybe more the higher the frecuency, which is logic because the higher the frecuency, the less number of samples per wave).
 Then, it is clear to me that, even not being sure of guessing a "16/24 blind test" in any sound situation, I'm sure I'm going to have a better experience all along 5 minutes of any music in 24bits than I'm going to have it in16 bits.
 And anyhow, last but not least,... God bless mp3, 128 Kbs for quite a good bunch amount of reasons!


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## headwhacker

^ I hope you read the thread from start before you post that comment. Nevertheless, welcome to head-fi


----------



## RRod

efeuvete said:


> I think I have something to say here. This:
> Before any noise proccesing (Your text is very clear on this issue) the system must "take" samples of an analogic audio signal; And it is clear that the error in taking samples in 24 bits is 256 (2^8) times lesser than in 16 bits. In other words, 16 bits has a "minimun fixed" error of about 15,6 parts in a million and 24 bits only about 0,06 parts in a million.
> So, the point to me is: Can my ears notice a 15,6 part in a million error in every sample of every sounding sound wave?
> I woud say that, listening closely and carefully, yes, they can (And maybe more the higher the frecuency, which is logic because the higher the frecuency, the less number of samples per wave).


 
  
 Everyone talks about extra levels from 24bits but doesn't bother to understand how they manifest themselves. The signal you get after picking a "level" is the original, un-rounded signal plus the rounding error that you get when you pick the level. At high amplitudes, that rounding error basically sounds like white noise. It's only when you get down to low signal amplitudes that the rounding error begins to manifest as harmonic distortion. The level of these errors is of course lower at 24bits, but that just results in a lower noise floor, and there is little music that gets anywhere the noise floor of even 16bits.
  
 What you can hear doesn't matter if you're letting knowledge of the files bias you.


----------



## cjl

efeuvete said:


> I think I have something to say here. This:
> Before any noise proccesing (Your text is very clear on this issue) the system must "take" samples of an analogic audio signal; And it is clear that the error in taking samples in 24 bits is 256 (2^8) times lesser than in 16 bits. In other words, 16 bits has a "minimun fixed" error of about 15,6 parts in a million and 24 bits only about 0,06 parts in a million.
> So, the point to me is: Can my ears notice a 15,6 part in a million error in every sample of every sounding sound wave?
> I woud say that, listening closely and carefully, yes, they can (And maybe more the higher the frecuency, which is logic because the higher the frecuency, the less number of samples per wave).
> ...


 
 Here's the thing: 15.6 parts per million works out to be (if properly dithered) a broad-spectrum noise floor (much like tape hiss) at about -110dBFS or so. That's not even close to audible unless you have the volume up so loud that a normal signal level would blow out your eardrums or your transducers (or both). Also, your logic about higher frequencies being affected worse shows why you should learn the math before just trying to guess about things like this - quantization error due to bit depth is frequency independent, and affects all frequencies equally (though you usually do shape the dither to have a higher noise level at high frequencies, simply because that is less audible).
  
 In addition, a blind test has no particular time limits. If you have a "better experience" over long periods of time, that is testable. Fast-switch, level matched tests have been the ones people are best able to discern small differences on in the past though, so chances are that over 5 minutes, you wouldn't even notice fairly large changes, much less subtle ones.


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## efeuvete

My only intention posting here has been trying to clarify a bit ideas regarding 24/16 bits differences, not arguing just for the sake of arguing; I mean, maybe 15,6 parts in a million can have a different impact on differents ears or even no impact at all (Though my "bias" does not accept this last option) but... 15,6 parts in a million rounding error are 15,6 parts in a million regardless:
 - How this rounding error is supossed to be manifesting (I mean, a complete wave with a 15,6 ppm rounding error to start from, in every one of its samples, does not make a distorted wave but a subtly different one).
 - The volume of your amp, because what math says here is that there is a 15,6 part in a million rounding error, being the wave as big as a thunder or as little as a butterfly flapping.


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## bigshot

efeuvete said:


> I mean, maybe 15,6 parts in a million can have a different impact on differents ears


 
  
 And if the wind is blowing just the right direction, a pig might fly.
  
 The truth is, there are plenty of controlled listening tests on this and to date, no one has been able to hear a -110dB noise floor when listening to music at normal volume levels. So there is no reason to believe that there are ears out there in the wild that perform any better than the plethora of ears that have already been tested.


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## efeuvete

Bigshot..."no one has been able to hear a -110dB noise floor when listening to music at normal volume levels"...
 It's you who talk about noise, not me. I'm talking about two different wave forms. I mean, Is a male voice a distorted female voice or viceversa?


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## bigshot

Do you think there is a human on earth that can hear distortion at -110dB?
  
 It's fine to entertain yourself with interesting math problems, but we are talking about real world sound here, not just theory. There are hundreds of listening tests of redbook vs high bit/sample rate audio. They all come out the same.


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## efeuvete

I insist: I'm not talking about discerning distorsion nor noise against clear sound, but about different wave forms. An Eflat 5th octave note on a recorder and that very same note on a metal flute are just different, none of them is more distorted than the other.


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## StanD

efeuvete said:


> I insist: I'm not talking about discerning distorsion nor noise against clear sound, but about different wave forms. An Eflat 5th octave note on a recorder and that very same note on a flute are just different, none of them is more distorted than the other.


 
 Whatever change to the waveform you propose would at best be too small for a human to discern. We are not comparing the timbre of flutes and recorders, that is so out of context and is not going to stick to the wall,especially at the 5th octave.


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## bigshot

Sorry wrong thread!


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## efeuvete

Good filtering work! That's all I can say here.


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## castleofargh

here we go again ^_^.
 "IZ better so I must hear it". when your flute is recorded, the samples are still only 1value each time. and what's recorded will be the flute and all the noises in the room. compared to the actual values we would get for the flute alone, it's different, and a much bigger difference than what 16 vs 24bit does. still you don't tell symphony orchestras to record instruments one by one in an anechoic chamber right? but if you really believe in your stuff you should demand it, the "improvement" would be massive compared to 16bit noise/error (call it as you like).
 so if a human is perfectly ok with noises unrelated to music that are pretty loud, why should he be concerned by lower noise that has been dithered to be even less noticeable? when you record a band you get still only one measure by sample, so are each instruments creating errors on the sound of the other one? that's the logic you're using efeuvete by saying that the quantization error change the sound of one instrument.
 just accept that we're dealing with soundwave and we record the mess that is the total of sounds. in that mess 16bit does impact the sound only at about -96db on the record. that's a fact and thinking it's bigger than that is just not understanding soundwaves.
  
 now does that noise matter? should we improve it? sure! why not, improvement is always ok. after you remove the noise in the studio from electric stuff, the noises from the human playing, the remastering process that will change the signal a good deal more than quantization noise, then the noise of your sound system that is not always below -96db, even if it is on measurements those measurements are done with specific output to get the best reading, then the noise in your room, then the noise of your own body. if we can reduce all those, then of course going to 24bit would become cool. getting a sense of proportions help finding out what matters and what doesn't. a small rain isn't my priority when I'm swimming.


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## RRod

efeuvete said:


> I insist: I'm not talking about discerning distorsion nor noise against clear sound, but about different wave forms. An Eflat 5th octave note on a recorder and that very same note on a metal flute are just different, none of them is more distorted than the other.


 
  
 By distortion we just mean "content that was not in the original signal." When you round sampled values to 24 or 16-bit values, you are adding distortion to the signal (255.9728 becomes 256; there's a distortion). What matters is what this rounding "sounds like", and as I said above, it sounds either like white noise when your signal amplitude is high, or it sounds like harmonic distortion when you signal values are low. Dither gets rid of the problem of the harmonic distortion at low levels, but adds broadband noise. Noise shaping fixes this by moving this noise up to the high frequencies. This has all been worked out.


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## cjl

efeuvete said:


> Bigshot..."no one has been able to hear a -110dB noise floor when listening to music at normal volume levels"...
> It's you who talk about noise, not me. I'm talking about two different wave forms. I mean, Is a male voice a distorted female voice or viceversa?


 

 We talk about noise because most 16 bit content is dithered, which removes the rounding error and replaces it with a constant, low-level noise floor. If you don't dither, there is no inherent noise floor, but you get quantization distortion instead (which will also be hugely below the signal and completely inaudible). It's covered pretty well in this video (the whole video is worth watching, but the part specifically addressing bit depth is around 8 minutes in): http://xiph.org/video/vid2.shtml


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## efeuvete

I'm afraid I'm going to be a solitary voice on this forum (Or at least, on this my one and only thread) but I must tell you this:
 The misunderstanding about my opinion seems to be related to the use of the word "noise" because in real life there are as much kind of "noises" as there are sounds, musical or not. Please consider you surely will be disapointed with a flute without its characteristc noise because that noise gives to the flute a good part of its personality, and, of course, it has nothing to do with white or pink noises.
 I think all of you say regarding "dither" and about all the processing of "noise inherent to digitalization" is perfect (And useful to my understanding of these things too) but I'm not talking about that noise (Nor about any noise for that matter), I'm talking about the diference between original sound and sampled sound. Whatever you do with sampled sounds, as sophisticated as it can be, you do it with... a sampled sound, which means, with a different sound of the original as your starting point; It can be no other way. And the more different when the less sampling frequency and the less deep (Here comes "my" 15,6 ppm difference).
 For instance, when RRod says "content that was not in the original signal" he forgets that you never know "the original signal" for the obvious reason that you only can know and manipulate a "sampled signal".
 And, please, do not consider me a kind of a sound maniac because I'm more than happy with my 128Kbs Mp3 music (Which, by the way, I found somehow difficult to differentiate from 320Kbs Mp3). All these ideas and tests came to me trying to keep the better digitized sound for one or two musical pieces which are special to me.


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## Avi

efeuvete said:


> I'm talking about the diference between original sound and sampled sound. Whatever you do with sampled sounds, as sophisticated as it can be, you do it with... a sampled sound, which means, with a different sound of the original as your starting point; It can be no other way. And the more different when the less sampling frequency and the less deep (Here comes "my" 15,6 ppm difference).


 
  
Nyquist–Shannon sampling theorem <-- (Link)


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## manbear

Yes, the sampled sound is technically different from the original. The part you're missing is that you can't hear the difference. Even if you could (you cant), it would be a moot point in practice because the noise floor of your amp (probably) and the ambient noise in your house (definitely) are going to be louder than the noise floor introduced by dithering.


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## efeuvete

avi said:


> Nyquist–Shannon sampling theorem <-- (Link)


 
  
 In the field of digital signal processing, the *sampling theorem* is a fundamental bridge between continuous signals (_analog_ domain) and discrete signals (_digital_ domain). Strictly speaking, it only applies to a class of mathematical functions whose Fourier transforms are zero outside of a finite region of frequencies
 (Wikipedia, rRed mark mine)


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## efeuvete

manbear said:


> Yes, the sampled sound is technically different from the original. The part you're missing is that you can't hear the difference. Even if you could (you cant), it would be a moot point in practice because the noise floor of your amp (probably) and the ambient noise in your house (definitely) are going to be louder than the noise floor introduced by dithering.


 

 "...The part you're missing is that you can't hear the difference..."
 So, then you go back to the begining, I mean, you go to the "Seemingly Universal Law" of "If you hear differences you're bias dreaming"
 Well, I don't follow that law more than I can follow any other opinion; mine for instance.
 (And please, I repeat, I'm not talking about noise, I'm talking about wave differences)


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## bigshot

efeuvete said:


> I'm afraid I'm going to be a solitary voice on this forum


 
  
 Welcome to Sound Science! We operate on different principles here. Here, we try to understand the acoustic and electrical principles, and try to avoid flowery descriptions and subjective error.
  


efeuvete said:


> you go to the "Seemingly Universal Law" of "If you hear differences you're bias dreaming"
> Well, I don't follow that law more than I can follow any other opinion; mine for instance.
> (And please, I repeat, I'm not talking about noise, I'm talking about wave differences)


 
  
 There are ways to prove whether a difference exists. You can analyze the waveform or do controlled listening tests. Both of those methods come to the same result. The differences between high bitrate and redbook all lie outside of the range of human hearing.


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## cjl

efeuvete said:


> I think all of you say regarding "dither" and about all the processing of "noise inherent to digitalization" is perfect (And useful to my understanding of these things too) but I'm not talking about that noise (Nor about any noise for that matter), I'm talking about the diference between original sound and sampled sound. Whatever you do with sampled sounds, as sophisticated as it can be, you do it with... a sampled sound, which means, with a different sound of the original as your starting point; It can be no other way. And the more different when the less sampling frequency and the less deep (Here comes "my" 15,6 ppm difference).


 
 And what you are failing to notice is that the sampled signal can be looked at as the original signal with a couple of modifications. Specifically, the sampled signal is the original signal with:
  
 1) Bandlimiting. This removes frequency content above ~20kHz, and is due to the sample rate alone. This has nothing to do with bit depth
 and
 2) Quantization with either:
    a)Dither. This adds a low-level, uncorrelated noise floor (very much like tape hiss). The result will be the original signal plus a low-level his more than a hundred dB below the main signal level (for 16 bit).
    or
    b) Quantization distortion. If dither isn't used, you do get those slight rounding errors you mentioned before. This is called quantization distortion, since it is caused by quantizing the signal level into discrete bins. This tells you exactly how much difference there is between the original signal and the quantized version. This is worse (from an audibility perspective) than dither is, but even so, the level of distortion added is well below what has ever been shown to be audible (at least when 16 bits are used).
  
 You don't have to sit here guessing about it - all of this has been extensively mathematically studied, quantified, and characterized. We don't need to wonder what it would sound like, nor do we have to wonder about how it will affect the signal.


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## bigshot

cjl said:


> 2) Quantization with either:
> a)Dither. This adds a low-level, uncorrelated noise floor (very much like tape hiss). The result will be the original signal plus a low-level his more than a hundred dB below the main signal level (for 16 bit).
> or
> b) Quantization distortion. If dither isn't used, you do get those slight rounding errors you mentioned before. This is called quantization distortion, since it is caused by quantizing the signal level into discrete bins. This tells you exactly how much difference there is between the original signal and the quantized version. This is worse (from an audibility perspective) than dither is, but even so, the level of distortion added is well below what has ever been shown to be audible (at least when 16 bits are used).


 
  
 And if he is interested in how those things work, he can click through the top link in my sig file and get the straight dope and downloadable examples.


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## manbear

efeuvete said:


> "...The part you're missing is that you can't hear the difference..."
> So, then you go back to the begining, I mean, you go to the "Seemingly Universal Law" of "If you hear differences you're bias dreaming"
> Well, I don't follow that law more than I can follow any other opinion; mine for instance.
> (And please, I repeat, I'm not talking about noise, I'm talking about wave differences)




I'm not invoking any universal law, and I'm less quick to call bias than most people here. We are talking about a specific sound you claim to hear, and it's physically impossible for you to do so. Imagine if I said I could hear the hearbeat of a person across the street. Its just too quiet, below the limits of human hearing.

The only wave differences are in the form of inaudibly quiet noise. You clearly don't understand the technical concepts here. Learn how the Nyquist theorem and the process of dither apply to your question.


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## efeuvete

I do not like to be boring so, I think this one will be my last post on this thread:
  
 "In the field of digital signal processing, the sampling theorem is a fundamental bridge between continuous signals (analog domain) and discrete signals (digital domain). _Strictly speaking, it only applies to a class of mathematical functions whose Fourier transforms are zero outside of a finite region of frequencies"_
 (Nyquist Theorem, Wikipedia; italics mine)
  
 ... And natural musical sounds do not observe that condition up to the detail required by the theorem.


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## RRod

efeuvete said:


> I'm afraid I'm going to be a solitary voice on this forum (Or at least, on this my one and only thread) but I must tell you this:
> 
> For instance, when RRod says "content that was not in the original signal" he forgets that you never know "the original signal" for the obvious reason that you only can know and manipulate a "sampled signal".
> And, please, do not consider me a kind of a sound maniac because I'm more than happy with my 128Kbs Mp3 music (Which, by the way, I found somehow difficult to differentiate from 320Kbs Mp3). All these ideas and tests came to me trying to keep the better digitized sound for one or two musical pieces which are special to me.


 
  
 You forget that when our ears pick up a signal, we don't know what the original is either. The ADC takes the signal from the mic, bandlimits it, then quantizes it. Thus the stored signal has the same frequency content as the bandlimited signal, plus the error due to quantization. None of this requires knowing which signal you're going to get into the mic, but it does require the bandlimiting.


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## cjl

efeuvete said:


> I do not like to be boring so, I think this one will be my last post on this thread:
> 
> "In the field of digital signal processing, the sampling theorem is a fundamental bridge between continuous signals (analog domain) and discrete signals (digital domain). _Strictly speaking, it only applies to a class of mathematical functions whose Fourier transforms are zero outside of a finite region of frequencies"_
> (Nyquist Theorem, Wikipedia; italics mine)
> ...


 
 Hence the bandlimiting.
  
  
 (Note that this is a completely separate thing from the quantization you were talking about before, and has nothing to do with the bit depth).


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## manbear

efeuvete said:


> I do not like to be boring so, I think this one will be my last post on this thread:
> 
> "In the field of digital signal processing, the sampling theorem is a fundamental bridge between continuous signals (analog domain) and discrete signals (digital domain). _Strictly speaking, it only applies to a class of mathematical functions whose Fourier transforms are zero outside of a finite region of frequencies"_
> (Nyquist Theorem, Wikipedia; italics mine)
> ...




Now you can hear sounds outside of 20-20k Hz too. Your hearing extends infinitely... Good one.


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## bigshot

efeuvete said:


> "In the field of digital signal processing, the sampling theorem is a fundamental bridge between continuous signals (analog domain) and discrete signals (digital domain). _Strictly speaking, it only applies to a class of mathematical functions whose Fourier transforms are zero outside of a finite region of frequencies"_
> (Nyquist Theorem, Wikipedia; italics mine)


 
  
 Now, we jump from bitrate to sampling rate... OK.
  
 Thankfully, the region of frequencies perfectly covered by 16/44.1 are slightly broader than the region of frequencies the human ear can hear. It's the same with bitrate. The noise floor is below the range where human ears can hear it. Redbook audio exceeds human hearing on every aspect of hearing. Perfect sound for human ears.


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## efeuvete

cjl, manbear, bigshot, RRod:
 Natural noise does not go inside nor outside any frecuency band because noise is not a periodic function.


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## manbear

efeuvete said:


> cjl, manbear, bigshot, RRod:
> Natural noise does not go inside nor outside any frecuency band because noise is not a periodic function.




The only way your ears hear is by measuring frequency and amplitude. All sound contains this information. All the sound you can hear is inside the band from 20 to 20k Hz.

Are you talking about some kind of noise that's not a sound?


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## lamode

efeuvete said:


> Can my ears notice a 15,6 part in a million error in every sample of every sounding sound wave?


 
  
 No, it's noise like tape hiss, but much much quieter. In almost any real life situation, it is inaudible. See the video in my sig for a demonsration of the noise.


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## analogsurviver

> You forget that when our ears pick up a signal, we don't know what the original is either. The ADC takes the signal from the mic, bandlimits it, then quantizes it. Thus the stored signal has the same frequency content as the bandlimited signal, plus the error due to quantization. None of this requires knowing which signal you're going to get into the mic, but it does require the bandlimiting.


 
 Enough is enough.
  
 While you guys are arguing 24 vs 16 etc, that is - to a point - excusable.
  
 If you are going to listen to a live feed from the mike, level equalized to within 0.1 dB or less, and ANY bandlimited signal ( be it due to ADC, PCM, DSD or miniaturized Martians, additional amplifier(s), etc - you name it )
  
*THERE WILL BE AUDIBLE DIFFERENCE. Period.*
  
 That is what @efeuvete trying to tell you all with his note played on flute, violin, bamboo or kazoo - each of these instruments have their *TIMBRE *- and if band limited (for any reason ), they will sound less than real. And all of you can weave your answers and "proofs" in whatever way, it will not change that fact a single bit.
  
 I read "by the time all human breath is removed, along with room noise, etc, etc, etc" - a domain made possible by the PCM and DSP and being unknown in analog days - you are going to remove any vestiges of real sound heard live. And if THAT is your reference (for ABXing etc ) - I , unfortunately, can see how redbook or even AAC 256 may seem "enough" for you. A flute played and heard live DOES have it noise - and so does the human being playing it. If an absolute precision and attempt at perfection in music is desired - look no further than Frank Zappa and his work on Synclavier. No way ANY human being will ever be able to play this ultra fast precisely written music with anything even approaching the aplomb machines are capable of. But it is artificial and beyond the scope what is normally associated with "music" as we know it.
  
 There is absolutely NO bandlimiting required in recording - at least not in theory. Band limiting is there solely as a consequence of the imperfect microphones and whatever is following them in the chain - and not any theorethical reason. It is OK to try to limit the digitized signal to some reasonable file size (and ultimately cost ) - but it is NOT OK to say that bandlimited signal is "perfect" and "audibly transparent". 
  
 If anyone of you will come up with the old warhorse "you can not compare live feed to (whatever passed trough any other additional device except (perfect?) wire )" -
 because, you obviously can not go back in time and compare exactly the same sound x-times to satisfy the ABX statistics - I will say some _*VERY *_direct word.
  
 Above does require one to have an extremely clean signal path - I have yet to hear a mixing desk that I would consider using. And it goes in this style all the way to the end. And it is CLEARLY audible even on $25 IEMs ( like Xiaomi Piston 2.0 ) - not to mention serious stuff. 
  
 In horse terms, redbook is about managing broken in domesticated horses - not some wild herd leader who would rather die than being saddled. And most of you would rather see this proud wild horse dead - if he refuses to get saddled and wearing blinds - limping along calmly, according to the prescriptions of the redbook beat.
  
 I am not saying there are no possibilities to utilize the redbook better - there are. But trying to supress anything and anybody who dares to feel and think beyond redbook in these pages is already bordering on - disgusting.


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## lamode

efeuvete said:


> I'm afraid I'm going to be a solitary voice on this forum (Or at least, on this my one and only thread) but I must tell you this:
> The misunderstanding about my opinion seems to be related to the use of the word "noise" because in real life there are as much kind of "noises" as there are sounds, musical or not. Please consider you surely will be disapointed with a flute without its characteristc noise because that noise gives to the flute a good part of its personality, and, of course, it has nothing to do with white or pink noises.


 
  
 Consider this... a flute is playing, then a bass guitar joins in. When the bass guitar joins in, the waveform becomes radically different because all sounds add together to produce one waveform, regardless of how many instruments are playing.
  
 The quantization error you talk about is equivalent to noise being played in the background of the flute track, but so quiet that it's inaudible. It doesn't affect the sound of the flute itself. Clearer now?


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## lamode

analogsurviver said:


> If you are going to listen to a live feed from the mike, level equalized to within 0.1 dB or less, and ANY bandlimited signal
> ...
> *THERE WILL BE AUDIBLE DIFFERENCE. Period.*
> ....
> But trying to supress anything and anybody who dares to feel and think beyond redbook in these pages is already bordering on - disgusting.


 
  
 What's bordering on disgusting is the incessant promotion of falsehoods and ignorance.
  
 If you do the reverse and play only the content above 22kHz (the difference between the live feed and BL feed) then guess what you hear... nothing, because there is no audible difference.
  
 Nada, zip, nichts, rien, ничего, 何も, नथिंग


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## manbear

analogsurviver said:


> Enough is enough.
> 
> While you guys are arguing 24 vs 16 etc, that is - to a point - excusable.
> 
> ...




All I'm getting from this post is that you think you can hear high frequencies north of 20 Khz. Is that accurate or am I missing something here?


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## cjl

efeuvete said:


> cjl, manbear, bigshot, RRod:
> Natural noise does not go inside nor outside any frecuency band because noise is not a periodic function.


 
 All natural noise that isn't inside the frequency band from about 0-20kHz or so is imperceptible to humans, so it's irrelevant anyways.


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## efeuvete

lamode, just to think about this, not to argue: The error I'm talking about (Which I repeat I'm not sure I can ear it, I just say that in some cases I "feel" differences) will happen too when an instrument A is strongly loud over other instrument B because in that case, the error will be mainly audible on instrument A.


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## limpidglitch

efeuvete said:


> I think all of you say regarding "dither" and about all the processing of "noise inherent to digitalization" is perfect (And useful to my understanding of these things too) but I'm not talking about that noise (Nor about any noise for that matter), I'm talking about the difference between original sound and sampled sound. Whatever you do with sampled sounds, as sophisticated as it can be, you do it with... a sampled sound, which means, with a different sound of the original as your starting point; It can be no other way. And the more different when the less sampling frequency and the less deep (Here comes "my" 15,6 ppm difference).
> For instance, when RRod says "content that was not in the original signal" he forgets that you never know "the original signal" for the obvious reason that you only can know and manipulate a "sampled signal".
> And, please, do not consider me a kind of a sound maniac because I'm more than happy with my 128Kbs Mp3 music (Which, by the way, I found somehow difficult to differentiate from 320Kbs Mp3). All these ideas and tests came to me trying to keep the better digitized sound for one or two musical pieces which are special to me.


 
  
 We know no other sounds than sampled sounds.
 A pressure wave travels through air, in your ear, and causes the tympanic membrane to vibrate. That vibration is amplified by the malleus, incus, and stapes, and ends up in the cochlea.
 In the cochlea the vibrations will elicit standing waves across a long narrow and tapered membrane, and ciliated cells situated alongside it will sense these vibrations. If the vibrations are big enough the cells will send a chemical signal to a nearby neuron, which in turn will send a single, discrete voltage pulse onward to the central nervous system. Thus are sounds sampled. And it's bandwidth limited as well, as you'd expect. A typical auditory nerve fiber has a refractory period of a few hundred µs*, meaning the rate is at best a few kHz. Thankfully they are massively parallelized, making the resolution we experience possible.
 These discrete signals eventually end up in the primary auditory cortex where they are converted into frequencies and amplitudes occurring at specific times. With the help of auxiliary cortical areas these magnitudes are shaped into information we can use (a middle aged man, my dad, standing roughly five metres behind me on my left is asking me if I want an ice cream. He might be getting a bit inpatient).


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## lamode

efeuvete said:


> lamode, just to think about this, not to argue: The error I'm talking about (Which I repeat I'm not sure I can ear it, I just say that in some cases I "feel" differences) will happen too when an instrument A is strongly loud over other instrument B because in that case, the error will be mainly audible on instrument A.


 
  
 It seems that English is not your first language, which makes this conversation a little tricky. Anyway... if I understood you correctly, you are talking about a digital recording of instrument A and B playing at the same time and A is much louder? This will make no difference. The quantization error will be the same amplitude regardless of how many instruments are playing or how loud. The quantization error is limited to half of the least significant bit (LSB).


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## efeuvete

Yes, despite my english, you understood me correctly; And if I understand you, what you say is the same thing I say: The error would be equaly there on a flute solo or on the flute+bass duo as a proportional sum of each instrument error.


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## cjl

No, the error doesn't get larger just because more instruments are playing. The error is the same (and inaudibly small) for a recording of a solo violin or the whole symphony orchestra.


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## efeuvete

cjl, except for "inaudibly small" you are saying the same thing I'm saying


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## StanD

efeuvete said:


> cjl, except for "inaudibly small" you are saying the same thing I'm saying


 
 Are you saying that you are capable of hearing that half bit and have done so?


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## efeuvete

What I'm saying is that I'm not sure 15,6 parts per million sampling error is unaudible.


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## StanD

efeuvete said:


> What I'm saying is that I'm not sure 15,6 parts per million sampling error is unaudible.


 
 No it's not, unless you are test equipment.


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## efeuvete

test equipment doesn't hear


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## StanD

efeuvete said:


> test equipment doesn't hear


 
 And meat popsicles (humans) cannot measure or hear a half bit in this context,


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## efeuvete

I'm not sure, that's all.


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## lamode

efeuvete said:


> I'm not sure, that's all.


 
  
 If you watched the video in my signature, as I suggested, you would stop worrying about this


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## castleofargh

efeuvete said:


> I'm not sure, that's all.


 

 lucky for us, an opinion still doesn't change a fact. I suggest you hold on your great math achievement about errors(didn't even try to see if it's the right value, that's how important it isn't to the topic), and try to read more about audio, how it works in our sound systems and how it works in our ears.
 or just look how the best headphone or speaker you can buy will still create "errors" very much louder than the 16bit quantization. yet you don't worry about that too much. but hey why worry about the loud obvious stuff when we can speculate about super low level noises? that's what you say right!
  
 aside from that, listen to whatever you like. we're only contesting the audibility here. not the fact that it can measurably improve something.


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## bigshot

efeuvete said:


> I'm not sure, that's all.


 

 That is a wonderful realization. Because until you admit you don't know, you won't take the time to do the research to actually know. We've been down that road and we are happy to share resources with you if you want to be sure. Too many people argue from ignorance and don't really want to know.


----------



## stv014

It can be mathematically proven that if white noise with a triangular distribution and at a peak level of +/-1 LSB is mixed to the input signal before quantization by rounding to the nearest integer, then, as long as clipping is avoided and the dither noise is uncorrelated to the input, the quantized output is guaranteed to meet the following conditions:
- any DC input, even if it is a fractional value, will produce the same output level on average, in other words, "steps" in the level are statistically eliminated
- the quantization error will always be white noise at an RMS level of 0.5 LSB, so it is not audibly modulated by the input
Dithered quantization therefore basically sounds like white noise hiss in analog equipment, and its perceived level is determined by the sample resolution and the dither/noise shaping algorithm used, as well as the sample rate to some extent (as a higher sample rate moves more of the same overall noise energy outside the audio band).
  
This video shows the effects of quantization and dithering in practice.
  
 Edit: never mind, I suspect *efeuvete* will not be convinced by any arguments, so I leave this debate.


----------



## efeuvete

castleofargh: ... "I suggest you hold on your great math achievement about errors(didn't even try to see if it's the right value, that's how important it isn't to the topic), and try to read more about audio, how it works in our sound systems and how it works in our ears"... (Way too proud comment, to my understanding)
 bigshot... "Too many people argue from ignorance and don't really want to know"... (Just in case, that's not my case)
 stv014: ...If white noise with a triangular distribution and at a peak level of +/-1 LSB is mixed to the input signal before quantization... (Yes, but what I say is about natural musical instrument noise, not about "Synthetic" noise)
  
 So, after some two dozens of posts received, here they are three more reasons to write again: But please, before concluding anything, try to follow me; I'll try to be the shorter I can and I'll try do the best of my limited english knowledge.
 Firstly I must clarify that all I'm going to write here is related to the difference between a real musical sound and a "nude" sampled wave, so, the first informatión which goes to any HiFi equipment, therefore before any kind of proccesing inside starts. Here I go:
  
 1) Any REAL musical sound has two audible parts: The main and first one is a (more or less) periodic function of any form (Let's call it A) and a second one: The natural instrument noise (Not ambient noise), a non periodic function, so without any wave form, just pressure changes following not known rule (Let's call it part B). And is important to say here that, nevertheless, this noise has "its own unknown" rules; for instance, it is different at different frecuencies of part A.
  
 2) This part B, natural instrument noise, is of course part of the perceived instrument sound and an important one too. An example could be the noise sounds inherent to a transverse flute or to a human voice (I remember that the old synthetizers, the ones before "samplers era", always added "The supposed best noise they can do" to simulate both the difference between its pure sinusoidal waveform and the "not so pure sinusoidal" real flute waveform, and to simulate too this real flute natural noise)
  
 3) Nyquist theorem can only be applied to part A, but it has nothing to say regarding part B because part B does not follow any (known) rules. So, what we can get of part B (In CD quality level) is no more that the numeric information the equipment can get with its 16 bits depth and its 44.100 samples per second. There are no "tricks" here to deduce more information. Then, I must consider that 24 bits is 15,6 parts por million more precise than 16 bits and that 96.000 Hz means more than twice the samples of 44.100 Hz (For instance, a sampled normal 700 Hz wave at 44.100 Hz has about 63 samples per wave and sampled at 96.000 Hz has about 137 Samples per wave)
  
 4) Despite of all this, 16 bits and 44100 Hz can get "a good lot" of natural instrument noise and so, is difficult to hear the difference between 16-44.100 and 24-90.000. Nevertheless, the test which makes me think that the difference is audible has been done with recordings of the sixties which when I was young I used to listen in vinyl more than weekly. That's why I'm not sure at all I can hear easily that difference in recordings new to me.
  
 And that's all. These are my arguments to say "I'm not sure there is no difference" and really I didn't expect that only a member (Analogsurviver) out of some six on the forum, does consider worthwhile what I've said.
  
 Best regards to all forum members.


----------



## RRod

efeuvete said:


> 3) Nyquist theorem can only be applied to part A, but it has nothing to say regarding part B because part B does not follow any (known) rules. So, what we can get of part B (In CD quality level) is no more that the numeric information the equipment can get with its 16 bits depth and its 44.100 samples per second. There are no "tricks" here to deduce more information. Then, I must consider that 24 bits is 15,6 parts por million more precise than 16 bits and that 96.000 Hz means more than twice the samples of 44.100 Hz (For instance, a sampled normal 700 Hz wave at 44.100 Hz has about 63 samples per wave and sampled at 96.000 Hz has about 137 Samples per wave)


 
  
 No one is arguing that 24/96 can get a more accurate recreation of a transient; they are arguing that the improvement is inaudible.
  
 Take two 700Hz sine wave samples, one at 16/44.1 and one at 24/96, and see if you can tell a difference in a blind test. That's the kind of thing we do around here.


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## efeuvete

But my point is that natural musical sounds have natural instrument noise which should not be "killed"; Natural sounds are not sine wave forms; Not even perfect periodical sounds of any wave form


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## lamode

efeuvete said:


> But my point is that natural musical sounds have natural instrument noise which should not be "killed"; Natural sounds are not sine wave forms; Not even perfect periodical sounds of any wave form


 
  
 The errors in a 16/44 waveform are so small that they are around -100dB, and therefore inaudible in the real world. No arguments about the nature of an instrument are relevant here.


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## efeuvete

"arguments about the nature of an instrument are not relevant here" sounds to me like a strange dogma in a forum about musical sounds.


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## manbear

1) It's irrelevant that B is non-periodic. B is still a sound and it has some kind of wave form. If it didnt, then it wouldnt be a sound at all.

3) Nyquist applies to B in the exact same way it applies to A. There is no difference. Both are sounds with frequencies in the finite range of human hearing.


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## efeuvete

Manbear, from my first post up to this one I had no intention of disturbing anybody but I'm afraid this idea is not correct:
  
 "Nyquist applies to B in the exact same way it applies to A. There is no difference. Both are sounds with frequencies in the finite range of human hearing"


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## FFBookman

Of course it's audible in stereo-mixed music. You try to isolate things with mono test tones and you wonder why you can't capture it.  Sad, weird, to be a scientist with such limited tools.
  
 Rent a recording studio, they need the business, and do some 24bit audio tests there. You will easily hear it in the pan steps, in the depth and clarity of reverb decay, in the presentation of the soundstage as a clear, 3D, thing. 
  
 Good signal chain will reveal much, even at 16/44, that most people don't even hear. Even on cheap speakers. 
  
 At 24/44 and higher sampling rates, there is more data which provides a more nuances rendering of the music. You will never capture this in the lab.
  
 Science must adapt and grow to learn how to perform proper music sound quality tests. 
  
  
 If you hear 24bit music played on a competent system and you come away thinking that people don't need to hear that, you are anti-music in my book.  You can be pro-music and pro-science, ya know.


----------



## FFBookman

lamode said:


> The errors in a 16/44 waveform are so small that they are around -100dB, and therefore inaudible in the real world. No arguments about the nature of an instrument are relevant here.


 

 Talking about waveform pictures are we?
  
 Recorded music is all about the stereo presentation, timing cues, and the timbre of the tones. Your waveforms tell you nothing important as far as sound quality. They are a pretty side effect of your ADC or DAC chip. A visual signpost to the wrong way of thinking about sound.


----------



## StanD

ffbookman said:


> Talking about waveform pictures are we?
> 
> Recorded music is all about the stereo presentation, timing cues, and the timbre of the tones. Your waveforms tell you nothing important as far as sound quality. They are a pretty side effect of your ADC or DAC chip. A visual signpost to the wrong way of thinking about sound.


 
 The final recorded music that is played back is the sum of all instruments playing as a single value at one point in time. The deviation of 1/2 bit at 16 bits of resolution is simply below our ability to hear. Jacking that up to 24 bits does not improve the situation. This has been proven time and time again. If you wish to go hires, nobody here is stopping you, however, we need not go along the ride.


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## FFBookman

stand said:


> The final recorded music that is played back is the sum of all instruments playing as a single value at one point in time. The deviation of 1/2 bit at 16 bits of resolution is simply below our ability to hear. Jacking that up to 24 bits does not improve the situation. This has been proven time and time again. If you wish to go hires, nobody here is stopping you, however, we need not go along the ride.


 

 I'm not trying to take you on a ride. I'm answering the accusation that 24bit PCM audio is a "myth" or "snake oil".  It's neither. It's been used in production and in listening to classical music for a while now. You need the right DAC and analog stage to really show it's advantages, but most mid-level and up gear will render it well enough.
  
 My mastering engineer says that he can use all the bits, all the extra space, to render a better sound at any resolution, so he recommends delivery in 24/192 and will immediately take it to 24/192 using his high end converters regardless of delivery format. He's also been delivering final masters to Apple's mastered for iTunes program as 24bit PCM for years now. Apple currently down samples and then sells lossy mp3 and sometimes "lossless" 16/44 alac  (Not lossless, since it's been down sampled, but hey why get accuracy in the way of marketing talk.)
  
 I'm not sure it's headroom alone, I think there's something in the dithering that alters the transients and the timing cues. The soundstage suffers after dither, and the overall tone and number of "voices" you hear clearly is diminished.  None of this is measurable or can be drawn on a screen as a waveform, so it goes ignored by some, disregarded by the ignorant.


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## FFBookman

Can anyone here PROVE to me that dither is perfectly stereo matched?
  
 Dither is fuzz. Is that fuzz exactly the same on each channel? 
  
 Even if it's mathematically the same on each channel, can you prove to me that my detection of it doesn't alter either the soundstage or the timbre of the recording?
  
 But "sound science" has no way to measure such things, which is why it's defeated in the practical world all the time. 24bit audio sounds better in the studio before dither is applied. Everyone knows this. It's a compromise for storage space/bandwidth.  Always has been.


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## RRod

ffbookman said:


> Of course it's audible in stereo-mixed music. You try to isolate things with mono test tones and you wonder why you can't capture it.  Sad, weird, to be a scientist with such limited tools.
> 
> Rent a recording studio, they need the business, and do some 24bit audio tests there. You will easily hear it in the pan steps, in the depth and clarity of reverb decay, in the presentation of the soundstage as a clear, 3D, thing.
> 
> ...


 
  
 I've heard it on HD800s with a V200 and a Bifrost, and couldn't ABX 24 versus 16 on anything. In fact, I can get many older tracks down to 14 and 12 bits and still not be able to tell a difference in a blind test. Either you've done this kind of thing for yourself or you haven't. If you have, then fine; if you haven't, well there you go.


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## StanD

ffbookman said:


> I'm not trying to take you on a ride. I'm answering the accusation that 24bit PCM audio is a "myth" or "snake oil".  It's neither. It's been used in production and in listening to classical music for a while now. You need the right DAC and analog stage to really show it's advantages, but most mid-level and up gear will render it well enough.
> 
> My mastering engineer says that he can use all the bits, all the extra space, to render a better sound at any resolution, so he recommends delivery in 24/192 and will immediately take it to 24/192 using his high end converters regardless of delivery format. He's also been delivering final masters to Apple's mastered for iTunes program as 24bit PCM for years now. Apple currently down samples and then sells lossy mp3 and sometimes "lossless" 16/44 alac  (Not lossless, since it's been down sampled, but hey why get accuracy in the way of marketing talk.)
> 
> I'm not sure it's headroom alone, I think there's something in the dithering that alters the transients and the timing cues. The soundstage suffers after dither, and the overall tone and number of "voices" you hear clearly is diminished.  None of this is measurable or can be drawn on a screen as a waveform, so it goes ignored by some, disregarded by the ignorant.


 
 All unproven anecdotes. Real tests show otherwise. We can all agree that higher resolution is useful for the recording/processing/mixdown processes, however, that's where it ends. Better DACs and Amps cannot help with human limitations.


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## FFBookman

No one listens to me, that's cool. Who am I? Let me try to restate the crux of my argument.
  
 You can't ABX test for music quality.
  
 New tests, or a refined testing process, is needed to accurately reflect music sound quality.
  
 ABX test is poison to the way we consume music.
  
 ABX is useless for attempting to track music sound quality enjoyment because it changes the way we process our music.
  
 ABX makes garbage results for music tests. They should be thrown out with the old kitty litter.
  
  We don't listen to music to compare it to a different quality.  Our emotional response to music changes immediately when trying to pick or detect a quality.
  
 Our listening MODE changes the second you try to introduce such a test. We are, at that point, listening in a different way, and reacting in a different way than we do when enjoying our own music normally.
  
 ABX tests tell you if people can pick something out from a rapidly changing environment.
  
 It does not say which one sounds better in the long run. It doesn't track which one they would rather own if given the ability to "test drive" properly.
  
 I can and have failed an ABX test trying to determine between 8bit and 16bit. I know how these things work. It's very hard to detect and very easy to confuse people's perceptions given the parameters of this wholly unsuited tests. You can play the same file 5x in a row and people will usually think they heard different versions. It's crud from the test, or more accurately, the proof that the test does not match our musical listening abilities. 
  
 If you truly consider yourself a man of science you will understand these criticisms of the testing environment and method.  ABX results always show confusion so they are always used by the side that claims less quality.
  
 The burden of proof is not on me to prove what I and millions more hear, the burden of proof is on your scientific principles to design a proper test to reflect (prove) reality.


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## StanD

I see we're heading back to mysticism.


----------



## RRod

ffbookman said:


> If you truly consider yourself a man of science you will understand these criticisms of the testing environment and method.  ABX results always show confusion so they are always used by the side that claims less quality.
> 
> The burden of proof is not on me to prove what I and millions more hear, the burden of proof is on your scientific principles to design a proper test to reflect (prove) reality.


 
  
 If you consider yourself a man of science you'd agree on the importance of blind testing. And the burden of proof is on you: there are millions more who can't hear the difference between 192k mp3s and the original wav.


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## bigshot

My friend, you are in the WRONG forum to be treating audio as a philosophy, not a science. You can parade around and make anti-intellectual statements that might get serious consideration from people who don't know better, but around here, it really won't get you very far. It just serves to make you seem like a troll.


----------



## FFBookman

rrod said:


> If you consider yourself a man of science you'd agree on the importance of blind testing. And the burden of proof is on you: there are millions more who can't hear the difference between 192k mp3s and the original wav.


 

 BS.  You really think there's millions of people that can't hear lossy?   My grandma can hear it. Everyone can hear it.  Maybe their music choice or their playback systems or their hearing deficiencies  mask it, but you are really on a forum called "head-fi" saying that there's no actual difference between an mp3 and a 16/44 ?  wow.
  
 Proof you only care about horrible, lossy, degraded things. Welcome to 1996. Sure it sounds the same, sure.
  
 What kind of horrible playback gear do you have to think that an MP3 is the same thing?  Geez. It's like a time machine up here. No scratch that, no one ever thought they sounded as good.
  
 You know why they throw out data don't you? 
  
 No free lunch.  128k bitrate doesn't sound like 256k. That sounds worse than 320k. That sounds worse than 600-1400k bitrate (CD). That sounds worse than 24bit files which can push nitrates as high as 4000k. They sound amazing, like the band is playing in your room. With real amps and instruments. 
  
 I am not sure if there's anything beyond 24/192 with PCM, I know no one is recording at that resolution, and I don't think you need all that to reproduce analog tape masters accurately.  But I know without a doubt there's plenty of information beyond 16/44 that they have to downsample (remove) and cover up with fuzz (dither).


----------



## RRod

ffbookman said:


> BS.  You really think there's millions of people that can't hear lossy?   My grandma can hear it. Everyone can hear it.  Maybe their music choice or their playback systems or their hearing deficiencies  mask it, but you are really on a forum called "head-fi" saying that there's no actual difference between an mp3 and a 16/44 ?  wow.
> 
> Proof you only care about horrible, lossy, degraded things. Welcome to 1996. Sure it sounds the same, sure.
> 
> ...


 
  
 Yes, there are millions of people who can't hear it. That's why it exists! I can make out 192 but not much higher, and as I said I tried these things out on an HD800+V200+Bifrost (how much more $$ you want for a good system?) Right now I use 256k AAC, and on classical music that you try to use an example of needing 24bit but that actually doesn't. I'm also a statistician, so I know all about throwing out data. The millions of people who can't distinguish various mp3 bitrates vs. wav are not outliers. Keep fooling yourself all you want.


----------



## bigshot

ffbookman said:


> You really think there's millions of people that can't hear lossy?   My grandma can hear it. Everyone can hear it.


 
  
 Would you like to back that statement up? I have a lossless file with nine randomly ordered samples of three lossy codecs... Fraunhofer, LAME and AAC at 192, 256 and 320... plus one lossless sample mixed in. Ten music samples altogether. Would you like to do a listening test to determine whether you can identify which is which? You can take as much time as you like, and listen on your own equipment. The only rule is that you have to rank them based on listening, not by opening the file in a sound editing program.
  
 Easy, right? Even your grandmother could do it, right?
  
 Let me know if you prefer FLAC or ALAC format.


----------



## StanD

ffbookman said:


> :
> My grandma can hear it
> :
> No free lunch.


 
 How old is your grandma that she can still hear above 10 kHz?
 So what have you been eating for lunch that we haven't?


----------



## FFBookman

bigshot said:


> My friend, you are in the WRONG forum to be treating audio as a philosophy, not a science. You can parade around and make anti-intellectual statements that might get serious consideration from people who don't know better, but around here, it really won't get you very far. It just serves to make you seem like a troll.


 

 I hear ya, I know i came onto this thread throwing down.
  
 I'm trying to help you folks in "sound science" by getting you to think bigger. Your love of ABX tests, test tones, and waveform drawerings are hurting the rest of us. You are being exploited in the name of convenience over quality.
  
 In the first 30 years of the digital revolution, maybe convenience was more important.
  
 Who wanted 3 24bit songs on a CD, playing on a player where prices started at $1k?   Who wanted to fill their iPod with 30 24bit songs that barely sounded any better since Apple spends $5 on audio in iDevices?  Not me.
  
 But now, go buy yourself a proper DAP an start loading it up with files. Put on your MP3's, why not?  Rip some CD's again, wow, you won't believe how much better they sound. Then buy a couple things at 24bit and after digesting it for a few weeks come back here and tell me how you can't hear any difference at all.  Then I'll say "sorry" for you and move on, I suppose.
  
  
 Are any of you working on new types of listening tests?  I'm interested in that.
 Are any of you doing research into stereo differences in various DAC's?
 Are any of you attempting to develop measurements of soundstage and instrument timbre?  
  
 These are very difficult problems, this 3D listening. Much harder than waveform math. Anyone up to the challenge?


----------



## RRod

ffbookman said:


> These are very difficult problems, this 3D listening. Much harder than waveform math. Anyone up to the challenge?


 
  
 I get more 3D from binaural recordings and HRTF DSP than your extra bits. So you can keep those.


----------



## FFBookman

bigshot said:


> Would you like to back that statement up? I have a lossless file with nine randomly ordered samples of three lossy codecs... Fraunhofer, LAME and AAC at 192, 256 and 320... plus one lossless sample mixed in. Ten music samples altogether. Would you like to do a listening test to determine whether you can identify which is which? You can take as much time as you like, and listen on your own equipment. The only rule is that you have to rank them based on listening, not by opening the file in a sound editing program.
> 
> Easy, right? Even your grandmother could do it, right?
> 
> Let me know if you prefer FLAC or ALAC format.


 
  
 OH a challenge!  Me like.  FLAC is fine, I don't care. I bet it shows that no one can tell anything!  Great test. 
  
 Do you care that your results are useless because your ears can't do direct comparison tests with music quality?
  
 I have a challenge for you.  Get a new DAP, preferably a PonoPlayer, and just listen to it. Don't test, don't math, don't prod, poke, just listen. You will get the Pono Smile and forget all about math for a precious few seconds. Then you can go back to proving that just didn't happen.
  
  
 Back to your test so you don't accuse me of totally ignoring it - I suppose if I'm allowed to review them at my leisure, on my own systems, and IF I knew the material well, I could see some value in the results. Point me to the files and I'll do my best to give you feedback on them.
  
 But I don't care about lossy codecs.
 I really don't. I don't care which one sounds better. I have no reason for lossy anymore. This was far more important to me in 2000 than 2015. I can put 128gb on my fingernail, I don't need lossy.
  
 If you could re-do this test with 24bit and downsampled 16bit, and really put the work into the provenance of the files, I am a willing participant.
  
 We need newly released 24bit files, then we need to downsample them using various dithering algorithms and converters. The low end can be reflected in apple products, using built in converters and free software. The high end can be done by a mastering engineer with real converters. If we need a middle, an interface like a focusrite will have converters better than an apple product.
  
 The end result would be 3-4 16bit versions of an original 24bit file. That's what the record labels do. Strip them of labels, include with the original 24bit, and let the tester pick which one they like the best after living with them for a few days. It would show you if people could pick out 24bit on their system, and which kind of dithering algorithm is the favorite.


----------



## FFBookman

rrod said:


> I get more 3D from binaural recordings and HRTF DSP than your extra bits. So you can keep those.


 

 Yeah I don't disagree there. More than 1 way to skin a cat. I hate that saying.
  
 That's why I'm interested in MQA. Even though they are using mp3-like lossy compression at the encoding level, they claim they have tweaked what frauhauffer did to increase transient and spatial accuracy. That could be good for stereo sound.
  
 They are also using the signed DAC to decode on a tighter clock than PCM DAC's can. This also *could* show some serious improvement in the presentation.  
  
 Bad thing is this new encoding is incompatible with current DAC's, all 10 billion of them in the world. So its possibly a dead product, but I have some theories on how forces are trying to replace PCM with MQA.


----------



## bigshot

ffbookman said:


> Do you care that your results are useless because your ears can't do direct comparison tests with music quality?


 
  
 Are you saying that you can't discern the sound quality difference between lossy and lossless? I'm offering you a way to prove that you can hear a difference. If you say that you can't, then we agree. I don't think you can hear a difference either.
  
 I've taken the time to put together an useful comparison. If you are interested and appreciate the effort being made to help you understand what you hear, then great. I'm happy to share the test with you. If you aren't interested in finding out the truth and won't agree to the very simple terms of the test, then no one needs to waste their time.
  
 There's no point arguing. No one learns anything by arguing. They learn by observing and analyzing their observations. I am politely offering you an opportunity to do that.
  
 I have this lossy test all prepared and ready to go. If you show that you can easily discern the differences in codecs and bitrates, I'm sure someone would be happy to assemble a similar 16 vs 24 bit test for you.


----------



## FFBookman

stand said:


> How old is your grandma that she can still hear above 10 kHz?
> So what have you been eating for lunch that we haven't?


 
  
 Standard fail #1 -- music and sound quality is about way more than frequency.  Think outside of the waveform. Frequency is but one parameter of sound, why can't you people understand that?
  
 My grandma is 99. I honestly have never tested mp3's on her.  I was exaggerating, you caught me. She likes polka cassettes.
  
 But my 65 year old father in law and the rest of his friends he's played it for all hear 24bit improvement on the pono player. One guy heard it in 5 seconds, said "it's like the whole thing". His other friend, also in his 60's, said "it's like being in a recording studio". 
  
 I am aware they are reacting partly to the signal chain ,that's why i'm interested in a 24bit vs 16bit downsample shootout.


----------



## RRod

ffbookman said:


> Standard fail #1 -- music and sound quality is about way more than frequency.  Think outside of the waveform. Frequency is but one parameter of sound, why can't you people understand that?
> 
> My grandma is 99. I honestly have never tested mp3's on her.  I was exaggerating, you caught me. She likes polka cassettes.
> 
> ...


 
  
 I tried out 24 vs 16 on every one of the free Linn 24-days-of-Xmas downloads. No luck on anything. Pick one you want and I'll put up comparison versions tonight or tomorrow.


----------



## FFBookman

bigshot said:


> Are you saying that you can't discern the sound quality difference between lossy and lossless? I'm offering you a way to prove that you can hear a difference. If you say that you can't, then we agree. I don't think you can hear a difference either.
> 
> I've taken the time to put together an useful comparison. If you are interested and appreciate the effort being made to help you understand what you hear, then great. I'm happy to share the test with you. If you aren't interested in finding out the truth and won't agree to the very simple terms of the test, then no one needs to waste their time.
> 
> ...


 

 I can hear the loss, of course. Everyone can hear the artifacts.  Is this really 1997, did I walk into a time warp? Are you really believing that lossy is not actually lossy, that it makes no difference?  
  
 I will try the test but since ABX tests don't really prove anything other than FUD I'm prepared for my results to fall into your argument. I'll risk that and respond honestly. I have been fooled before, I heard an MP3 coming off of youtube through my living room that I thought was CD for a few minutes, and I had trouble picking out 8bit through a browser on my laptop. 
  
 Point me to the great BigShot Lossy Test and I'll play along. But I'm far more interested at pushing the bar forward, not fooling people into thinking they don't miss what used to be there.


----------



## FFBookman

If you want to do the real test of 24bit masters verse 16bit downsamples I can offer up some materials and connections. I have about 10 albums in open FLAC @ 24bit, can do the low-end Apple conversion and the mid-range focusrite conversion to 16bit, and know someone with the converters to do the high-end versions.  I don't know that he will work for free though, I'd have to ask if he's interested in helping in the name of science.
  
 I'd love to throw in DSD but it introduces different mastering sessions as a variable. I'd rather stick with a single source file at 24bit and test multiple down samples of the same source.


----------



## StanD

ffbookman said:


> Standard fail #1 -- music and sound quality is about way more than frequency.  Think outside of the waveform. Frequency is but one parameter of sound, why can't you people understand that?
> 
> My grandma is 99. I honestly have never tested mp3's on her.  I was exaggerating, you caught me. She likes polka cassettes.
> 
> ...


 
 You missed the point about accuracy as it is not just about frequency. You also refuse to accept that meat popsicles have their limitations.
 Anyone, like your father in-law, hearing things without a proper test and under your suggestions is not going to give any valid results. You should listen more carefully to Inspector Harry Callahan, "A man's got to know his limitations."


----------



## bigshot

Test sent.


----------



## FFBookman

ffbookman said:


> If you want to do the real test of 24bit masters verse 16bit downsamples I can offer up some materials and connections. I have about 10 albums in open FLAC @ 24bit, can do the low-end Apple conversion and the mid-range focusrite conversion to 16bit, and know someone with the converters to do the high-end versions.  I don't know that he will work for free though, I'd have to ask if he's interested in helping in the name of science.
> 
> I'd love to throw in DSD but it introduces different mastering sessions as a variable. I'd rather stick with a single source file at 24bit and test multiple down samples of the same source.


 

 My 24bit albums I'd offer up for test material (we should pick something we all know and love)
  
 Rolling Stones - Sticky Fingers
 The Who - Who's Next
 The Cars - The Cars
 Sam Cooke - Portrait of a Legend
 Led Zeppelin - III
 Led Zeppelin - Physical Graffiti
 Slave - Slave
 Red Hot Chili Peppers - Blood Sugar Sex Magik
 Bootsy's Rubber Band - Ahh The Name Is


----------



## RRod

ffbookman said:


> If you want to do the real test of 24bit masters verse 16bit downsamples I can offer up some materials and connections. I have about 10 albums in open FLAC @ 24bit, can do the low-end Apple conversion and the mid-range focusrite conversion to 16bit, and know someone with the converters to do the high-end versions.  I don't know that he will work for free though, I'd have to ask if he's interested in helping in the name of science.
> 
> I'd love to throw in DSD but it introduces different mastering sessions as a variable. I'd rather stick with a single source file at 24bit and test multiple down samples of the same source.


 
  
 I offered what I offered. If you have a friend who can do conversions then he can help you out with your own tracks. I'd bet that most tracks wouldn't even need dither, so no fancy algorithms necessary; just chop off the bits.


----------



## StanD

rrod said:


> I offered what I offered. If you have a friend who can do conversions then he can help you out with your own tracks. I'd bet that most tracks wouldn't even need dither, so no fancy algorithms necessary; just chop off the bits.


 
 What no rounding? 
	

	
	
		
		

		
			





 You're gonna hear about that.


----------



## RRod

stand said:


> What no rounding?
> 
> 
> 
> ...


 
  
 I know, I've moved on from anti-scientific to sacrilegious!


----------



## bigshot

ffbookman said:


> My 24bit albums I'd offer up for test material (we should pick something we all know and love)
> 
> Rolling Stones - Sticky Fingers
> The Who - Who's Next
> ...


 

 Those are all analogue recordings, aren't they? And some of those aren't particularly well recorded. Shouldn't it be a native high bitrate recording?


----------



## StanD

stand said:


> What no rounding?
> 
> 
> 
> ...


 
  
  


rrod said:


> I know, I've moved on from anti-scientific to sacrilegious!


 
 Perhaps the extra bits can be dealt with by an exorcism.


----------



## FFBookman

bigshot said:


> Those are all analogue recordings, aren't they? And some of those aren't particularly well recorded. Shouldn't it be a native high bitrate recording?


 

 Yep. I think that's important. 
  
 See, this "sound science" goes both ways.  Producers, mixing engineers, and mastering engineers adjust for the marketplace. The marketplace is the dominant format and the mainstream playback devices of the era.
  
 When CD took over in the 80's it allowed for far more overall volume and subwoofer kick without skipping the needle. Producers took advantage of this. Music production changed. If you believe redbook has any weaknesses, producers also adjusted for those.
  
 I find it's soundstage suspect, it's center undefined, and it's mid-high's completely unnatural and crispy. Its sub-lows are OK but it's mid-lows (like bass guitar or standup) are weak.  So producers worked ways to compensate for this.  Also, are you really going to hear a 24bit advantage if all your plug-ins operate at 16bit, if all the samples are 16/44 and if most of your digital instruments all create 16/44?  No.
  
 When MP3 took over 15 years go it happened again, and modern muscs seems more alive, louder, and more present, but it's all due to creative ways to cope with bandwidth restriction. There's a lot less data there, trying to sound like more.
  
 Going back to analog material allows us to hear music without any digital trickery, without any digital compensation for other degradation. It allows to hear real instruments, real analog reverbs, real room delays, real instrument timbre, analog EQ, and of course fine, professional microphones and recording techniques.
  
 My entire belief is that 24bit PCM sounds closer to the analog tape masters than does 16bit PCM. So when I do listening tests I try to keep my material pure and obvious, and avoid modern material with it's multiple layers of deception.


----------



## castleofargh

ffbookman said:


> Science must adapt and grow to learn how to perform proper music sound quality tests.


 
 like your uncontrolled ones that are the only way to get a result that agrees with you? you're sure that's how science should work. we should forget blind testing and rent a studio to make sighted listening? wow that's got to be the only real way, the method is so clean and objective.
 please don't talk about what is science itself, a non stop questioning and evolving way to understand stuff, while anything you write screams the denial of method and the enforcement of your own bias.
  
 you look for anything that is stereo to prove your idea is right, found the idea that dither might not be perfectly stereo and hang onto that idea without any concern for the fact that we're talking sound at -96db. plz tell me what sound system you use that has such a perfect left/right balance that you notice the -96db difference while everything else is perfectly aligned? we all want those devices and speakers/headphones(no really we do!). you're fighting the ant while carefully avoiding the elephant in the room.
 but 16bit has to be the culprit because you decided it.
 blind test doesn't say the same? well blind tests are wrong.
 measurements say you're wrong? there is more than measuring a soundwave in audio!
 you always have an excuse not to look at the truth.
 the album was made with a few microphones and the guys most likely recorded one after the other in the same studio room on the same chair at the same place(at least for most modern stuff), the mix generated the synthetic stereo placement and probalby 2 microphones with 2different signals and different phases from where they are, where mixed to get one instrument to sound "better". but when you don't get all the great soundstage, it has to be 16bit. and in fact you can't know because you reject blind testing. so you're just stating your faith in 24bit while giving us advices about science. amazing!
  
  
 now efeuvete talks about the recording of 2 different sounds as if it would make any difference to the movement of the membrane in a microphone. as if anything recorded was more than 1pressure at one moment at one point. as if the air could contract and expand instantly without limits and let you hear stuff like a pure square wave.
 of course anything in the air is at least a portion of a wave. it has to else how would it travel? so yes we could record sounds that are not waves(generated on a computer for example). and then yes maybe we would need another format and 16/44 or even PCM would fail. but no speaker or headphone could make those sounds without massive distortions, because the air would desagree a little^_^. the result is that we hear waves(or portions of them) because of how the sound travels.  and that's why digital audio works.
 but again it's as if recording in 16bit was the reason why sound is messed up. not microphones, mix, sound system, our ears. no all this is 100% perfection with all the care of phase alignment from A to Z and perfect retrieval of uncorrupted data thanks to magic unicorns.
 who cares about all the stuff that are worst than 16bit? who cares that people fail to actually hear it? efeuvete has his idea and build up a theory that matches it, based on his very own understanding of audio. I see no science there.
  
 analogsurviver also is by association anti 16/44, as he is pro anything way up. but this time it's for the ultrasounds and getting all the music by getting all the frequencies. rejecting the fact that most humans old enough to not believe in santa will most likely not hear them. so his foundation theory comes from something false. otherwise he does have a very rational reasoning. by that I mean that if ultrasounds were beneficial to music, then most of his arguments would stand. that's really not the case for efeuvete and you IMO.
  
 so here are 3 people saying that 16bit is audibly inferior but not even for the same reasons ... and of course without an scienctific method to show it.
 while the "we don't listen to music we just read graphs" secret society, made of most of the others in this topic(not arguing that numbers gives me truth, that's BS. just saying we agree for the same reasons at least), are saying 16bit vs 24bit isn't, or at least shouldn't be audible in normal listening. and we all explain it with:
 1/ the difference between the 2 resolutions is at about -96db and below, everything above is the same. when it isn't, it has to do with the sound system not with 16bit.
 2/ people repeatedly fail to identify differences this low in blind tests.
 therefore people repeatedly fail to identify 16bit => we say the errors from dithered 16bit aren't audible to people on usual albums at usual listening levels. and we're still waiting for people to show us otherwise in a convincing way (because we do believe in the scientific method and would accept a repeatable method showing we were wrong).
  
 we could give you the subconscious part where we still get it but we don't know we do. but if it is so, how did you know 24bit sounded better? see the slight flaw with that idea? if it's not conscious, well it isn't. and if the effect could manifest on a conscious level, then people wouldn't fail blind tests so consistently. it's one or the other but we can't bend reality just to fit in that 16bit is audibly bad.
  
  


> Standard fail #1 -- music and sound quality is about way more than frequency.  Think outside of the waveform. Frequency is but one parameter of sound, why can't you people understand that?


 
 ok.
 a waveform includes amplitude and timing, there is zero need to think outside of a waveform. frequency obviously is just one parameter.  on a record we get rid of the timing and only keep the amplitude because it will be up to our replay device to put the samples at regular intervals.
 the result is again a waveform with time and amplitude(else it wouldn't be a waveform). if there is more to sound than this, then it is on no record on earth because a microphone doesn't know better than making a waveform in volts.
 and stereo is but 2 waveforms at the same time. if you have timing errors and you do at least in a measurable way, 16bit isn't your problem in that regard.
 just try to get rid of your tunnel vision and look at the entire audio process and the individual precision of all parts. pretending that -96db noise is the reason for bad stereo rendering is simply ludicrous. just by looking at right/left balance of a headphone you should understand how insignificant dither is.


----------



## StanD

castleofargh said:


> :
> now efeuvete talks about the recording of 2 different sounds as if it would make any difference to the movement of the membrane in a microphone. as if anything recorded was more than 1pressure at one moment at one point. as if the air could contract and expand instantly without limits and let you hear stuff like a pure square wave.
> :


 
 C'mon haven't you heard of the Quantum Microphone. That's one of Prof. Emmet Brown's finest works.


----------



## manbear

efeuvete said:


> Manbear, from my first post up to this one I had no intention of disturbing anybody but I'm afraid this idea is not correct:
> 
> "Nyquist applies to B in the exact same way it applies to A. There is no difference. Both are sounds with frequencies in the finite range of human hearing"




Sorry buddy. It's exactly correct. There is nothing magical or different about the instrument noise you're talking about. Sound is sound and it's as simple as that. All of it can be sampled the same way (by your ears too, not just by ADCs).

If you don't believe me, explain how your ears can physically hear these sounds that are impossible to sample.


----------



## manbear

ffbookman said:


> Standard fail #1 -- music and sound quality is about way more than frequency.  Think outside of the waveform. Frequency is but one parameter of sound, why can't you people understand that?
> 
> My grandma is 99. I honestly have never tested mp3's on her.  I was exaggerating, you caught me. She likes polka cassettes.
> 
> ...




All that your ears physically pick up are frequency and amplitude. There is nothing more to sound. Stereo image, timing cues, etc. are simply the result of having two ears at once.

What is so special about Pono? You talk about it as if its the only way to experience 24 bit music

Have you considered that your 24 bit recordings are using different masters? Have you tried compressing one to 16 bits?


----------



## lamode

ffbookman said:


> Recorded music is all about the stereo presentation, timing cues, and the timbre of the tones. Your waveforms tell you nothing important as far as sound quality.


 
  
 If the difference between the original waveform and the reconstructed waveform is under -100dB then the difference is inaudible in the real world. In other words, the recorded sound is inaudibly different from the original. I would say that's an absolute indicator of SQ.


----------



## lamode

ffbookman said:


> I'm not trying to take you on a ride. I'm answering the accusation that 24bit PCM audio is a "myth" or "snake oil".  It's neither. It's been used in production


 
  
 This has been raised and answered at least 20 times in this thread already. 24 bit is used in production for other reasons.


----------



## lamode

ffbookman said:


> Can anyone here PROVE to me that dither is perfectly stereo matched?


 
  
 Inaudible in left channel.
 Inaudible in right channel.
  
 Yep, seems perfectly matched to me.


----------



## lamode

ffbookman said:


> You can't ABX test for music quality.
> 
> ABX test is poison to the way we consume music.


 
  
 Complete nonsense.
  
 If you can't hear a difference between A and B then by definition they are inaudibly different.
  
 If there was a "night and day" difference between 24 and 16 bit audio, people would have no problem passing an ABX test.
  
 Your argument is bordering on ludicrous.


----------



## lamode

ffbookman said:


> No free lunch.  128k bitrate doesn't sound like 256k. That sounds worse than 320k. That sounds worse than 600-1400k bitrate (CD). That sounds worse than 24bit files which can push nitrates as high as 4000k.


 
  
 Again, a nonsense argument. Increased data rates may MEASURE better but at some point they will stop SOUNDING better to the human ear as the limits of human hearing are surpassed.
  
 All that matters to humans is the minimum format required to achieve that. Any more data is just wasted bandwidth.


----------



## StanD

lamode said:


> Again, a nonsense argument. Increased data rates may MEASURE better but at some point they will stop SOUNDING better to the human ear as the limits of human hearing are surpassed.
> 
> All that matters to humans is the minimum format required to achieve that. Any more data is just wasted bandwidth.


 
 Unfortunately we are not all *Übermensch. *






 Neil Young must be very disappointed of me.


----------



## lamode

ffbookman said:


> Then buy a couple things at 24bit and after digesting it for a few weeks come back here and tell me how you can't hear any difference at all.  Then I'll say "sorry" for you and move on, I suppose.


 
  
 He already told you that he couldn't hear the difference between 14 and 16 bit in a test, so you can deliver your apology now.
  


ffbookman said:


> These are very difficult problems, this 3D listening. Much harder than waveform math.


 
  
 More nonsense. I'm beginning to sense the presence of a troll.


----------



## lamode

ffbookman said:


> Do you care that your results are useless because your ears can't do direct comparison tests with music quality?


 
  
 LOL, you do realize that you suggested a few posts ago that listening to 16 and 24 bit files side-by-side would reveal a big difference, right?


----------



## lamode

ffbookman said:


> But my 65 year old father in law and the rest of his friends he's played it for all hear 24bit improvement on the pono player. One guy heard it in 5 seconds, said "it's like the whole thing". His other friend, also in his 60's, said "it's like being in a recording studio".


 
  
 Oh good grief, this isn't funny any more. Please stop trolling.


----------



## StanD

lamode said:


> Oh good grief, this isn't funny any more. Please stop trolling.


 
 You should see him carrying on in the I love Pono thread.


----------



## lamode

ffbookman said:


> I find it's soundstage suspect, it's center undefined, and it's mid-high's completely unnatural and crispy. Its sub-lows are OK but it's mid-lows (like bass guitar or standup) are weak.  So producers worked ways to compensate for this.  Also, are you really going to hear a 24bit advantage if all your plug-ins operate at 16bit, if all the samples are 16/44 and if most of your digital instruments all create 16/44?  No.


 
  
 You forgot to add that the relative humidity goes up, a distinct 4kHz peak is audible, and that if there is a full moon, the treble sound-stage is compromised, and if the power generation station which feeds the recording studio is nuclear, the mid bass slows down, (except for the first Wednesday of every month of course!) and that if anyone in the recording studio is wearing a digital watch, there will be a high-frequency pulsating noise which only golden-ears can hear, and if the mixing desk sits on a carpeted floor, the bass will sound woolly, and if there is blue wallpaper in the studio, vocals will sound more forward, and if the singer is weather leather shoes, there will be less sibilance, and if the recording studio is more than 200 metres above sea level, the midrange will sound lifeless, and if any of the microphone jacks are chrome plated, there will be a high frequency ringing, and if there is an airport near the recording studio, there will be more jitter, and if there is a polypropylene capacitor anywhere in the circuit, even digital circuit, the dynamic range will be reduced by 0.7dB.... Did I miss anything?


----------



## StanD

lamode said:


> You forgot to add that the relative humidity goes up, a distinct 4kHz peak is audible, and that if there is a full moon, the treble sound-stage is compromised, and if the power generation station which feeds the recording studio is nuclear, the mid bass slows down, (except for the first Wednesday of every month of course!) and that if anyone in the recording studio is wearing a digital watch, there will be a high-frequency pulsating noise which only golden-ears can hear, and if the mixing desk sits on a carpeted floor, the bass will sound woolly, and if there is blue wallpaper in the studio, vocals will sound more forward, and if the singer is weather leather shoes, there will be less sibilance, and if the recording studio is more than 200 metres above sea level, the midrange will sound lifeless, and if any of the microphone jacks are chrome plated, there will be a high frequency ringing, and if there is an airport near the recording studio, there will be more jitter, and if there is a polypropylene capacitor anywhere in the circuit, even digital circuit, the dynamic range will be reduced by 0.7dB.... Did I miss anything?


 
 Yes, Iran has secret plans to make portable nuclear reactors to power Ponos.


----------



## efeuvete

manbear said:


> Sorry buddy. It's exactly correct. There is nothing magical or different about the instrument noise you're talking about. Sound is sound and it's as simple as that. All of it can be sampled the same way (by your ears too, not just by ADCs).
> 
> If you don't believe me, explain how your ears can physically hear these sounds that are impossible to sample.


 

 I'm not a math expert at all but I've always thought that this condition in Nyquist theorem:
 "Strictly speaking, it only applies to a class of mathematical functions whose Fourier transforms are zero outside of a finite region of frequencies"
 means that Nyquist theorem only applies to periodic functions (Natural noise is not a periodic function, it has no waveform).
 Of course, I can be wrong.


----------



## RRod

efeuvete said:


> I'm not a math expert at all but I've always thought that this condition in Nyquist theorem:
> "Strictly speaking, it only applies to a class of mathematical functions whose Fourier transforms are zero outside of a finite region of frequencies"
> means that Nyquist theorem only applies to periodic functions (Natural noise is not a periodic function, it has no waveform).
> Of course, I can be wrong.


 
  
 It means that you have to have band-limited signals. Since they are bandlimited, they cannot also have finite support in the time domain (i.e. be time-limited). But our signals, of course MUST be time-limited, because we don't capture an infinitely long sound wave. Thus we cannot *perfectly* bandlimit; there will always be some high frequency content we don't actually get out and that gets aliased. But we can make that of such small magnitude as to not matter.


----------



## Greenears

lamode said:


> You forgot to add that the relative humidity goes up, a distinct 4kHz peak is audible, and that if there is a full moon, the treble sound-stage is compromised, and if the power generation station which feeds the recording studio is nuclear, the mid bass slows down, (except for the first Wednesday of every month of course!) and that if anyone in the recording studio is wearing a digital watch, there will be a high-frequency pulsating noise which only golden-ears can hear, and if the mixing desk sits on a carpeted floor, the bass will sound woolly, and if there is blue wallpaper in the studio, vocals will sound more forward, and if the singer is weather leather shoes, there will be less sibilance, and if the recording studio is more than 200 metres above sea level, the midrange will sound lifeless, and if any of the microphone jacks are chrome plated, there will be a high frequency ringing, and if there is an airport near the recording studio, there will be more jitter, and if there is a polypropylene capacitor anywhere in the circuit, even digital circuit, the dynamic range will be reduced by 0.7dB.... Did I miss anything?



Crystals

The machine that goes "ping"

Directional silver ethernet cables


----------



## richie037

Those audiopholes believe it's this simple, if it was, i would allready havy done it years ago. One audioidiot site said lossless mp3 and flac...... There ain't no such thing as lossless dig audio. Losless when you trigger it to 0 or 1 say it all and the endresult may sound better, but don't you think they just colored the sound in the dsp . But themain purpose everybody try to do is recreate to cutaway harmonics when started using comprised files... Buy a good audio enhancer and chand the sound until you like it or play records on a good system instead of digital succery. No , the real good cd player until 1990 could play a cd way better then these days, with there toriod and fill metall bi servo drived lasers. But via a pc your sounds sucks if you use the standard lg things. For 2 thousand your cd player sound perfect.... no cambridge stuff....But a soundcart of mine has a phono input and i don't here a lot difference with bitrates, and it ain't a technics 1200, just a normal axiom system and with a preamp between it using line input with another card, you keep a glassy sound but a converter which you buy ain't a studio converter, so gives your upscaling a warmer sound, and as it sounds better, who cares. But the mastering used for the better cd's and vinyl is done by using the tape..... Downside..... the vinyl ain't cheap and to call it hi-fi..... First they overpower cd's and vinyl and when mastered right on heavy vinyl it's hifi...... , def. of hifi.


----------



## StanD

Huh?


----------



## stv014

> Originally Posted by *FFBookman* /img/forum/go_quote.gif
> 
> Going back to analog material allows us to hear music without any digital trickery, without any digital compensation for other degradation. It allows to hear real instruments, real analog reverbs, real room delays, real instrument timbre, analog EQ, and of course fine, professional microphones and recording techniques.
> 
> My entire belief is that 24bit PCM sounds closer to the analog tape masters than does 16bit PCM. So when I do listening tests I try to keep my material pure and obvious, and avoid modern material with it's multiple layers of deception.


 
  
 Some older analog recordings may sound better than some newer digital ones, but that is mainly because of better recording and mastering. If you converted them to CD format, and compared that to the original under controlled conditions (which you will probably not do), you would likely have major difficulties telling them apart.
  
 24 bit PCM only sounds closer to analog tape than 16 bit PCM if the analog tape does not have much worse noise floor than the lower resolution PCM, which I guess is not something that frequently happens in practice. Additionally, the music itself needs to have enough dynamic range to make it possible to hear the noise floor in the first place.
  
 In any case, if you do not mind increasing the size of PCM audio by 50%, and you want to do this primarily for better sound stage and imaging, then those bits are better spent on adding an extra channel for depth information than making quantization noise that is already inaudible in 99% of cases even more inaudible.


----------



## stv014

ffbookman said:


> I can hear the loss, of course. Everyone can hear the artifacts.
> ...
> I will try the test but since ABX tests don't really prove anything other than FUD I'm prepared for my results to fall into your argument.


 
  
 Of course, everyone can _imagine_ the artifacts, as long as they know which sample is which.
  
 What do you think about tests where the samples are known to be identical, but the listener is lied to that they are different, and an "audible" difference is detected in a sighted comparison ? If this is possible (and has indeed been done already, for example with a fake switchbox that always selects the same input), then how do you know for sure without ABX testing that your perceived 16 bit or even high bitrate lossy artifacts are real ?


----------



## kraken2109

richie037 said:


> Those audiopholes believe it's this simple, if it was, i would allready havy done it years ago. One audioidiot site said lossless mp3 and flac...... There ain't no such thing as lossless dig audio. Losless when you trigger it to 0 or 1 say it all and the endresult may sound better, but don't you think they just colored the sound in the dsp . But themain purpose everybody try to do is recreate to cutaway harmonics when started using comprised files... Buy a good audio enhancer and chand the sound until you like it or play records on a good system instead of digital succery. No , the real good cd player until 1990 could play a cd way better then these days, with there toriod and fill metall bi servo drived lasers. But via a pc your sounds sucks if you use the standard lg things. For 2 thousand your cd player sound perfect.... no cambridge stuff....But a soundcart of mine has a phono input and i don't here a lot difference with bitrates, and it ain't a technics 1200, just a normal axiom system and with a preamp between it using line input with another card, you keep a glassy sound but a converter which you buy ain't a studio converter, so gives your upscaling a warmer sound, and as it sounds better, who cares. But the mastering used for the better cd's and vinyl is done by using the tape..... Downside..... the vinyl ain't cheap and to call it hi-fi..... First they overpower cd's and vinyl and when mastered right on heavy vinyl it's hifi...... , def. of hifi.


 
 wat


----------



## lamode

richie037 said:


> Those audiopholes believe it's this simple, if it was, i would allready havy done it years ago. One audioidiot site said lossless mp3 and flac...... There ain't no such thing as lossless dig audio. Losless when you trigger it to 0 or 1 say it all and the endresult may sound better, but don't you think they just colored the sound in the dsp . But themain purpose everybody try to do is recreate to cutaway harmonics when started using comprised files... Buy a good audio enhancer and chand the sound until you like it or play records on a good system instead of digital succery. No , the real good cd player until 1990 could play a cd way better then these days, with there toriod and fill metall bi servo drived lasers. But via a pc your sounds sucks if you use the standard lg things. For 2 thousand your cd player sound perfect.... no cambridge stuff....But a soundcart of mine has a phono input and i don't here a lot difference with bitrates, and it ain't a technics 1200, just a normal axiom system and with a preamp between it using line input with another card, you keep a glassy sound but a converter which you buy ain't a studio converter, so gives your upscaling a warmer sound, and as it sounds better, who cares. But the mastering used for the better cd's and vinyl is done by using the tape..... Downside..... the vinyl ain't cheap and to call it hi-fi..... First they overpower cd's and vinyl and when mastered right on heavy vinyl it's hifi...... , def. of hifi.


----------



## Ruben123

I like this discussion and I like it even more that I can listen to HiFi music through my Samsung i9000 phone with Voodoo kernel. It sounds like magic, and even though it doesnt, it does. I hear it!!
 Oh and btw, it's loaded with 16 bit flacs. Excellent combo. I hear birds singing that were NEAR the studio with it's windows closed.


----------



## StanD

ruben123 said:


> I like this discussion and I like it even more that I can listen to HiFi music through my Samsung i9000 phone with Voodoo kernel. It sounds like magic, and even though it doesnt, it does. I hear it!!
> Oh and btw, it's loaded with 16 bit flacs. Excellent combo. I hear birds singing that were NEAR the studio with it's windows closed.


 
 Can you make out what they were saying?


----------



## Ruben123

The bird is the word.
  
 lol
  
 If you like some fun, head over to the Pono thread. If you believe what they say, you are brought back to the recording studio when listening through that player. Awesome!! They hear it and while no DBT is done they hear it so it must be true.
  
 Period!!!


----------



## StanD

ruben123 said:


> The bird is the word.
> 
> lol
> 
> ...


 
 Once in awhile I go there for a good laugh. The closest they come to Science is Science Fiction.


----------



## cjl

ffbookman said:


> No one listens to me, that's cool. Who am I? Let me try to restate the crux of my argument.
> 
> You can't ABX test for music quality.


 
 So you can't hear music quality?
  
 All ABX does is it presents you with two files and asks you to discern the difference by listening. There's no limitation on how long you can listen for it to be a valid test, there's no limitation on what you should be listening for (be it ambience, soundstage, stereo image, or weird artifacts in a particular flute solo). The only requirement is that you listen to the music, and see if you can tell the difference.
  
 I find it very strange that people will wax poetically about how "musical" something sounds with a new component/format/etc, and how the change is night and day, and audible with any remotely decent equipment, but when asked a question as simple as "Pick the musical one out of these two samples, and do it consistently", suddenly the difference is so subtle that it can't be done? It doesn't stand up to scrutiny.


----------



## cjl

efeuvete said:


> 1) Any REAL musical sound has two audible parts: The main and first one is a (more or less) periodic function of any form (Let's call it A) and a second one: The natural instrument noise (Not ambient noise), a non periodic function, so without any wave form, just pressure changes following not known rule (Let's call it part B). And is important to say here that, nevertheless, this noise has "its own unknown" rules; for instance, it is different at different frecuencies of part A.


 
 I know this is old at this point, but I wanted to address this...
  
 If a part of the music cannot be broken down into periodic frequency components, that means that it is effectively DC. You can't hear this, nor is it reproducible by speakers. Therefore, it doesn't matter. A randomly varying signal can still be described based on its frequency content, and just because you can't see a simple, repeating pattern doesn't mean the signal doesn't contain any frequencies.


----------



## StanD

cjl said:


> I know this is old at this point, but I wanted to address this...
> 
> If a part of the music cannot be broken down into periodic frequency components, that means that it is effectively DC. You can't hear this, nor is it reproducible by speakers. Therefore, it doesn't matter. A randomly varying signal can still be described based on its frequency content, and just because you can't see a simple, repeating pattern doesn't mean the signal doesn't contain any frequencies.


 
 Just like White, Red and Pink noises have different frequency distributions. I doubt that our friend will acknowledge any of this.


----------



## RRod

stand said:


> Just like White, Red and Pink noises have different frequency distributions. I doubt that our friend will acknowledge any of this.


 
  
 I bet he could ABX them apart, though ^_^


----------



## Ruben123

Doesn't he just mean the sound the fingers make when touching the guitar is apart from the sound of the guitar itself? I.e. the music.


----------



## RRod

ruben123 said:


> Doesn't he just mean the sound the fingers make when touching the guitar is apart from the sound of the guitar itself? I.e. the music.


 
  
 And we're saying that if we can hear it, it's because it causes disturbances in the air that our ears can pick up, and those disturbances get picked up by microphones too. That means they make it to the ADC, and the parts we can hear get through the anti-aliasing filter and get sample values.


----------



## bigshot

Sound is frequencies.


----------



## Steve Eddy

bigshot said:


> Sound is frequencies.




So says Fourier. 

se


----------



## castleofargh

cjl said:


> efeuvete said:
> 
> 
> > 1) Any REAL musical sound has two audible parts: The main and first one is a (more or less) periodic function of any form (Let's call it A) and a second one: The natural instrument noise (Not ambient noise), a non periodic function, so without any wave form, just pressure changes following not known rule (Let's call it part B). And is important to say here that, nevertheless, this noise has "its own unknown" rules; for instance, it is different at different frecuencies of part A.
> ...


 

 yup he can get an idea of what we're talking about with something like this


  the more frequencies the less it looks like it can be made of waves, but it always is.


----------



## efeuvete

cjl said:


> So you can't hear music quality?
> 
> All ABX does is it presents you with two files and asks you to discern the difference by listening. There's no limitation on how long you can listen for it to be a valid test, there's no limitation on what you should be listening for (be it ambience, soundstage, stereo image, or weird artifacts in a particular flute solo). The only requirement is that you listen to the music, and see if you can tell the difference.
> 
> I find it very strange that people will wax poetically about how "musical" something sounds with a new component/format/etc, and how the change is night and day, and audible with any remotely decent equipment, but when asked a question as simple as "Pick the musical one out of these two samples, and do it consistently", suddenly the difference is so subtle that it can't be done? It doesn't stand up to scrutiny.


 
  
 Hi, I must say again that If I write about this topic here is because I think it is an interesting one but, please, the fact that my opinion is obviously different of the main forum opinion, does not mean that I have any intention of disturbing anybody with it in any way. Once this is said, I go:
  
 cjl: As I've said before, I'm not a math expert at all, so I may be wrong; But as I understand the Fourier Transform, what this math device can do is to give a "Frequency domain" interpretation of a sampled sound but, that does not mean that its interpretation is always an exact reproduction of any kind of sample.
  
 I say this because following what a Fourier Transform (FT) is, what I can see is a relevant mathematical device which represents PERIODIC functions of any form as sums of infinite senoidal functions. So, even being those sums infinite, as the sum terms decrease in weight, is obvious to me that the FT will do a good job analyzing musical sounds; Even to the point of giving us a minimun level of sampling frequency via the Nyquist Theorem. But...
  
 Then came the idea of applying this FT to any sequence of numbers which, as it seems, as a lot af math and physical applications as FT provides (almost always) a representation of that sequence of numbers as a sum of infinite weighted frecuencies (Frecuency domain). Well, no doubt this can be very useful (Depending of what the original sequence can mean) but, as far as I can understand, I'm not sure at all about "how precise is" the FT of a sampled thunderstorm noise or of a sampled slammed door noise (Natural NO PERIODIC noises).
 Surely, the aproximation would be significative, and useful too, but I see no way to know how big is the difference between the FT of the sample of a natural noise and the sample itself, nor the error I can have applying Nyquist theorem to that sample. And, being the topic here the very subtle difference between 16/44100 and 24/96000, I' not sure at all "my" musical instrument noise does not get lost in the FT game.
  
 Maybe here on the forum there's people who know maths enough (Or know sombebody who knows maths enough) to tell me if my opinion is wrong and why and, bellieve me, I would be as happy to fundament it as to discard it.
  
 PS/ Castleofargh: I'm not saying that noise is not perceived as a sum of waves (It well can be, I don't know), what I'm saying is that I don't know if FT, and therefore, Nyquist theorem, is applicable samplig natural noise.


----------



## efeuvete

cjl said:


> I know this is old at this point, but I wanted to address this...
> 
> If a part of the music cannot be broken down into periodic frequency components, that means that it is effectively DC. You can't hear this, nor is it reproducible by speakers. Therefore, it doesn't matter. A randomly varying signal can still be described based on its frequency content, and just because you can't see a simple, repeating pattern doesn't mean the signal doesn't contain any frequencies.


 

 Sorry, I've quoted wrong. This one is the post I should have quoted.


----------



## RRod

efeuvete said:


> Sorry, I've quoted wrong. This one is the post I should have quoted.


 
  
 The requirement for the Sampling Theorem is that the signal must be bandlimited as I mentioned previously. If a signal is bandlimited, then the samples can exactly reproduce the signal (the proof of the theorem gives a theoretical reconstruction via sinc functions). But we can't bandlimit perfectly since our signals are finite in length, but we have ways of reducing this issue.
  
 Whatever complex sound you think of first has to go through the microphone, which already has its own frequency limits. It then has to go through the ADC, which first hits it with an anti-aliasing filter so that the signal becomes approximately bandlimited. It then gets sampled. This is true at 24/96 just as much as at 16/44.1. The difference is the frequencies that get through, not magic non-frequency material.


----------



## efeuvete

I've posted twice instead of editing a detail. Sorry


----------



## efeuvete

rrod said:


> The requirement for the Sampling Theorem is that the signal must be bandlimited as I mentioned previously. If a signal is bandlimited, then the samples can exactly reproduce the signal (the proof of the theorem gives a theoretical reconstruction via sinc functions). But we can't bandlimit perfectly since our signals are finite in length, but we have ways of reducing this issue.
> 
> Whatever complex sound you think of first has to go through the microphone, which already has its own frequency limits. It then has to go through the ADC, which first hits it with an anti-aliasing filter so that the signal becomes approximately bandlimited. It then gets sampled. This is true at 24/96 just as much as at 16/44.1. The difference is the frequencies that get through, not magic non-frequency material.


 

 RRod, I've read your post, sorry I did not mention it but as I can not say up to what point "we have ways of reducing this issue" I didn't want to look like as critizicing somethiing which I do not know deeply enough.


----------



## bigshot

Within the range of human hearing (dynamics and frequency range), CD quality sound and high bitrate sound are bit for bit identical. High bitrate files don't contain any additional information that humans can hear. They contain information that is beyond the frequencies that ears can hear and below the threshold where things are too quiet to hear.


----------



## efeuvete

Maybe now you're going to wish killing me (If not already before) because I'm going to give you a "magical" example to synthetize how I see this "FT thing":
 Imagine I use a telephone guide as a source of some thousand numbers and I reduce them to the range -32768 +32768. Then, if I apply a Fourier Transform to those numbers, I will surely get a "Frequency Domain" image of the Guide.
  
 ¿Does that FT mean that telephone numbers follow frequency patterns?
  
 I'll log out for a while, just in case.


----------



## cjl

Interestingly, yes, you can extract frequency information from a set of numbers like that, and in some cases, it can even be meaningful (depending on how you arranged them in the first place).


----------



## StanD

efeuvete said:


> Maybe now you're going to wish killing me (If not already before) because I'm going to give you a "magical" example to synthetize how I see this "FT thing":
> Imagine I use a telephone guide as a source of some thousand numbers and I reduce them to the range -32768 +32768. Then, if I apply a Fourier Transform to those numbers, I will surely get a "Frequency Domain" image of the Guide.
> 
> ¿Does that FT mean that telephone numbers follow frequency patterns?
> ...


 
 While you're in hiding you can go square and study up on Walsh Analysis for some fun.


----------



## efeuvete

stand said:


> While you're in hiding you can go square and study up on Walsh Analysis for some fun.


 
  
 (Mr, or Mrs) StanD, I'm afraid I have the same reasons to study that "Walsh Analysis" than you for studying concert piano. Anyhow, if you try studying Karl Popper, maybe you'll have even some more. That guy taught me to, orderly and respectfully, doubt.


----------



## StanD

efeuvete said:


> (Mr, or Mrs) StanD, I'm afraid I have the same reasons to study that "Walsh Analysis" than you for studying concert piano. Anyhow, if you try studying Karl Popper, maybe you'll have even some more. That guy taught me to, orderly and respectfully, doubt.


 
 This forum is about Sound Science, not concert piano. I did study a lot of music theory, Diatonic and Chromatic Harmony, Counterpoint, Orchestration and so on. None of that helps with the bits and bytes of sound reproduction,


----------



## efeuvete

But be trained to doubt is good for sound science too (As "my sound science case", win or lost, may end showing).


----------



## RRod

efeuvete said:


> Maybe now you're going to wish killing me (If not already before) because I'm going to give you a "magical" example to synthetize how I see this "FT thing":
> Imagine I use a telephone guide as a source of some thousand numbers and I reduce them to the range -32768 +32768. Then, if I apply a Fourier Transform to those numbers, I will surely get a "Frequency Domain" image of the Guide.
> 
> ¿Does that FT mean that telephone numbers follow frequency patterns?
> ...


 
  
 Here, read this and then we can talk more about the FT, FS, DTFT, and DFT:
 http://www.dspguide.com/pdfbook.htm
  
 In a recording/playback chain without any digital filtering you don't even need to do any transforms. There it's all about the *analog* anti-aliasing and anti-imaging filters, and the electrical characteristics of the ADC, DAC, and transducers.


----------



## bigshot

It's interesting that someone would be well versed in the minutia of digital audio, but so oblivious to the basics.


----------



## sonitus mirus

efeuvete said:


> (Mr, or Mrs) StanD, I'm afraid I have the same reasons to study that "Walsh Analysis" than you for studying concert piano. Anyhow, if you try studying Karl Popper, maybe you'll have even some more. That guy taught me to, orderly and respectfully, doubt.


 
  
 Popper is great, and very well-known.  I've always considered science to be a type of orderly and respectful doubt. 
	

	
	
		
		

		
			




  
 Admittedly, I know very little about the subjects being discussed compared to others that routinely participate, but I do see where partial knowledge can be misleading. It seems that this forum is continually bombarded with sophomoric questioning of things that have already been quite clearly defined, and some confuse the knowledgeable replies to be that of closed-minded individuals.  A closed-minded person is the antithesis of a pro-science person.    
  
 Nothing enlightening is being introduced to refute what has already been rigidly proven.  I enjoy the discussions, nonetheless, and I am eagerly awaiting for some significant breakthrough to be introduced.


----------



## efeuvete

sonitus mirus said:


> Popper is great, and very well-known.  I've always considered science to be a type of orderly and respectful doubt.
> 
> 
> 
> ...


 
 Regarding your comment "Nothing enlightening is being introduced to refute what has already been rigidly proven" you must consider this significative fact: 24bits and 96000 Hz are worldwide used. Without that fact I would not have started to doubt, nor tried to check myself the differences with 16/44100, nor send any post here.


----------



## bigshot

It's used in the studio to aid in mixing and mastering. What does that have to do with playing music on a home stereo system?


----------



## efeuvete

rrod said:


> Here, read this and then we can talk more about the FT, FS, DTFT, and DFT:
> http://www.dspguide.com/pdfbook.htm
> 
> In a recording/playback chain without any digital filtering you don't even need to do any transforms. There it's all about the *analog* anti-aliasing and anti-imaging filters, and the electrical characteristics of the ADC, DAC, and transducers.


 

 Thanks RRod, sincerely, but I'm afraid that book is too much to my interest in DSP. Everything started listening to some old sixties music and now... I think is time for me to go back there. In 24/96 if I can...   but as happy if I must do it in mp3, 128Kbs.


----------



## StanD

bigshot said:


> It's used in the studio to aid in mixing and mastering. What does that have to do with playing music on a home stereo system?


 
 Perhaps he doesn't understand the numerical advantage of doing computations while processing audio in the studio which offers nothing during playback for a listener.


----------



## cjl

efeuvete said:


> Regarding your comment "Nothing enlightening is being introduced to refute what has already been rigidly proven" you must consider this significative fact: 24bits and 96000 Hz are worldwide used. Without that fact I would not have started to doubt, nor tried to check myself the differences with 16/44100, nor send any post here.


 
 24 bit has significant advantages when mixing dozens of tracks together, or layering effects, to minimize the compounding of errors. It's completely unnecessary for playback. Similarly, 96kHz or higher for recording makes sense so as to prevent aliasing (and so as to be able to use a gentle antialiasing filter). For the final result though, 16/44.1 is more than adequate, as a digital low pass filter can be applied before the downsampling that will not cause any audible error, and the errors up to 1LSB on 16 bit audio are completely inaudible.


----------



## MacacoDoSom

...part B does not follow any (known) rules?
  
 what? part B follows some unknown rules? some metaphysical ones? that's because part B is not sound at all... 'it's all in the mind'


----------



## MacacoDoSom

macacodosom said:


> ...part B does not follow any (known) rules


 
 ...part B does not follow any (known) rules?
  
 what? part B follows some unknown rules? some metaphysical ones? that's because part B is not sound at all... 'it's all in the mind'


----------



## MacacoDoSom

efeuvete said:


> I'm not a math expert at all but I've always thought that this condition in Nyquist theorem:
> "Strictly speaking, it only applies to a class of mathematical functions whose Fourier transforms are zero outside of a finite region of frequencies"
> means that Nyquist theorem only applies to periodic functions (Natural noise is not a periodic function, it has no waveform).
> Of course, I can be wrong.


 

 you are wrong... natural or unnatural noise, is sound or it isn't (visual noise, mental noise, smelling noise...) if is sound it has a waveform...


----------



## jcx

strictly band limited says nothing about randomness within the bandwidth limit
  
 and its not a problem that strictly "mathematical ideal" conditions are not met, no "infinite time" as required by "zero frequency content outside the band" - its simply not required in practice because of noise in all analog signals sound, microphone and electronic - we only have to do a "good enough" job in the digitization, digital audio processing combined with real human hearing limitations


----------



## manbear

Nyquist doesn't need a "periodic" function to work. (Quotes because instrument noise or other messing looking waves can still be made up of periodic waves, as others have tried to explain). But you could sample a straight line, a random squiggle, an irregular square wave, whatever--the approximation errors would be inaudible (if we incorrectly assume that these things could ever be recorded or played through speakers in the first place).


----------



## dprimary

ffbookman said:


> If you want to do the real test of 24bit masters verse 16bit downsamples I can offer up some materials and connections. I have about 10 albums in open FLAC @ 24bit, can do the low-end Apple conversion and the mid-range focusrite conversion to 16bit, and know someone with the converters to do the high-end versions.  I don't know that he will work for free though, I'd have to ask if he's interested in helping in the name of science.
> 
> I'd love to throw in DSD but it introduces different mastering sessions as a variable. I'd rather stick with a single source file at 24bit and test multiple down samples of the same source.


 

 Sorry that is not a test of 16 bit vs 24bit. That is a test of different software and hardware bit depth reduction, dither and who knows how many other variables. All the samples have to run through same process. You can download a demo of RX4 and dither them all at 24 bit, 20, 18, 16, 12, 8.
  
 I would be impressed if you can pick out the 12 bit version without turning up the volume above 80dB. If the recording was analog there is a good chance the tape noise is greater then the dither noise even if you do turn it louder.


----------



## castleofargh

dprimary said:


> ffbookman said:
> 
> 
> > If you want to do the real test of 24bit masters verse 16bit downsamples I can offer up some materials and connections. I have about 10 albums in open FLAC @ 24bit, can do the low-end Apple conversion and the mid-range focusrite conversion to 16bit, and know someone with the converters to do the high-end versions.  I don't know that he will work for free though, I'd have to ask if he's interested in helping in the name of science.
> ...


 

 don't worry he rejects any and all controlled testing methods that would show what you try to explain to him.


----------



## lamode

Yet another reason why we don't need digital audio systems with a frequency range up to 192kHz or more:


----------



## analogsurviver

lamode said:


> Yet another reason why we don't need digital audio systems with a frequency range up to 192kHz or more:


 
 Time for some dusting of (long) obsolete charts ? 
  
 http://www.cco.caltech.edu/~boyk/spectra/spectra.htm


----------



## lamode

analogsurviver said:


> Time for some dusting of (long) obsolete charts ?
> 
> http://www.cco.caltech.edu/~boyk/spectra/spectra.htm


 
  
 Thanks. Nice to see articles like this, though anything beyond 20kHz is only of academic interest, and even in this article the author admits that the ultrasonic frequency is mostly very low energy with the exception of the cymbals:
  


> "At least one member of each instrument family (strings, woodwinds, brass and percussion) produces energy to 40 kHz or above, and the spectra of some instruments reach this work's measurement limit of 102.4 kHz. Harmonics of muted trumpet extend to 80 kHz; violin and oboe, to above 40 kHz; and a cymbal crash was still strong at 100 kHz. In these particular examples, the proportion of energy above 20 kHz is, *for the muted trumpet, 2 percent; violin, 0.04 percent; oboe, 0.01 percent; *and cymbals, 40 percent."


 
  
 Should a chart show a range up to 40kHz even if the output is 30dB down from the fundamentals? Or 60dB? There has to be a line drawn.
  
 If humans could hear 50kHz, and most recording studios used microphones capable of recording this, I might sit up and take notice


----------



## castleofargh

I hate cymbals. I pretty much abandoned some of my favorite music genres just to get rid of it, or at least have it used in moderation. maybe that's it, I hate it because I'm able to perceive ultrasounds?


----------



## StanD

castleofargh said:


> I hate cymbals. I pretty much abandoned some of my favorite music genres just to get rid of it, or at least have it used in moderation. maybe that's it, I hate it because I'm able to perceive ultrasounds?


 
 All of this talk of ultrasonic hearing abilities is driving me bats. I feel sorry for sonar operators on submarines. 
	

	
	
		
		

		
		
	


	



 I have no idea where people come up with this stuff???


----------



## spook76

An very interesting interview with Steven Wilson arguably one of the best artist/recording engineers (Porcupine Tree and solo artist as well as the engineer behind Opeth to name one) today discussing high resolution music. http://www.digitaltrends.com/features/interview-steven-wilson-on-high-res-hand-cannot-erase/

If you have not heard 'The Raven That Refused To Sing' from Steven Wilson with Alan Parsons (after an almost 30 year away from the engineering, he was the man behind Pink Floyd's Dark Side Of The Moon) you need to listen. Whether you like the music the recording is impeccable.

Edit: I own the 24/96 of both of Steven Wilson's last two albums bought off his website as well as a 16/44 Redbook. In all candor, I cannot hear a difference between the high res and redbook with my rig.


----------



## kraken2109

spook76 said:


> An very interesting interview with Steven Wilson arguably one of the best artist/recording engineers (Porcupine Tree and solo artist as well as the engineer behind Opeth to name one) today discussing high resolution music. http://www.digitaltrends.com/features/interview-steven-wilson-on-high-res-hand-cannot-erase/
> 
> If you have not heard 'The Raven That Refused To Sing' from Steven Wilson with Alan Parsons (after an almost 30 year away from the engineering, he was the man behind Pink Floyd's Dark Side Of The Moon) you need to listen. Whether you like the music the recording is impeccable.
> 
> Edit: I own the 24/96 of both of Steven Wilson's last two albums bought off his website as well as a 16/44 Redbook. In all candor, I cannot hear a difference between the high res and redbook with my rig.


 
 While i disagree with him here, his engineering always sounds very good. I am a big fan of his 5.1 mixes too.


----------



## spook76

kraken2109 said:


> While i disagree with him here, his engineering always sounds very good. I am a big fan of his 5.1 mixes too.



I do know if I disagree with him as he guesses that high res improvement is maybe a 0.1% difference. I also think he he hits the nail on the head with his comment about the confort knowing it is 24/96 which to me is the expectation bias (placebo effect) driving high resolution music.


----------



## analogsurviver

lamode said:


> Thanks. Nice to see articles like this, though anything beyond 20kHz is only of academic interest, and even in this article the author admits that the ultrasonic frequency is mostly very low energy with the exception of the cymbals:
> 
> 
> Should a chart show a range up to 40kHz even if the output is 30dB down from the fundamentals? Or 60dB? There has to be a line drawn.
> ...


 
 There is more to this HF business that should be "inaudible" - it is NOT. It is my experience for more than 30 years by now - audibly, musically.
  
 It takes YEARS - if not decades - to develop/train this capability - preferably listening to live music (while recording it - helps a lot ). Great Musicians are, technically speaking, nothing but incredibly precise time modulated noise makers - and their time domain "performance" lies well above average person in general - or people on this thread, me included. 
  
http://m.phys.org/news/2013-02-human-fourier-uncertainty-principle.html
  
 What has all ABX boiled down for me?  Useful for gross large difference comparisons only - in an ABX, the difference between redbook and something serious can easily not be heard at all - or deemed too little to bother.
 But - that is a BIG BUT - in the long run, like listening to an entire work ( listening to my recording of rehearsal of Bach's Johannes Passion in 192/24 vs DSD128 from two days ago while typing this ) in full resolution and its "equivalent" in redbook on high quality equipment leaves absolutely no doubt which IS better. 
  
 It is in those 0.XYZ %,  below -60 dB, less than 14 microsecond rise times, etc. It is NOT a big difference, it can be glossed over easily - as it HAS BEEN since the "advent" of CD - yet it makes all the difference.
  
 But - and this is an even, in real world, bigger BUT than the one above - 50 kHz and beyond capable microphones and EVERYTHING ELSE IN STUDIO CAPABLE OF SUPPORTING THIS BANDWIDTH means, for all practical purposes - upgrade - or probably more cost effective REPLACEMENT - of all the gear.  And has to go hand in hand with MUCH better recording practice than "20 kHz is enough" that established itself after CD. And THAT is bound to be fiercely opposed - with untold loads of CD evangelists helping along the way.
  
 Hearing recordings of the same acoustic bass ( THE instument for testing audio gizmos - par excellence ) recorded with microphones from Earthworks that are essentially the "same" design - but mikes limited to 20, 30,40 and 50 kHz respectively - SOUND appreciably different ( within "house" sound - VERY neutral ) from each other - or other makes/models. Earthworks USED TO distribute these recordings on redbook CDs - which still trounce the mp3s (or whatever) they have satisfied themselves with on internet in the recent years.
  
 DISCLAIMER: I am in no way affiliated with Earthworks nor am I using any of their equipment (yet). I cite it because, IMO, they have done the most regarding opening the mind regarding life beyond 20 khz - in microphones, at least.
  
 There is at least one microphone capable of 100 kHz bandwidth - Sanken CO100K. To record this "flat", PCM of greater than 192 kHz sampling is required - and for DSD, it means at least DSD256. Such recorder(s) exist already - and is likely to expand in availability in near future.
  
 Even if humans can not hear PURE SINE WAVE tones above 20 kHz - according to the above link, CAN - and - DO - discern extremely small time shifts (or whatever one might want to call it ). And is a reason why an elderly person, who is working with music/sound for all practical purposes whole life, despite the fact that his/hers hearing of pure tones is limited to 10 or even below kHz, CAN reliably describe the performance of the supertweeter better than a teenager with "perfect" hearing to 20 kHz - but without the training/experience required.
  
 Bottom line - it is NOT that easy to break the sound by science in bits and pieces and then on the ground of a SINGLE parameter  claim anything - for certain. Just because (your) present equipment does not support it does not necessary mean that under proper conditions it will not be audible. There is a cost vs performance limitation, of course - but what once was unobtainable for most people, can be today had in midrange equipment. Today's TOTL gear with prohibitive prices will eventually trickle down to something still good enough and affordable to the dedicated enthusiast.


----------



## analogsurviver

stand said:


> All of this talk of ultrasonic hearing abilities is driving me bats. I feel sorry for sonar operators on submarines.
> 
> 
> 
> ...


 
 Submarines are THE place for the best electroacoustics gear - anywhere, anytime. PERIOD. Because their very survival depends on it - nothing less, nothing more.
  
 In one nice review of recent DECCA boxes of CDs ( like 50 CD box ), it was clearly written that this breaktrough in what was later coined DECCA Full Frequency Sound *** was a direct consequence from the war effort by British engineers in order to produce equipment with which German subs could be detected before it was too late. Today, it is how sensitive/quiet the listening gear can be - once located, a sub today is barely beyond being a sitting duck. And it is acoustic detection that still reigns supreme underwater - why on earth any reasonably powerful navy on earth is supporting biologists researching "whales" 
	

	
	
		
		

		
		
	


	




 ?


----------



## analogsurviver

castleofargh said:


> I hate cymbals. I pretty much abandoned some of my favorite music genres just to get rid of it, or at least have it used in moderation. maybe that's it, I hate it because I'm able to perceive ultrasounds?


 
 Watch and listen at you own peril...:


----------



## StanD

analogsurviver said:


> Submarines are THE place for the best electroacoustics gear - anywhere, anytime. PERIOD. Because their very survival depends on it - nothing less, nothing more.
> 
> In one nice review of recent DECCA boxes of CDs ( like 50 CD box ), it was clearly written that this breaktrough in what was later coined DECCA Full Frequency Sound *** was a direct consequence from the war effort by British engineers in order to produce equipment with which German subs could be detected before it was too late. Today, it is how sensitive/quiet the listening gear can be - once located, a sub today is barely beyond being a sitting duck. And it is acoustic detection that still reigns supreme underwater - why on earth any reasonably powerful navy on earth is supporting biologists researching "whales"
> 
> ...


 
 Electronic detection gear has nothing to do with what humans can hear. That or using whales or bats or whatever, makes for a ridiculous analogy in the context of this thread. I think much of your claims of super hearing capabilities are also absurd, even with 30 years of intensive training. As I've said before, Harry Callahan knew the truth when he said (in character), "A man's got to know his limitations."
 Your other bits about above 20 kHz are also meaningless as we do not have the ability to hear this. Timeshifts with sine waves above 20 kHz, where do you come up with this stuff? Does that mean if I employ microwaves I can travel back to the age of dinosaurs?
 If you believe this stuff and can enjoy it, that's fine with me, only I will NOT buy into it, not a single penny.


----------



## RRod

analogsurviver said:


> Time for some dusting of (long) obsolete charts ?
> 
> http://www.cco.caltech.edu/~boyk/spectra/spectra.htm


 
  
 So the article boils down to this: Oohashi. I'd love to know why this EEG stuff hasn't been followed up by a rigorous blind test where people are asked to identify when HF content is turned on or off, on equipment that handles the HF without audible IMD. Seriously, that's all you'd have to do to convince me that we should stick to higher bit rates (though 24 bits is pretty much dead in the water in terms of end-user delivery for me). If there's a well-done study with adequate sample size, I'd love to see it.


----------



## jcx

even Earthworks only include 50 kHz mic in their drum mic kit
  
 fast mics have to have small diameter diaphragms and because of the small size they are too noisy for anything but close mic application - capturing sound that never reaches the audience
  
 large diaphragm legendary vocal mics may roll off as low as ~14 kHz
  
 useful studio practice and reproducing a "realistic" musical recording in the home are different


----------



## bigshot

Theory is great, but numbers can be deceiving. When the average person looks at frequency numbers, they don't take into account the fact that every octave doubles the number. Humans can hear something like ten octaves. But 20,000Hz to 40,000Hz is the exact same size range as 80Hz to 160Hz. When you talk about ultrasonic frequencies, 50kHz is just a hair over one octave above the top end of human hearing. Barely enough to spit at... Plus there isn't much to hear up there except for cymbal crashes even if you could hear it... Plus even if you could hear it, it would probably be masked by lower level harmonics in the cymbal crashes. Plus it's the an octave beyond the least important octave when it comes to sound quality.
  
 It really doesn't make sense why audiophiles spend so much time worrying about things they can't hear and not about the stuff they can.


----------



## analogsurviver

jcx said:


> even Earthworks only include 50 kHz mic in their drum mic kit
> 
> fast mics have to have small diameter diaphragms and because of the small size they are too noisy for anything but close mic application - capturing sound that never reaches the audience
> 
> ...


 
 It is true that fast mics have small diameter and are more noisy - but that they are too noisy for anything but close mic application is NOT entirely true .  
  
 I consider large diaphragm mics detrimental for the true fidelity. OK if one wants a particular flavour of the sound - beyond that - no.
  
 It does take MUCH MORE than common knowledge - but listen how "noisy" is this recording done with matched pair 1/4 inch omnis ( there is air conditioning coming off the right side - NOT mic noise ...) - far away enough that even camera could not catch them, way beyond what audience in the parter could get to hear :
  

  
 Original recording is DSD128 - and it sounds incomparably better than above YT.


----------



## Don Hills

analogsurviver said:


> Watch and listen at you own peril...:


 
  
 ... to absolutely nothing above 18 KHz.


----------



## analogsurviver

don hills said:


> ... to absolutely nothing above 18 KHz.


 
 It is YouTube -
  
 not something serious ...


----------



## lamode

analogsurviver said:


> What has all ABX boiled down for me?  Useful for gross large difference comparisons only - in an ABX, the difference between redbook and something serious can easily not be heard at all - or deemed too little to bother.
> But - that is a BIG BUT - in the long run, like listening to an entire work ( listening to my recording of rehearsal of Bach's Johannes Passion in 192/24 vs DSD128 from two days ago while typing this ) in full resolution and its "equivalent" in redbook on high quality equipment leaves absolutely no doubt which IS better.


 
  
 There is no time limit on an ABX test, so this argument doesn't hold.


----------



## analogsurviver

lamode said:


> There is no time limit on an ABX test, so this argument doesn't hold.


 
 Meant was that it does not work for me if I have to switch it "all the time". 
  
 Added "something/whatever" missing from the redbook DOES add to the realism - but it is more feeling over long(er) period of time than any reasonable ABX allows for between switching.
  
 An analogy: if and when sleeping in a very cold room, the difference between a great and merely good blanket is far lesser than between either of them and having to go to pee in that cold - not a pleasent thing to do. Which creates enough stress for you not to care much about the blanket quality - as long as there is one to keep you warm. ABX to me is a similar distraction causing enough stress - but I am using it for larger differences.
  
 You do not get to appreciate the difference between the blankets - unless given enough uninterupted time with both of them - say at least one hour with each.
  
 Same with music/sound reproduction. An ABX may well never persuade one to get hirez over redbook - but having it over weekend will. If not right away, the NEXT day one returns to redbook - "something" will be missing - and once experienced, it will be hard to deny and resist the advantage it is offering.


----------



## dprimary

The only limitation to ABX it you can't know what you are listening too. Which seems to be too stressful for audiophiles and makes all the  night and day differences vanish.


----------



## bigshot

If you can't hear it in an ABX, it probably doesn't matter anyway


----------



## sonitus mirus

Making an ABX impractical or unlikely for any objective listener to attempt is not going to change anything.  You can still configure an ABX for 10 albums or 2 months between test samples.  Red Book is "serious", and I sincerely doubt anyone can absolutely hear a difference between it and an equivalent high resolution format.
  
 Has audiophile marketing finally stooped so low that it has to claim that an ABX is not reliable for removing expectation bias?


----------



## RRod

sonitus mirus said:


> Has audiophile marketing finally stooped so low that it has to claim that an ABX is not reliable for removing expectation bias?


 
  
 From what I've seen on the forums lately, the answer is yes.


----------



## dazzerfong

If something is patently better, it should be pretty damn obvious upfront. The fact that you even have to try so damn hard to listen for a difference indicates that it's a redundant 'improvement', let alone an 'improvement' in the first place.


----------



## sonitus mirus

dazzerfong said:


> If something is patently better, it should be pretty damn obvious upfront. The fact that you even have to try so damn hard to listen for a difference indicates that it's a redundant 'improvement', let alone an 'improvement' in the first place.


 
  
 Apparently the differences are night and day, but you need a weekend to come to this conclusion.  I suppose the difference would be night and day, night and day.


----------



## dazzerfong

sonitus mirus said:


> Apparently the differences are night and day, but you need a weekend to come to this conclusion.  I suppose the difference would be night and day, night and day.


 
 Great, we got ourselves a funny guy here. And they say science is boring.


----------



## Don Hills

analogsurviver said:


> It is YouTube -
> 
> not something serious ...


 

 It still renders your point about cymbal ultrasonics null.


----------



## analogsurviver

don hills said:


> It still renders your point about cymbal ultrasonics null.


 
 You can use any good analog or genuine native DSD or PCM ( 88.2 kHz and up )  recording using >20 kHz microphones. It has been possible for at least 30 years .


----------



## analogsurviver

dprimary said:


> The only limitation to ABX it you can't know what you are listening too. Which seems to be too stressful for audiophiles and makes all the  night and day differences vanish.


 
 It is NOT the only limitation of ABX. At least not ABX possible with Foobar2000. 
  
 I have written MANY times it can not ABX any native DSD vs PCM - because it has to convert DSD to PCM before it can perform an ABX. In addition, it makes a noise /click whenever switching from DSD to PCM - making listener at least aware that formats are being changed. This makes most if not all of the advantages of DSD vs PCM moot - rendering such ABX totally and utterly useless. For this DSD vs PCM ABX, an entirely analog ABX box would be best; but having analog potenentiometers and switches capable of reliably switching between DSD and PCM within 0.1 dB or less difference in level is going to be expensive. 
  
 Depending on format and on software to convert one format to another, there can be approx 0.3 dB difference in level among any of the combinations - requiring to be compensated for at the playback. An analog box capable of doing it would run you at least $ 500 - or more.
  
 Accept it - computers and digital in general used for sound is great - up to a point. Beyond that, it becomes a liability. There is one hell of a lot sonic difference between merely playing back a native DSD file - between a stock unit and an unit modified by me. BOTH of them are most likely better than most of equipment used by the members on this thread. Playing these files via my PC and DSD DACs at hand DOES sound different - in short, equipment used  can have enough of an influence to render PCM vs DSD issue moot.
  
 According to the "all amplifiers sound the same" brigade, it does not matter - in the slightest.
  
 But it DOES. Pity it can not be transmitted to other end via computers - it takes real hardware to do it.  Computer can only transmit digital file - which it does perfectly. Beyond that, all analog circuits ARE critical - and for cost considerations, there is almost zero commercially available equipment done right - regardless the price. Even 5 figure boxes contain parts that could and should have been better - particularly at these high prices.


----------



## bigshot

analogsurviver said:


> It is NOT the only limitation of ABX.


 
  
 ABX's chief problem is that it puts the lie to paragraphs and paragraphs of theory without practical experience behind it. That can be very embarrassing.


----------



## analogsurviver

bigshot said:


> ABX's chief problem is that it puts the lie to paragraphs and paragraphs of theory without practical experience behind it. That can be very embarrassing.


 
 What is truly embarassing ?
  
 Once musicians, who listen ( mainly, mostly, with very few exceptions far in between ) without technical knowledge, WILL start to "thank" for "the perfect sound forever" - not only privately, but publicly so.
  
 No amount of real world or virtual ink - or how I call it - chicken squawking - will do the trick. Getting the musician to listen to his/hers recording does - every time.
 But it has to be made well and played back on quality enough gear. And it IS tedious to the max - you have to start with each individual person from the start. It is only a smidge less difficult to convince and work with a musician #(n+1) after being recommended to him/her by the musician #(n).
  
 Guess what ? About a week ago, the artistic director of a vocal group, a female, most definitely without any technical background, asked me in a caffe if I know "those tiny black discs you put on the CD to make it sound better" ? Of course, they are called CD mats, usually made from carbon fibre - I use them for approx 7 years now - and told her to come by to collect hers the next day, as I always happen to have a few samples in stock. 
  
 Musicians LISTEN - and they WILL go to reasonable lengths when they discover that something is representing their efforts better. Whatever it takes. And although I have been recording with said person numerous times by now, direct to CD-R with the obligatory CD mat included, I never specifically pointed out the use of the CD mat. But it DID come around - because she DOES care about how she listens to music - even if it is still from the CD.
  
 For now...


----------



## dazzerfong

analogsurviver said:


> What is truly embarassing ?
> 
> Once musicians, who listen ( mainly, mostly, with very few exceptions far in between ) without technical knowledge, WILL start to "thank" for "the perfect sound forever" - not only privately, but publicly so.
> 
> ...


 

 Why not just fix up their recordings such that CD is the limiting factor, not the skill and abilities of the music producers? You are now really scraping at the bottom of the barrel if we're talking about CD mats rather than poor mastering.
  
 But that's another story for another time, isn't it? For now, 16 bit vs 24 bit.


----------



## analogsurviver

dazzerfong said:


> Why not just fix up their recordings such that CD is the limiting factor, not the skill and abilities of the music producers? You are now really scraping at the bottom of the barrel if we're talking about CD mats rather than poor mastering.
> 
> But that's another story for another time, isn't it? For now, 16 bit vs 24 bit.


 
 First of all, bad recording can not be "mastered" into a decent one. Better yes, good enough - no. Regardless in which format it appears. Skill and abillities of the music producer will always be the limiting factor - there is no way in the world to make a multimiked recording sounding realistic.
  
 24 vs 16 bit is not that objectionable - as I have written before. It is really hard to get music and venue that will challenge the 96 dB dynamic range - at least once mastered for final delivery. Minimum delivery format should be 88.2/16 . Recording is another matter - 24 bit is required for headroom and later post production. Or better still, going DSD .
  
 What I do find objectionable is limited frequency response. 
  
 And the inability of the laser optics to read the CDs well enough. Which has to be helped by a mat... - remember, you have to rip CDs somehow to get them on the hard disk - it applies here too, regardless if you no longer listen to a CD player.
  
 Which is not to say that I do not find listening to 192/32 floating amazing - on equipment that can do it justice. Amazing enough to consider ABXing it against redbook a total and utter waste of time - but, this time at least, Foobar2000 does allow for it properly.


----------



## RRod

analogsurviver said:


> Which is not to say that I do not find listening to 192/32 floating amazing - on equipment that can do it justice. Amazing enough to consider ABXing it against redbook a total and utter waste of time - but, this time at least, Foobar2000 does allow for it properly.


 
  
 It's easy enough to convert between Redbook and 32f/192, and I'm sure there's got to be an ABX program out there that handles floating-point properly.


----------



## analogsurviver

rrod said:


> It's easy enough to convert between Redbook and 32f/192, and I'm sure there's got to be an ABX program out there that handles floating-point properly.


 
 Foobar2000 ABX works well enough - as well as Korg Audiogate for conversions - but what is the point of the ABX of 192/32float vs redbook, IF the original is hirez or even DSD ?


----------



## RRod

analogsurviver said:


> Foobar2000 ABX works well enough - as well as Korg Audiogate for conversions - but what is the point of the ABX of 192/32float vs redbook, IF the original is hirez or even DSD ?


 
  
 What do you mean? The whole point of ABX is to see if you can really hear the different between hi-res and Redbook when blind… If 32f/192 sounds so amazing, then it should be obvious to your ears when we suddenly shift to a 16/44.1 version.


----------



## StanD

analogsurviver said:


> :
> *What I do find objectionable is limited frequency response. *
> :


 
 What about people making superhuman claims that are not possible?


----------



## lamode

analogsurviver said:


> Guess what ? About a week ago, the artistic director of a vocal group, a female, most definitely without any technical background, asked me in a caffe if I know "those tiny black discs you put on the CD to make it sound better" ? Of course, they are called CD mats, usually made from carbon fibre - I use them for approx 7 years now - and told her to come by to collect hers the next day, as I always happen to have a few samples in stock.
> 
> Musicians LISTEN - and they WILL go to reasonable lengths when they discover that something is representing their efforts better. Whatever it takes. And although I have been recording with said person numerous times by now, direct to CD-R with the obligatory CD mat included, I never specifically pointed out the use of the CD mat. But it DID come around - because she DOES care about how she listens to music - even if it is still from the CD.
> 
> For now...


 
  
 Musicians are no different to anyone else when it comes to expectation bias.
 At the end of the day, whether you use magic power cable or carbon fibre CD mats, the sound is no different, and you certainly don't suddenly develop the ability to hear ultrasonics!


----------



## StanD

lamode said:


> Musicians are no different to anyone else when it comes to expectation bias.
> At the end of the day, whether you use magic power cable or carbon fibre CD mats, the sound is no different, and you certainly don't suddenly develop the ability to hear ultrasonics!


 
 It may have taken 30 years but he can hear Ultrasonics. So can the Easter Bunny.


----------



## lamode

analogsurviver said:


> And the inability of the laser optics to read the CDs well enough. Which has to be helped by a mat... - remember, you have to rip CDs somehow to get them on the hard disk - it applies here too, regardless if you no longer listen to a CD player.


 
  
 Oh dear lord... This HAS to be trolling. CD drives do a great job of retrieving CD data. If the drives were constantly misreading bits, then no CD-ROM based software would ever work. Please think before you post, it's getting tiresome. Anyway, CDs are pretty old technology now and becoming irrelevant - I switched to FLAC over 10 years ago.


----------



## analogsurviver

lamode said:


> Musicians are no different to anyone else when it comes to expectation bias.
> At the end of the day, whether you use magic power cable or carbon fibre CD mats, the sound is no different, and you certainly don't suddenly develop the ability to hear ultrasonics!


 
 Sorry, you obviously did not understand it regarding the CD mats - the musician in question DEMANDED the CD mat - AFTER she heard it at some other place. If you think this is an expectation bias, you are dead wrong - she knows her singers as she works with them practically on daily basis - and CD without CD mat simply does not deliver. Plain and simple.
  
 The only exceptions are CD players like turntables, where you put the CD disc upside down and read it from the top - and where the whole surface of the CD is being supported. There were Pioneer and Sony transports offering this in the past. Also, more recently, Yamaha produced some transport that also works well, despite precluding the use of any mat(s).
  
 I have heard cases where CD mat did not change a thing - because the entire system was too poor to begin with. On quality equipment, it IS night and day.
  
 No cable, let alone power cable, can have that much effect as a decent CD mat.  Before you equate these again, a blind AB with the use of help from a friend ( no, ABX comparators have not learned to place and remove CD mat on revolving disc for instantaneous ABX - yet ) should tell you a LOT about your system. If your CD player is not one of the exceptions mentioned above, If you can not hear anything, the equipment is too poor - and needs replacing. 
  
 Cruel - but true.


----------



## lamode

analogsurviver said:


> Sorry, you obviously did not understand it regarding the CD mats - the musician in question DEMANDED the CD mat - AFTER she heard it at some other place. If you think this is an expectation bias, you are dead wrong - she knows her singers as she works with them practically on daily basis - and CD without CD mat simply does not deliver. Plain and simple.
> 
> The only exceptions are CD players like turntables, where you put the CD disc upside down and read it from the top - and where the whole surface of the CD is being supported. There were Pioneer and Sony transports offering this in the past. Also, more recently, Yamaha produced some transport that also works well, despite precluding the use of any mat(s).
> 
> ...


 
  
 Pure nonsense.


----------



## analogsurviver

lamode said:


> Pure nonsense.


 
 The sooner you will try it, the sooner you will know the truth.
  
 Then is up to you; in case it does not "work", to invest in improvement of the gear - or soldier on with existing gear. Except in this case you lose on the enjoyment of your CDs - as well as credibility about your capability to hear "anything". It is OK to say "I do not wish/can not afford such equipment, but realize that CD mat works on better equipment" - "It does not provide any difference" is NOT OK.
  
 Provided your CD player is not among those exceptions mentioned.


----------



## analogsurviver

lamode said:


> Oh dear lord... This HAS to be trolling. CD drives do a great job of retrieving CD data. If the drives were constantly misreading bits, then no CD-ROM based software would ever work. Please think before you post, it's getting tiresome. Anyway, CDs are pretty old technology now and becoming irrelevant - I switched to FLAC over 10 years ago.


 
 TRY it - before writing such a statement again.
  
 The mechanism at work is a simple one - CD disc vibration. Of course, each CD player has error correction algorithm - which puts the about the last sequence of signal in case the signal is not being read right. What that algoritm is putting in in case of signal lost, is an _*approximation *_- not the correctly read data. Less triggering of this error correction = better sound. As simple as that.
  
 CDs are, like it or not, at the present still the most widespread way for most people to listen to recorded music. Therefore are important - and worth playing back as well as possible.


----------



## Head Injury

analogsurviver said:


> Before you equate these again, a blind AB with the use of help from a friend ( no, ABX comparators have not learned to place and remove CD mat on revolving disc for instantaneous ABX - yet ) should tell you a LOT about your system.


 
 Rip a CD with and without a CD mat, then ABX the resulting FLAC files with the Comparator plugin. Or hell, just compare the files in Audacity, or see if the checksums match. Easy. But, I'm sure there's some mystical reason for why this won't work as well as slowly switching CDs with a friend.


----------



## analogsurviver

head injury said:


> Rip a CD with and without a CD mat, then ABX the resulting FLAC files with the Comparator plugin. Or hell, just compare the files in Audacity, or see if the checksums match. Easy. But, I'm sure there's some mystical reason for why this won't work as well as slowly switching CDs with a friend.


 
 Hey, guys, do you LISTEN to your music using comparators, Audacity, etc ?
  
 I will do it once, just to satisfy your curiosity. Now, what IS required is a software that can rip CDs at maximum reading speed of 10x or less and does NOT switch, not even for a moment, to any higher speed. I usually use Nero 8 - because it is compatible with Yamaha CRW-F1E USB burner AND - important - supports Audio Master mode of burning the CDs - all under Windows 7. Later versions of Nero are not necessary back compatible, do not necessary support Yamaha burner on which I insist nomatterwhat - and original Nero 5 that came with Yamaha is no longer compatible with Win 7. I can still use it on a Win XP machine.
  
 Any suggestions for the ripping software that satisfies all of the above, besides Nero 5 ( XP ) and Nero 8 ( Win 7 ) ?


----------



## Krutsch

lamode said:


> Oh dear lord... This HAS to be trolling. CD drives do a great job of retrieving CD data. If the drives were constantly misreading bits, then no CD-ROM based software would ever work. Please think before you post, it's getting tiresome. Anyway, CDs are pretty old technology now and becoming irrelevant - I switched to FLAC over 10 years ago.


 

 CD-ROM error correction is different from Redbook - Redbook (i.e. CD audio) will actually attempt to handle read errors by interpolating data and presenting an approximation of the expected bits; CD-ROM (i.e. what computers use for data reads/writes also provides error detection, via Reed-Solomon, but any failure is simply missing bits - i.e. catastrophic - and the reading software is so notified via the I/O stack).
  
 But, you are correct, modern optical readers are incredibly robust and nothing like what I had to write software to interact with in the early days (when errors were common-place). The need for robust read interpolation was a requirement in the early days of Redbook, and the audiophile myth continues today that there are special readers required for audiophile sound.
  
 Today, you can use software to match your rip against open source signature databases and be 100% assured that your FLAC file is a perfect match to the original. Good times we live in...


----------



## RazorJack

CD mat, wow I had to Google that one.
  
 I keep learning tons of hilarious things in this thread though


----------



## bfreedma

To get the best out of a cd mat, do you have to color its outside edge with a green marker?.....


----------



## analogsurviver

bfreedma said:


> To get the best out of a cd mat, do you have to color its outside edge with a green marker?.....


 
 No.


----------



## analogsurviver

razorjack said:


> CD mat, wow I had to Google that one.
> 
> I keep learning tons of hilarious things in this thread though


 
 I bought it out of curiosity at first - and it did not last a day in my system.
  
 Because it was snatched from me by a friend - he offered to buy it at a price I would be foolish not to accept - but he"cursed" me for not demoing the mat to him two weeks before - because in this case, he would not have bought a new CD player... - because old player + CD mat was equal of the new CD player that can not accept CD mat(s). Without CD mat, old player lost to the new one - no contest.
  
 It can make that much of a difference. Needles to say, I repurchased it ASAP. It is the very same : 
  
 http://www.ebay.de/itm/CD-Matte-BLACK-DIAMOND-High-Tech-Kohlefaser/361259778725?_trksid=p2047675.c100005.m1851&_trkparms=aid%3D222007%26algo%3DSIC.MBE%26ao%3D1%26asc%3D20140117130753%26meid%3Db8b9246b80924cec833f734e4871ea7f%26pid%3D100005%26rk%3D2%26rkt%3D6%26sd%3D121600720012&rt=nc
  
 For those listening to silver discs (it also works wonders for DVD ... ), it is the least expensive way to significantly improve the performance. No affiliation with the seller, only satisfied customer.


----------



## bfreedma

analogsurviver said:


> bfreedma said:
> 
> 
> > To get the best out of a cd mat, do you have to color its outside edge with a green marker?.....
> ...




Lol. It wasn't a serious question but I suppose I should ask if you've actually tried it.


----------



## lamode

analogsurviver said:


> The sooner you will try it, the sooner you will know the truth.


 
  
 I don't have to hang a sock over the kitchen sink to know that that won't reduce the error rate on my CD player. Enough with the trolling.


----------



## lamode

analogsurviver said:


> TRY it - before writing such a statement again.
> 
> The mechanism at work is a simple one - CD disc vibration. Of course, each CD player has error correction algorithm - which puts the about the last sequence of signal in case the signal is not being read right. What that algoritm is putting in in case of signal lost, is an _*approximation *_- not the correctly read data. Less triggering of this error correction = better sound. As simple as that.
> 
> CDs are, like it or not, at the present still the most widespread way for most people to listen to recorded music. Therefore are important - and worth playing back as well as possible.


 
 You are assuming that there are signifcant errors to begin with, and if there are, you can use a second or third pass to confirm the data rather than the primitive early error handling you described. But anyway, CD playback has shifted to CD-ROM drives and ripping. I sold my last CD player in the late 90s! (Sony CDP-X777ES).


----------



## lamode

krutsch said:


> CD-ROM error correction is different from Redbook...


 
  
 Yes, I'm well aware, but I was trying to keep things simple for our resident troll


----------



## analogsurviver

lamode said:


> I don't have to hang a sock over the kitchen sink to know that that won't reduce the error rate on my CD player. Enough with the trolling.


 
 No trolling . You can find similar reviews of basically the same thing : 0.3 mm thick precisely cut carbon fiibre disc : 
  
 http://www.stereomojo.com/MILLENNIUM%20CD%20MAT%20REVIEW/MillenniumCDMatReview.htm


----------



## analogsurviver

bfreedma said:


> Lol. It wasn't a serious question but I suppose I should ask if you've actually tried it.


 
 No.


----------



## analogsurviver

lamode said:


> You are assuming that there are signifcant errors to begin with, and if there are, you can use a second or third pass to confirm the data rather than the primitive early error handling you described. But anyway, CD playback has shifted to CD-ROM drives and ripping. I sold my last CD player in the late 90s! (Sony CDP-X777ES).


 
 CD-ROM drives benefit much in the same way - mechanically, they are pretty much the same, holding the disc only by the clamp around the disc hole.


----------



## bigshot

I have a turntable in my microwave oven. When it would turn, it would go over little bumps and rattle a little. I took the turntable apart and cleaned the little wheels under it. Now my CDs play with greater transparency and sparkling highs. It seems to have an even bigger effect on SACDs than it does on CDs, and DVDs seem to be about the same. Next up, I plan to rotate the tires on my car to see if it improves the soundstage of my car stereo.


----------



## analogsurviver

bigshot said:


> I have a turntable in my microwave oven. When it would turn, it would go over little bumps and rattle a little. I took the turntable apart and cleaned the little wheels under it. Now my CDs play with greater transparency and sparkling highs. It seems to have an even bigger effect on SACDs than it does on CDs, and DVDs seem to be about the same. Next up, I plan to rotate the tires on my car to see if it improves the soundstage of my car stereo.


 
 Now this IS trolling 
	

	
	
		
		

		
		
	


	




.


----------



## Opportunist

I was recommended by an enthusiastic friend to buy a CD mat. Initially, I was impressed by what I perceived to be better definition and less glare. But after a while, I started to sense that something was missing - the music failed to engage the way it used to and the foot-tapping rhythm was much reduced. I stopped using the mat and am happy again - the lustre of strings, the full concert hall ambience and other micro-details are back. - I think that this could be due to the fact that the mat by introducing a distance between the platter and the magnetic clamp reduces the grip on the disc and makes it more prone to vibrate.


----------



## bigshot

I have the lights on a dimmer switch in my listening room. I find that music is darker and richer when I turn the lights down, and it has a sharp glare when they are turned up full blast. I'm going to try switching to those new LED lights and see if it improves the sound.


----------



## Bytor123

Shouldn't all this be testable - with the mat vs without the mat? Why would you not want to prove the difference by being able to say which is which consistently? If there was such a difference we'd be all over it.


----------



## dprimary

analogsurviver said:


> It is NOT the only limitation of ABX. At least not ABX possible with Foobar2000.
> 
> I have written MANY times it can not ABX any native DSD vs PCM - because it has to convert DSD to PCM before it can perform an ABX. In addition, it makes a noise /click whenever switching from DSD to PCM - making listener at least aware that formats are being changed. This makes most if not all of the advantages of DSD vs PCM moot - rendering such ABX totally and utterly useless. For this DSD vs PCM ABX, an entirely analog ABX box would be best; but having analog potenentiometers and switches capable of reliably switching between DSD and PCM within 0.1 dB or less difference in level is going to be expensive.
> 
> ...


 
 ABX is a methodology not a specific piece of software or hardware. To ABX DSD and PCM would require more work and hardware. You could not get it down to a single variable, which I consider the gold standard. But it is not that hard either. In the past I just used a mixing console since level adjustment and routing is one of the functions of a mixing desk and I have access to plenty of them. Have one person patch it together and match levels, a second that does the switching and tracks the switching and then the listener(s) is not all that hard. Even in quick testing I calibrate, if for nothing else a fast sanity check, too often I catch something that was not set right.
  
 Even when there is a difference, which one is correct becomes the next question. If it is barely noticeable does it even matter? Unless you were involved in the production how would you know which is closest? 
  
 I'm surprised at that it is believed all amps sound the same? I have compared amplifier sound quality more times then I can remember often somewhat blind, but not in what I would call rigorous ABX test, though we often test amps till they shutdown and test them more to see how well they recover. Sound quality doesn't matter if it stopped working. Many of them have a noticeable sound quality difference, even on different lines from the same manufacturer. Low frequency quality is the most noticeable. I am planning to do more rigorous ABX testing in the future, but I need it to be portable and we can setup and complete the testing in 30-45 minutes. That is about all the time I can spend on it and not interfere with other testing and configuration.


----------



## bigshot

bytor123 said:


> Shouldn't all this be testable - with the mat vs without the mat?


 
  
 Of course it is. But the people who are capable of conducting proper tests aren't interested in disproving every hoodoo that comes down the pike. It would be like fighting the tide... especially the way the flood tides have been flowing around here lately. It might be more useful to test whether the cycles of the moon make people post silly things.


----------



## analogsurviver

bigshot said:


> Of course it is. But the people who are capable of conducting proper tests aren't interested in disproving every hoodoo that comes down the pike. It would be like fighting the tide... especially the way the flood tides have been flowing around here lately. It might be more useful to test whether the cycles of the moon make people post silly things.


 
 That is a bit arrogant - even for the likes of bigshot .
  
 Seriously, continuing in this tone will lead you nowhere. About the same as American pilots volunteering for what later became known as  Flying Tigers were told all Japanese airplanes are biplanes and all Japanese wore glasses.
  
 Then they learned - the hard way - what Zero is. And that Japanese pilots can see just fine.
  
 Without such arrogant attitude, quite a few lives could have been spared.
  
 I have witnessed first hand one of the "tests" regarding errors etc on the CDs. At a pro mastering engineer, using TOTL Plextor CD burner/ROM/whatever. It could not provide any conclusive evidence either for or against the use of the mat. - the numbers were too similar.
  
 Playing the same CD in a CD player with and without the CD mat was a MAJOR difference - to all of the present, not just me and him. And all voted that sound with the mat was better - better to the level that any ABXing would look foolishly stupid waste of time. Expectation bias? They were ALL, except me of course, convinced it will not be different - one iota.  There was quite some muttering into the chin afterwards...
  
 I believe a test could be made to scientifically measure the difference. It is far less of a hoodooo than your persistent claim all amplifiers sound the same and that "perfect sound forever" is the ultimate to which humankind should ever aspire.


----------



## OddE

analogsurviver said:


> I have witnessed first hand one of the "tests" regarding errors etc on the CDs. At a pro mastering engineer, using TOTL Plextor CD burner/ROM/whatever. It could not provide any conclusive evidence either for or against the use of the mat. - the numbers were too similar.
> 
> Playing the same CD in a CD player with and without the CD mat was a MAJOR difference




-A (very!) simple test would be to play the same track with & without mat, dumping the S/PDIF output to file and later comparing the two.


----------



## analogsurviver

odde said:


> -A (very!) simple test would be to play the same track with & without mat, dumping the S/PDIF output to file and later comparing the two.


 
 Good one. Will check if my PC has SPDIF input ( I am sure it has S/PDIF output - optical, IIRC ). 
  
 BTW - my Pioneer PD-6J CD/SACD player allows for switching the S/PDIF output off - producing audibly better sonics with its own inbuilt DAC than when S/PDIF output is enabled. Any idea what causes this ( I did read about it and forgot ) ?


----------



## lamode

analogsurviver said:


> Good one. Will check if my PC has SPDIF input ( I am sure it has S/PDIF output - optical, IIRC ).
> 
> BTW - my Pioneer PD-6J CD/SACD player allows for switching the S/PDIF output off - producing audibly better sonics with its own inbuilt DAC than when S/PDIF output is enabled. Any idea what causes this ( I did read about it and forgot ) ?


 
  
 You might be referring to the fact that SA-CD players do not offer a high res digital out on S/PDIF (to prevent people making digital copies) but can use high res internally.


----------



## analogsurviver

dprimary said:


> ABX is a methodology not a specific piece of software or hardware. To ABX DSD and PCM would require more work and hardware. You could not get it down to a single variable, which I consider the gold standard. But it is not that hard either. In the past I just used a mixing console since level adjustment and routing is one of the functions of a mixing desk and I have access to plenty of them. Have one person patch it together and match levels, a second that does the switching and tracks the switching and then the listener(s) is not all that hard. Even in quick testing I calibrate, if for nothing else a fast sanity check, too often I catch something that was not set right.
> 
> Even when there is a difference, which one is correct becomes the next question. If it is barely noticeable does it even matter? Unless you were involved in the production how would you know which is closest?
> 
> I'm surprised at that it is believed all amps sound the same? I have compared amplifier sound quality more times then I can remember often somewhat blind, but not in what I would call rigorous ABX test, though we often test amps till they shutdown and test them more to see how well they recover. Sound quality doesn't matter if it stopped working. Many of them have a noticeable sound quality difference, even on different lines from the same manufacturer. Low frequency quality is the most noticeable. I am planning to do more rigorous ABX testing in the future, but I need it to be portable and we can setup and complete the testing in 30-45 minutes. That is about all the time I can spend on it and not interfere with other testing and configuration.


 
 Of course is ABX methodology and not a piece of hardware of software.  But hardware and software has to support ABXing of whatever needs to be tested this way. And "measuring equipment", "ABX box", whatever it is, should introduce at least ten times lower imperfections than whatever is the object of the test.
  
 Mixing desks are not the most audibly transparent things in the universe - and I avoid them altogether. 
  
 With small differences, such as PCM 192/24 and DSD128, the best reference is live microphone feed - and an ocassional check with own ears in the acoustic space of the recording taking place. I have stated this gazillion of times - and generally got a response live music is not the relevant test signal as it can not be infinitely repeated exactly the same - for the purposes of the almighty ABX. Phew... - as if comparing decades old recordings at which they were most certainly never present in various digital versions and masters using various software was more relevant ...
  
 Your observation that amps differ primarily in the low frequencies is quite correct - but it is an idea frowned upon in the science threads on head-fi. Limit the microphone, compress the dynamic range, pack everything into a pleasing non-disturbing CD - and a good amp has almost nothing left at which it could excel at.
  
 These words are in no way meant for you personally - I just wanted to present you the "landscape"...


----------



## lamode

analogsurviver said:


> Your observation that amps differ primarily in the low frequencies is quite correct - but it is an idea frowned upon in the science threads on head-fi. Limit the microphone, compress the dynamic range, pack everything into a pleasing non-disturbing CD - and a good amp has almost nothing left at which it could excel at.
> 
> These words are in no way meant for you personally - I just wanted to present you the "landscape"...


 
  
 What are you talking about? That is not the 'landscape' at all.


----------



## analogsurviver

lamode said:


> You might be referring to the fact that SA-CD players do not offer a high res digital out on S/PDIF (to prevent people making digital copies) but can use high res internally.


 
 No. Pioneer PD-D6J http://www.pioneer.eu/eur/products/archive/PD-D6-J/page.html
 specifically allows for the S/PDIF on 44.1/16 to be output - or not. It is called  Pure Audio Mode by the Pioneer.  Remote controlled. It does not allow for S/PDIF output of DSD under any circumstances - at least not officially.
  
 And for fact it sounds better on its own DAC when S/PDIF signal is NOT allowed at the output. 
  
 As good as it is, it is bested by using S/PDIF out into an outboard DAC - the by now rare Technics SU-A60 ( preamplifier with a VERY good digital input/DAC, spec'd as 18 bit ). The particular forte of this Technics is the dynamic range - it can really make a difference.


----------



## bigshot

My Oppo blu-ray player can play SACD and DSD files internally and pass out through analogue out, or convert to PCM 24/88 and pass it out through HDMI. DVDs and blu-rays go out through HDMI 24/192. None of it really matters though. It's just numbers on a page.


----------



## analogsurviver

lamode said:


> What are you talking about? That is not the 'landscape' at all.


 
 OH - then we must have been reading two different forums called head-fi .
  
 Or have you forgoten how many times bigshot has stated all amps sound the same, CD redbook is enough for music, dynamic range of CDs should be limited to 40-50 dB - and that anything beyond that makes no sense and is waste of resources ? Or to the same effect, with variations on the wording ?
  
 According to him, no advance over what was basically available in 1980, is not only not required, but is unwelcome and detrimental. 
  
 The only thing I agree with him is that transducers (microphones, headphones, speakers ) should be improved - and within 20-20 kHz  at first.
  
 Other than that ...- he is for maintaining comfortable status quo - I am for pushing the envelope.


----------



## Don Hills

analogsurviver said:


> You can use any good analog or genuine native DSD or PCM ( 88.2 kHz and up )  recording using >20 kHz microphones. It has been possible for at least 30 years .


 
  
 True, but irrelevant. You illustrated a point about the ultrasonics of cymbals with an example that had none. All it shows is that you don't need the ultrasonics to reproduce a realistic cymbal sound.


----------



## RRod

analogsurviver said:


> OH - then we must have been reading two different forums called head-fi .
> 
> Or have you forgoten how many times bigshot has stated all amps sound the same, CD redbook is enough for music, dynamic range of CDs should be limited to 40-50 dB - and that anything beyond that makes no sense and is waste of resources ? Or to the same effect, with variations on the wording ?
> 
> ...


 
  
 None of that changes the fact that if I take your envelope, take out the contents, and put them into a smaller envelope, no one is the wiser unless they can see me switch the envelopes.


----------



## lamode

analogsurviver said:


> OH - then we must have been reading two different forums called head-fi .
> 
> Or have you forgoten how many times bigshot has stated all amps sound the same, CD redbook is enough for music, dynamic range of CDs should be limited to 40-50 dB - and that anything beyond that makes no sense and is waste of resources ? Or to the same effect, with variations on the wording ?
> 
> ...


 
  
 So now you are generalizing about the whole forum based on what one member (allegedly) posted? Hardly the 'landscape' of the forum. I think we all want the boundaries pushed... where it will actually make a difference.


----------



## analogsurviver

don hills said:


> True, but irrelevant. You illustrated a point about the ultrasonics of cymbals with an example that had none. All it shows is that you don't need the ultrasonics to reproduce a realistic cymbal sound.


 
 Ever listened to a real drum kit ? Live, acoustic, not over speakers - not a recording over youtube ?
  
 Then try to squeeze that into CD redbook ... - and see if it is still "realistic". 
  
 DSD128 comes close - but even more would be required for the sound that no longer can be discerned from live. CD has absolutely no chance with percussion, cymbals being only the most problematic/critical part.
 Rise times on say a rim shot are extremely fast - CD can NEVER reach the proper amplitude of this pulse, it is simply too slow.


----------



## threephi

With all this back and forth I'm curious if we can perhaps establish some common ground and work our way up.  Putting aside for the moment the question of whether expectation bias applies to hi-res vs. redbook under scientifically identical playback settings, do the hi-res believers accept that expectation bias or other non-audio-related factors can *ever *influence what we hear at all?
  
 Hearing is a human sense, and it can be tricked just as our other senses can.  Watch this video, which illustrates a few classic audio illusions.  The first example is particularly interesting as it demonstrates that what we SEE can significantly change what we HEAR. 

 https://www.youtube.com/watch?v=kzo45hWXRWU

 The existence of audio illusions is why rigorous scientific methods are *required *in order to distinguish whether X (where X=hi-res, CD mat, cable burn-in, etc. etc. etc.) actually improves the raw audio, or instead only tricks our brains into perceiving that it does.  Maybe X does improve it, maybe it doesn't.  Until a scientific test is performed that isolates that one factor by itself and controls for the relative ease with which humans can be tricked by external visual and social cues, it's not proven.
  
 I think one thing that should also be kept in mind in this discussion is that it is incorrect to describe someone who is affected by an audio illusion or expectation bias as "wrong".  It's not crazy, it's human.  We are all subject to it and we all experience it in different ways.


----------



## lamode

> CD can NEVER reach the proper amplitude of this pulse, it is simply too slow.


 
  
 How would you know? Every transducer you might be using is a lot slower than CD


----------



## lamode

threephi said:


> I think one thing that should also be kept in mind in this discussion is that it is incorrect to describe someone who is affected by an audio illusion or expectation bias as "wrong".  It's not crazy, it's human.  We are all subject to it and we all experience it in different ways.


 
  
 Sounds like double speak to me. If you're wrong, you're wrong. People need to man up and admit it.


----------



## bigshot

> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> Or have you forgoten how many times bigshot has stated all amps sound the same, CD redbook is enough for music, dynamic range of CDs should be limited to 40-50 dB - and that anything beyond that makes no sense and is waste of resources ? Or to the same effect, with variations on the wording ?


 
  
 Too much is never enough! We all need ear upgrades to appreciate all the hypersonic goodness.


----------



## bigshot

threephi said:


> I think one thing that should also be kept in mind in this discussion is that it is incorrect to describe someone who is affected by an audio illusion or expectation bias as "wrong".


 
  
 Solipsism rears its ugly head!


----------



## analogsurviver

lamode said:


> How would you know? Every transducer you might be using is a lot slower than CD


 
 Get real.
  
 ANY decent phono cartridge is at least twice faster than CD. Some go beyond 100 kHz.
  
 ANY decent microphone is faster - and there are mics, capable of being used for audio, that go beyond 100 kHz.
  
 Quite a few headphones are faster - Stax being the most prolific and known, with the response at very least to 30 kHz. Usually more. There are others that go at least to 40 kHz.
  
 There are MANY loudspeakers capable of approx 40 kHz - and there are designs that reach 100 kHz and beyond. 
  
 And I am not only familiar with them, I own at the very least one from the above - so I DO know, and can  record and reproduce faster than CD. And plan to go faster than my current capability - MUCH faster.


----------



## bigshot




----------



## threephi

lamode said:


> Sounds like double speak to me. If you're wrong, you're wrong. People need to man up and admit it.


 
  
  


bigshot said:


> Solipsism rears its ugly head!


 
 I hope you guys read my entire post, and watched the video.  I'm trying to establish a common point, and a way for both sides of this argument to understand the other--that as human beings, our senses lie to and fool all of us all the time.  Including our sense of hearing.  Ever seen an optical illusion?  Are we "wrong" when they make us see something that isn't actually there?  Are people who respond positively to placebos "wrong"?  Or are you both claiming that these universal illusions and sensual tricks don't apply to either of you?  This is a subtle but important point: there is a tiny space between "wrong" and "factually incorrect" in this context, where the fallibility and malleability of our human senses come into play.
  
 I think it should be clear that I am firmly on the anti-hi-res (etc.) side of this question since there is so much scientific evidence that contradicts it, and only anecdotal, speculative, or fuzzy evidence to support it.  But I hope that everyone on both sides of this argument are man (or woman) enough to admit that as humans, we are *all *subject to audio illusions, and being influenced by one isn't a character flaw or defect.


----------



## wakibaki

analogsurviver said:


> OH - then we must have been reading two different forums called head-fi .
> 
> Or have you forgoten how many times bigshot has stated all amps sound the same, CD redbook is enough for music, dynamic range of CDs should be limited to 40-50 dB - and that anything beyond that makes no sense and is waste of resources ? Or to the same effect, with variations on the wording ?
> 
> ...


 
  
  
  Did he say that? If he didn't he should have. Every now and again you pushers push the envelope out of shape. We need an antithetical model to put up against the barrage of senseless crap based on the necessity for recording sounds that nobody can demonstrate they can hear. There's a lot to be said for the status quo. In this instance.
  
 You're just another new-is-good vandal hacking away at anybody you can find who you think understands or otherwise represents a technology you don't understand, a technology you think has been superseded. There are dozens of guys like you out there all peas in a pod, all pushing the envelope, questioning assumptions, reading up little bits on google, all thinking you're God's gift, trying to rewrite stuff I learned at my mammy's knee, for no better reason than that you swallowed some advertiser's soft soap. One of the things I learned was:- if you swap the cow for some magic beans, don't come home.
  
 Don't you get it? We're all pushing the envelope, but some of us know where to push.


----------



## dazzerfong

analogsurviver said:


> First of all, bad recording can not be "mastered" into a decent one. Better yes, good enough - no. Regardless in which format it appears. Skill and abillities of the music producer will always be the limiting factor - there is no way in the world to make a multimiked recording sounding realistic.


 
   
 OK, record better. Poor choice of words. Also, heard of binaural? Pretty damn realistic to me.
  
 Quote:


> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> And the inability of the laser optics to read the CDs well enough. Which has to be helped by a mat... - remember, you have to rip CDs somehow to get them on the hard disk - it applies here too, regardless if you no longer listen to a CD player.


 
  
 Beauty of digital is that it's either all right or all wrong. Your CD mat might help with making it all right, but that's provided there's something wrong to begin with.
  
 ------------------------------------
  
 Correct me if I'm wrong, but I thought the burden of proof was necessary here? If you challenge what is established, get proof.


----------



## bigshot

wakibaki said:


> Did he say that? If he didn't he should have.


 
  
 Here is what I said...
  
 Redbook audio is able to capture everything humans can hear. The frequency response is stone flat. The range of frequencies exceed the range of human hearing. The levels of distortion are inaudible. The dynamic range is broad enough to cover stone silence to the threshold of pain. The differences between Redbook and higher bitrate/sampling rate audio all lie beyond the range of human hearing.
  
 Once mixed properly, music doesn't generally exceed a dynamic range of 40dB. Redbook far exceeds that.
  
 Every properly designed solid state amp should sound the same... flat frequency response and inaudible distortion levels. If you come across one that doesn't achieve audible transparency, there is probably something wrong with it... either by design or it is defective and is performing out of spec.
  
 The areas where the most benefits to "pushing the envelope" lie is in the transducers and room acoustics. It is counter productive to focus on frequencies you can't hear and not equalize the ones you can. Better speakers and better room acoustics give better sound. Bigger file sizes don't.
  
 All add this one free of charge... Inaudible frequencies are INAUDIBLE. They are not necessary (or even desirable) for the purposes of listening to music in the home.


----------



## lamode

analogsurviver said:


> Get real.
> 
> ANY decent phono cartridge is at least twice faster than CD. Some go beyond 100 kHz.
> 
> ...


 
  
 Nice try but no cigar (and obviously I was talking about speakers not cartridges...) . Frequency response is NOT the only factor in determining how quickly a system can start and stop, and a fast tweeter is not enough if the other drivers are slower. Even a high-end speaker's (JMLab Utopia, $30K) step response looks like this:
  

  
 A Stax headphone is faster but not faster than the CD format itself. Top-of-the-line SR-009:


----------



## lamode

threephi said:


> Are we "wrong" when they make us see something that isn't actually there?


 
  
 Yes, and the same goes for audio illusions. Luckily we can take measures to eliminate human error.


----------



## bigshot

The threshold of perception for timing error (i.e.: group delay) is 1 to 3 ms in the core frequencies. All other things being good, that speaker you have the measurement on would still sound fast enough for human ears.


----------



## lamode

bigshot said:


> The threshold of perception for timing error (i.e.: group delay) is 1 to 3 ms in the core frequencies. All other things being good, that speaker you have the measurement on would still sound fast enough for human ears.


 
  
 I was responding to analoguesurvivor's ridiculous statement that speakers/headphones were faster than the CD format.


----------



## bigshot

lamode said:


> I was responding to analoguesurvivor's ridiculous statement that speakers/headphones were faster than the CD format.


 

 He says a lot of stuff like that. I used to read and reply to it, but stating self evident things over and over got tiresome. Now I read everyone else's comments and blow past his.


----------



## threephi

lamode said:


> Yes, and the same goes for audio illusions. Luckily we can take measures to eliminate human error.


 
 Granted, then it's a semantic issue over the word "wrong".  I don't think it's the best word to describe sensory phenomena that arise completely as a consequence of how our brains are wired to lie to us under some circumstances, since "wrong" has a little bit of a sense of moral judgement to it.  In my opinion it is a special class of error since it does not involve making a mistake or overlooking something.  In other words, sensory illusions are not caused by dishonesty or miscalculation, and we are all subject to them.  It does perhaps involve letting go of a little bit of ego to admit that your ears, or eyes, etc. might be fooling you, but that applies to everyone.

 Yes I'm getting into philosophical territory, and I'm flogging the horse more than a little bit, but only because I think this is where we might start to find common ground: that we are all subject to these tricks and sensory illusions, audio included.  Entrenched disagreements like this one usually trace down to conflicts at fundamental levels, so I'd like to see how far down that disagreement carries.

 I'd really like to hear whether people on both sides of this bridge acknowledge that audio illusions exist, and that they personally are subject to them.  I know that I sure am.

 I listen to redbook FLAC on my playback devices of choice, and believe (backed by all the science I have seen), in terms of the source data, I am getting the best possible audio that I am able to hear.  Does that confidence in the suitability of the source file influence how it sounds to me?  Probably.  Hi-res (etc) advocates clearly feel the same thing only when playing hi-res files (etc).


----------



## Krutsch

threephi said:


> Yes I'm getting into philosophical territory, and I'm flogging the horse more than a little bit, but only because I think this is where we might start to find common ground: that we are all subject to these tricks and sensory illusions, audio included.


 
  
 I feel like I am in the Matrix and I keep seeing that same black cat walk by, over and over...


----------



## bigshot

threephi said:


> I listen to redbook FLAC on my playback devices of choice, and believe (backed by all the science I have seen), in terms of the source data, I am getting the best possible audio that I am able to hear.


 
  
 And scientific tests will tell you that you probably would get the exact same sound quality from AAC 256. Feel free to follow science instead of your gut feelings.


----------



## stv014

krutsch said:


> CD-ROM error correction is different from Redbook - Redbook (i.e. CD audio) will actually attempt to handle read errors by interpolating data and presenting an approximation of the expected bits; CD-ROM (i.e. what computers use for data reads/writes also provides error detection, via Reed-Solomon, but any failure is simply missing bits - i.e. catastrophic - and the reading software is so notified via the I/O stack).


 
   
While CD-ROM data tracks do use more error correction than Red Book (each 1/75 s = 2352 bytes sector includes 276 bytes of error correction code, 4 for error detection, and only 2048 bytes of actual data without the header), the latter does in fact already have significant error protection. For each 24-byte frame of raw audio data, there is 8 bytes of Reed-Solomon error correction code (= additional 784 bytes per 2352-byte sector), and the actual physical encoding uses eight-to-fourteen modulation, which further improves error tolerance. Additionally, the data is interleaved in a way that makes a large error at a single physical location (for example, because of a scratch) distributed over multiple frames as small, easily corrected errors. Overall, with the error correction, EFM, synchronization, and subchannel data, the total data size as physical pits on the CD is more than three times larger than the raw Red Book audio.

  
 In short, it should normally be possible to read a good quality disk that is not damaged without bit errors, and it is not true that every microscopic scratch or dust particle will turn into an error that has to be interpolated.


----------



## stv014

dprimary said:


> ABX is a methodology not a specific piece of software or hardware. To ABX DSD and PCM would require more work and hardware.


 
  
 One possible solution is to convert the DSD to PCM, and then back to DSD, and compare that to the original DSD. Although this may in theory be worse than listening to the PCM version directly, if no difference is detected in ABX testing, then that suggests that the lower sample rate of the PCM does not audibly degrade the sound quality.


----------



## analogsurviver

dazzerfong said:


> Beauty of digital is that it's either all right or all wrong. Your CD mat might help with making it all right, but that's provided there's something wrong to begin with.
> 
> ------------------------------------
> 
> Correct me if I'm wrong, but I thought the burden of proof was necessary here? If you challenge what is established, get proof.


 
 For some reason, it won't let me to comment on the first half. Since you are relatively new here - I AM the binaural guy here. It is the (commercial) presure of necessity to play music over loudspeakers that is keeping me from getting exclusively binaural - but it is my personal favourite. And I doubt I will ever use multimiking - two mike recordings and no mixing desk straight into the recorder. Post production limited to pasting together the best (bits of) takes. But if the producer wants his "mic minefield" or no recording at all ... - then it will be somebody else doing the recording. I meant these multimiked recordings are beyond salvation if realistic sound is to be obtained. 
 One can mix and master ad nuseaum - the result will sound artificial no matter what.
  
 I, too, thought that the beauty of the digital it is either right or all wrong. Then, I started to worry what will happen to my CD-R masters in the long run - it never happened to you that CD or CD-R went bad and could no longer be read ? First, I tried gold CD-Rs - claimed to have longevity of 100 years +, meant for archiving.
 The first time I was recording direct to this gold CD-R in a church, I cried LOUD some most unholly words - so MUCH better did this CD-R sound. As these CD-Rs were expensive and on their way out, alI  I could get was the remaining stock of 20-30 CD-Rs - clearly not sufficient for any serious work. I went into search mode for archival CD-Rs, particularly those that do not cost an arm and a leg. It was a three months search on the internet - and I did stumble upon then emerging CD-Rs that perform even better than gold ones and cost the same as regular ones - AND  guaranteed to be good for 100 years to boot. Adding CD mat was the final icing on the cake.
  
 I asked my friends to participate in CD-R shootout listening - among original pressed CD and copies made to "regular" CD-R, gold CD-R and this "super regular priced" CD-R. And it was established the SQ is in the same, ascending order. NO ifs and buts. CD mat was used at every stage, from ripping to playback.
  
 So much for the digital being either right or completely wrong. It obviously does have shades of grey. At least when disc is involved.
  
 Recently, this subject has been brought up on this pages. As my stock of these "super regular priced" CD-Rs has dwindled to a single 100pcs box, I went into search mode - again. And was finally able to procure some more - by now, they have become so rare ( everything that is too good at too low price gets discontinued...) that search even on ebay worldwide produces - zero results. Now it is down to somebody who abandoned CD-Rs altogether is selling off his/hers leftover - totally sporadic - so it is understandable why I do not wish to share the brand and model of these discs. They will be used for direct  to CD-R recording - exclusively. For copies, lesser CD-Rs will have to be used from now on.
  
 You are correct in establishing the proof is my burden.


----------



## dazzerfong

analogsurviver said:


> For some reason, it won't let me to comment on the first half. Since you are relatively new here - I AM the binaural guy here. It is the (commercial) presure of necessity to play music over loudspeakers that is keeping me from getting exclusively binaural - but it is my personal favourite. And I doubt I will ever use multimiking - two mike recordings and no mixing desk straight into the recorder. Post production limited to pasting together the best (bits of) takes. But if the producer wants his "mic minefield" or no recording at all ... - then it will be somebody else doing the recording. I meant these multimiked recordings are beyond salvation if realistic sound is to be obtained.
> One can mix and master ad nuseaum - the result will sound artificial no matter what.
> 
> I, too, thought that the beauty of the digital it is either right or all wrong. Then, I started to worry what will happen to my CD-R masters in the long run - it never happened to you that CD or CD-R went bad and could no longer be read ? First, I tried gold CD-Rs - claimed to have longevity of 100 years +, meant for archiving.
> ...


 
 Great! I love binaural too (except for synthetic or pop): very mind-screwy in the beginning to listen to, but becomes beautiful once you get used to it.
  
 That being said, I hate to say it, but I have to agree to disagree with your statement about digital having shades of grey. If it does, then it's called analog. The very definition of digital is that it's either a 1 or 0. Of course, real life is never that discrete (pun intended), so we just have a bunch of transistors and gates that interpolate everything as either a 1 or 0. Hence my extreme dubiousness when you said the gold CD's were superior in quality (and I'm not talking about after a few dozen years).

 If you insist though, what's your theory on why gold CD's are superior? I highly doubt you just gobbled it up: you must've done some research or at least some cursory reading to help explain your experience.


----------



## Krutsch

stv014 said:


> In short, it should normally be possible to read a good quality disk that is not damaged without bit errors, and it is not true that every microscopic scratch or dust particle will turn into an error that has to be interpolated.


 
  
 Yes, it is now, today, the case. I was just making the point that CD-ROM will only go so far to recover from read errors and CD Audio will "wing it" with interpolation when it can't read all the pits. I'm sure you don't think I believe that every scratch results in a CD-ROM read failure, due in large part to improvements in the pick-up lens and the laser, as well as tracking and the software.


----------



## analogsurviver

lamode said:


> Nice try but no cigar (and obviously I was talking about speakers not cartridges...) . Frequency response is NOT the only factor in determining how quickly a system can start and stop, and a fast tweeter is not enough if the other drivers are slower. Even a high-end speaker's (JMLab Utopia, $30K) step response looks like this:
> 
> 
> 
> A Stax headphone is faster but not faster than the CD format itself. Top-of-the-line SR-009:


 
 There IS one use of PCM/DSP I will be (grundgingly...) perhaps forced to adopt - filters for loudspeakers. It has been demonstrated a really decently made PCM/DSP filter (running at least at 192/24, preferrably twice that speed ... > VERY powerful computer just for the crossover ! ) can produce an almost perfect *SQUARE WAVE, *not
 only a pulse - measured by the microphone at the listening position. There are articles floating in the net.
  
 Stax DO have a problem - their stators are too thin and too prone to resonance - the bigger, the worse - and their fastest phones are, quite understandably, the baby Stax IEM - SR001MK2 or their new succesors. Simply because they are the smallest surface and least prone to get mechanically excited. I have Lamda Pro and SR001MK2 - and although the IEM can never image nearly as well as something that presents our pinna with for all practical purposes undisturbed free air response, the ringing in Lambda(s) is clearly audible and for some critical listening I will always use SR001. They are also less fatigueing over long periods of time - like watching long movies - for the same reason. I will have to make adapter to be able to use them with desktop amp instead of the limited portable amp. I can not find the measurements for the 001 at the moment, but I do remember it is superior in pulse to the 009.
  
 It is ridiculous but true - Xiaomi Piston 2.0 ($25 delivered worldwide) HAS better pulse response than SR-009 : http://www.innerfidelity.com/images/XiaomiPiston2.pdf
 Needless to say - I own this one too. And will order the Piston 3, according to the first impressions MUCH improved IEM, ASAP - and it costs < 30$ delivered.
  
 Regarding speakers - you obviously have chosen (deliberately?) something poor in pulse response. All it takes is approx 2K$ - some Magnaplanar in that price range. Higher up, Quad ESLs are CERTAIN to run rings around Utopia regarding pulse response - as do old Acoustats etc. And to make it even more riduculous, go out and measure this little guy (that will set you back approx whole $25 or so per driver - plus enclosure, of course) http://www.visaton.com/en/industrie/breitband/frs8_8.html As far as pulse response is concerned, it will drive Utopia into the ground so hard that nothing will be left to see above the ground level...
  
 It is THE speaker for vocals - as much as I like ESLs, this 80 mm paper cone taught me a lesson that is impossible to forget. It is not large, it does not play loud, it requires VERY powerful amp relative to its diminutive size ( I drive it with 75W/ch ), it obviously can not deal with (loud) bass, it will not court my resident bats - BUT in the all important midrange and pulse, it has VERY few peers. 
  
 Needless to say, it can be augumented with a decent bass driver and some supertweeter into a full range system intended for small(er) rooms. And crossover arranged using DSP in order to pass perfect pulse and square wave at the listening position. True - I did not make it reality - YET.


----------



## stv014

krutsch said:


> Yes, it is now, today, the case. I was just making the point that CD-ROM will only go so far to recover from read errors and CD Audio will "wing it" with interpolation when it can't read all the pits.


 
   
The point was that Red Book does use Reed-Solomon error correction, it just does not have the extra layer of error correction that is present on CD-ROM data. This did not seem to be clear from your post, so I tried to explain it better for those who could have thought that Red Book error tolerance is limited entirely to good optics and interpolation. The same applies to the comment regarding any small scratches (not) resulting in data errors.


----------



## lamode

stv014 said:


> One possible solution is to convert the DSD to PCM, and then back to DSD, and compare that to the original DSD. Although this may in theory be worse than listening to the PCM version directly, if no difference is detected in ABX testing, then that suggests that the lower sample rate of the PCM does not audibly degrade the sound quality.


 
  
 It is quite wrong to compare only the sampling rates when comparing PCM and DSD. The DSD's rate is faster but it takes a lot more samples for it to express a transient, for example, than PCM, which can express it in a single sample. DSD offers no advantage but does have the disadvantage that it is not compatible with DSP. I wish DSD would die already...
  
 As for your test, yes if people can't ABX a double conversion then it's safe to say the single conversion is also inaudible.


----------



## lamode

analogsurviver said:


> There IS one use of PCM/DSP I will be (grundgingly...) perhaps forced to adopt - filters for loudspeakers. It has been demonstrated a really decently made PCM/DSP filter (running at least at 192/24, preferrably twice that speed ... > VERY powerful computer just for the crossover ! ) can produce an almost perfect *SQUARE WAVE, *not
> only a pulse - measured by the microphone at the listening position. There are articles floating in the net.


 
  
 Finally, something we can agree on. Digital crossovers are seriously good, eliminating all sorts of problems in traditional passive crossover, simple amp configurations. Only thing holding me back is the considerable extra cost.
  


> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> Simply because they are the smallest surface and least prone to get mechanically excited.


 
  
 This is also a false conclusion. If it were true, then every IEM would demonstrate exceptional impulse response. Size does not necessarily make a driver faster or slower - there are other factors involved.
  


> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> Regarding speakers - you obviously have chosen (deliberately?) something poor in pulse response.


 
  
 No, I literally posted the first high end speaker I found.


----------



## analogsurviver

lamode said:


> Finally, something we can agree on. Digital crossovers are seriously good, eliminating all sorts of problems in traditional passive crossover, simple amp configurations. Only thing holding me back is the considerable extra cost.
> 
> 
> This is also a false conclusion. If it were true, then every IEM would demonstrate exceptional impulse response. Size does not necessarily make a driver faster or slower - there are other factors involved.
> ...


 
 Finally, something we can agree on. Backed by the cost con$ideration$
	

	
	
		
		

		
			





. Were it not for the money, I would already have that digital crossover - the measurements of finished loudspeakers looked to die for.
  
 You are wrong regarding the structural resonances in ESL drivers - given the same size thickness of the perforated plates, the smaller one will be resonating at higher frequency(es) - thus less likely to be  exited by music "overtones" than "fundamentals" of the larger drivers. A somewhat cheesy but effective (if implemented better) solution would be to use conductive textile mesh for stators - as used by Sennheiser in their Unipolar 2000 and 2002 electret headphones - FAR less "stator talk" than thin metal used by Stax. To those in possesion of Martin Logan ESLs : tap with a fingernail (or plastic pencil, to make 1000 % sure no electrical shock can occur ) on the stators (or better yet, have someone tapping on the stators while you are in the listening position ) with no music playing - whatever you will hear, SHOULD NOT BE THERE IN THE FIRST PLACE. It is true that ESL diaphragm does not have resonances - but it can not but follow the stators. Which DO have structural resonances ...- which are added to the original signal. This is the path a very well executed dynamic driver might be able to use to surpass the performance of ESLs.
  
 I did mention the International audio Review's review of Acoustat vs Bowers & Wilkins speakers; it presented the same type of measurements, both for pulse and square wave. Acoustat (full range ESL ) was correct (with understandable limitations ), B&W was even worse than Utopia - going from positive and negative response in 20-20 kHz range _*5 times ! *_As IAR is adamant about copyrights etc, I can not publish these decades old results; it remains the only audio publication that does not accept any advertising - and thus CAN publish whatever the real findings. But subscriptions are steep - 200$+/year. I only check a few snippets available for free online.


----------



## lamode

analogsurviver said:


> You are wrong regarding the structural resonances in ESL drivers ..


 
  
 Hold on... I was talking about impulse response and now you are suddenly talking resonances.


----------



## analogsurviver

lamode said:


> Hold on... I was talking about impulse response and now you are suddenly talking resonances.


 
 And what impedes better impulse response if not structural resonances - at least in ESLs ? 
  
 In Stax 009 there is a good initial response, marred by "later" - and that is stators flexing. Diaphragm is so light it is more than critically damped by the air enclosed in gaps and perforations of the stators, amplifier does not ring like that, what remains are stators - which diaphragm has no other way but to follow. 
  
 In good and bad...


----------



## castleofargh

well if the damping isn't electrical, it's mechanical. but guessing who's who from an impulse is beyond me I must say  
 maybe one day when I have more experience with those stuff.


----------



## lamode

analogsurviver said:


> And what impedes better impulse response if not structural resonances - at least in ESLs ?


 
  
 Resonances, frequency response, damping, etc... many factors play a role.


----------



## Don Hills

analogsurviver said:


> Ever listened to a real drum kit ? Live, acoustic, not over speakers - not a recording over youtube ?
> 
> Then try to squeeze that into CD redbook ... - and see if it is still "realistic".
> 
> ...


 
  
 I listened to live drums often in my youth. I even became quite good at tuning kits. I quit in disgust in the 80s when the "whacking a cardboard box with a wooden spoon" sound came into fashion and people wanted their drums to make that studio sound when playing live.
  
 As for fitting the sound on a CD, it's not the CD that limits it - it's almost always the amps and/or the speakers. A realistic rimshot requires an extraordinary peak output. I have some locally produced CDs, done without compression, where some snare hits and rimshots are 20 to 30 dB higher than the average program level yet don't sound unusually loud. (And yes, they sound as they did when being recorded.)
 Regarding the frequency response of a CD, I simply don't see the necessity of attempting to reproduce something you can't hear. I do agree that a higher sampling rate eases the design of the necessary filters.


----------



## analogsurviver

don hills said:


> I listened to live drums often in my youth. I even became quite good at tuning kits. I quit in disgust in the 80s when the "whacking a cardboard box with a wooden spoon" sound came into fashion and people wanted their drums to make that studio sound when playing live.
> 
> As for fitting the sound on a CD, it's not the CD that limits it - it's almost always the amps and/or the speakers. A realistic rimshot requires an extraordinary peak output. I have some locally produced CDs, done without compression, where some snare hits and rimshots are 20 to 30 dB higher than the average program level yet don't sound unusually loud. (And yes, they sound as they did when being recorded.)
> Regarding the frequency response of a CD, I simply don't see the necessity of attempting to reproduce something you can't hear. I do agree that a higher sampling rate eases the design of the necessary filters.


 
 Well, good to know. I agree that almost any commercially available recording of drums is "doctored" in order to accommodate playback on usually available equipment. 
  
 I am VERY aware of the fact that it does not sound unusually loud - yet it requires 120 dB + SPL capability on playback. It also means that any "average" music that would otherwise get recorded close to 0 dB is by now at -20 to - 30 dB down in level - decreasing the bit depth available for "music without rimshots" by 4 or 5 bit - which IS objectionable - and is a case for 24 bit recording and not only extended frequency response.
  
 Try to record once drum kit with at least 88.2 kHz sampling - and the recording should appreciably gain in "immediacy". The catch lies in rise times - put it simply, 88.2 allows for twice faster rise times - much closer to the real thing. Going to 192 or 384 (or DSD128 and up ) is a much more subtle difference. It works even if microphones do not extend much beyond 20 kHz - and really comes to life with something that goes at least to 40 kHz.
  
 Regardless of  our hearing limited to 20 kHz or less...
  
 The filters that can be made much more simple IS the reason why converting 44.1/whatever (can be MP3 as well..) to DSD64 or higher WILL open up in stage etc by
 a definitely audible margin. Although I try to stay within the original format as much as possible ( DSD to PCM and vice versa is a lossy process and should be avoided ) , there are recordings that benefit from this - sometimes enormously so. Conversion can also be on the fly ( using jRiver 19 for example ).
  
 Simple - use your ears ...


----------



## StanD

Unless one is listening to recorded drum solos, the available DR must be shared with the entirety of the music. If everyone always listened to volume the DR found on stage, there would be a lot of people stumbling around with hearing loss. Reality strikes.
 Gee, I thought Soundstage was due to recorded spacial queues and the ability of speakers/headphones to reproduce that. Amps and DACs are transparent and shouldn't be modifying it. Since we can't tell the difference between formats, bit depths and sample rates, that's not going to affect it.


----------



## analogsurviver

stand said:


> Unless one is listening to recorded drum solos, the available DR must be shared with the entirety of the music. If everyone always listened to volume the DR found on stage, there would be a lot of people stumbling around with hearing loss. Reality strikes.
> Gee, I thought Soundstage was due to recorded spacial queues and the ability of speakers/headphones to reproduce that. Amps and DACs are transparent and shouldn't be modifying it. Since we can't tell the difference between formats, bit depths and sample rates, that's not going to affect it.


 
 Even if that real life DR would be listened to on regular basis, it is still LESS than compressed masters of today - some with the whole DR of - 3 dB or less. That means NEVER below 90 dB or so  for the duration of the recording. This IS much more dangerous for hearing loss compared to something that is on average much more quiet and peaks of 120 dB represent maybe 1% of the time of the entire recording...
  
 Amps and DACs DO modify the soundstage - IF they are, for whatever reason, limited in frequency response. It most directly affects DEPTH. One difference that would be immediately perceived between DSD64 and DSD128 ( using amplification with extended frequency response > 100 kHz ) is precisely that - DSD128 allows for MUCH better depth information - it will allow one to follow the initial strike of say percussion instrument from the initial strike all the way to the boundaries of the recording venue - and will not be pancake flat as with redbook, with depth info limited to very shallow "window" - at best. If tennis reference works for you, it is like depth is limited to say a foot or two in front and behind the net on redbook - and at least to the limit lines of the field with DSD128. That is BIG difference.
  
 Clear enough? 
  
 Compared to live feed from mike, on location, to recorder(s) set to various DSD and PCM resolution(s) - not a recording done by others and without reference to actual live sound.
  
 And this is why I insist on frequency response beyond 20 kHz.
  
 It is most audible on binaural recordings. Particularly those done by what I have termed Binaural Natural - mics worn on one's own head/ears. It just does not get more realistic than that.


----------



## lamode

> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> The catch lies in rise times - put it simply, 88.2 allows for twice faster rise times - much closer to the real thing.


 
  
 Digital audio doesn't have a "rise time". 16/44 can go from 0 to 100% signal in one sample, as can all PCM.


----------



## lamode

analogsurviver said:


> Even if that real life DR would be listened to on regular basis, it is still LESS than compressed masters of today - some with the whole DR of - 3 dB or less. That means NEVER below 90 dB or so  for the duration of the recording. This IS much more dangerous for hearing loss compared to something that is on average much more quiet and peaks of 120 dB represent maybe 1% of the time of the entire recording...


 
  
 Yes, it's true that compressed recordings are more likely to inflict hearing damage due to the higher average SPL.
  


> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> it will allow one to follow the initial strike of say percussion instrument from the initial strike all the way to the boundaries of the recording venue - and will not be pancake flat as with redbook,


 
  
 False. There is nothing audible which DSD128 can capture which Redbook can't, whether it's a direct sound of an instrument or reverberation.


----------



## analogsurviver

lamode said:


> Digital audio doesn't have a "rise time". 16/44 can go from 0 to 100% signal in one sample, as can all PCM.


 
 False. 
  
 Digital yes - but 44.1/16 has to go trough output filter - and 22.1 kHz brick filter limits rise time to approx 14 microseconds. It is problematic enough that Legato Link filtering ( basically less sharp filter with effect below 20 kHz , but allowing less sharp filtering above, yielding a small but audible improvement in speed/rise time ) was developed. Even with Legato Link, CD is at least twice too slow compared to good analog or hirez.
  
 More about format comparisons here : http://www.lindberg.no/english/collection/004.pdf


----------



## RRod

analogsurviver said:


> False.
> 
> Digital yes - but 44.1/16 has to go trough output filter - and 22.1 kHz brick filter limits rise time to approx 14 microseconds. It is problematic enough that Legato Link filtering ( basically less sharp filter with effect below 20 kHz , but allowing less sharp filtering above, yielding a small but audible improvement in speed/rise time ) was developed. Even with Legato Link, CD is at least twice too slow compared to good analog or hirez.
> 
> More about format comparisons here : http://www.lindberg.no/english/collection/004.pdf


 
  
 It's not like higher res formats get better rise times from magic; they add in frequency content, at frequencies that we can't hear.


----------



## StanD

analogsurviver said:


> Amps and DACs DO modify the soundstage - IF they are, for whatever reason, limited in frequency response. It most directly affects DEPTH. One difference that would be immediately perceived between DSD64 and DSD128 ( using amplification with extended frequency response > 100 kHz ) is precisely that - DSD128 allows for MUCH better depth information - it will allow one to follow the initial strike of say percussion instrument from the initial strike all the way to the boundaries of the recording venue - and will not be pancake flat as with redbook, with depth info limited to very shallow "window" - at best. If tennis reference works for you, it is like depth is limited to say a foot or two in front and behind the net on redbook - and at least to the limit lines of the field with DSD128. That is BIG difference.
> 
> Clear enough?
> 
> ...


 
 Yes batlike hearing is a requirement, sheesh. DACs do not remove spacial queues and if you bought a DAC with an inferior FR that chops off treble, then you must have found a real dud on fleaBay. Only people with alien DNA can hear above 20 kHz.


----------



## manbear

Analog, what headphones/ speakers do you use to play back that content over 100khz on your DSD128 recordings?


----------



## analogsurviver

stand said:


> Yes batlike hearing is a requirement, sheesh. DACs do not remove spacial queues and if you bought a DAC with an inferior FR that chops off treble, then you must have found a real dud on fleaBay. Only people with alien DNA can hear above 20 kHz.


 
 Sorry, it does not work that way. It is a typical excuse to insinuate that people who claim there is requirement above CD redbook have inferiour equipment bought at fleabay.
  
 I did indeed buy my Korgs on fleabay - because it was more or less the only way, I bought them before they were available in my country - and after. And they do, regardless being now 9 years old, still use some of the best ADCs and DACs money can buy - on fleabay - or anywhere else.
  
 I CAN NOT HEAR ABOVE 20 kHz. I have written that at least one too many times. Yet the audible impression I stated IS correct - at least to my ears. Alien or not.


----------



## manbear

Even if you can't directly hear that ultrasonic content, you value a DAC and amp's ability to reproduce it. So what headphones do you use? Im not aware of any that hypothetically go that high.


----------



## StanD

manbear said:


> Analog, what headphones/ speakers do you use to play back that content over 100khz on your DSD128 recordings?


 
 Perhaps, sonar transponders left over from the cold war. Where does he come up with this stuff?


----------



## lamode

analogsurviver said:


> False.
> 
> Digital yes - but 44.1/16 has to go trough output filter - and 22.1 kHz brick filter limits rise time


 
  
 Well now you are talking about the specifics of D/A conversion, not the format itself. Yes, different low pass filters will limit the rise time of a DAC's output, but it is irrelevant because the rise time is enough for all audible sounds and more.


----------



## StanD

analogsurviver said:


> Sorry, it does not work that way. It is a typical excuse to insinuate that people who claim there is requirement above CD redbook have inferiour equipment bought at fleabay.
> 
> I did indeed buy my Korgs on fleabay - because it was more or less the only way, I bought them before they were available in my country - and after. And they do, regardless being now 9 years old, still use some of the best ADCs and DACs money can buy - on fleabay - or anywhere else.
> 
> I CAN NOT HEAR ABOVE 20 kHz. I have written that at least one too many times. Yet the audible impression I stated IS correct - at least to my ears. Alien or not.


 
 Barring imagination, if you can't hear it, you don't need it.


----------



## analogsurviver

manbear said:


> Analog, what headphones/ speakers do you use to play back that content over 100khz on your DSD128 recordings?


 
 I did write that many times before. Stax Lambda Pro and Technics SB-RX 50 . Both are good to approx 35-40 kHz - depending how you look at them or how they are measured.
  
 DSD128 DOES NOT extend over 100 kHz - at least not flat. It requires filtering because of the out of band noise - and its real filtered response is flat to about 50 kHz, with 6dB/octave filtering above. To reach flat 100 khz with DSD, DSD256 or higher is required. Too much $$$$ for me at the moment ( Merging Hapi w/DSD input cards, approx EUR 7000 and up, depending on number of channels/cards, DSD256. I will wait till DSD512 version will not appear. )
  
 I will try to hear the new Technics speaker that came out around December/January, based on SB-RX 50 from mid 80s - this one really does extend its response to 100 kHz and is a model of linearity in the "audible" range, too. Test (flying colours) of the small speaker appeared in German Stereoplay - but both speakers share the crucial midrange/tweeter coaxial driver.
  
 http://www.technics.com/global/
  
 http://www.hifiplus.com/articles/technics-reborn-and-the-direct-drive-elephant-in-the-room/


----------



## manbear

stand said:


> Perhaps, sonar transponders left over from the cold war. Where does he come up with this stuff?




http://www.batsound.com/?p=118

Just rewire it with the finest silver and add some tuning crystals


----------



## analogsurviver

stand said:


> Barring imagination, if you can't hear it, you don't need it.


 
 I WISH it was that simple.
  
 Please go trough last few posts - the improved pulse response DOES allow much finer gradations of what is audible and what not. It will allow, for example, the audibility of string section on quiet entry a few moments before on DSD than on PCM - depending on resolution of each, this time can take from few tenths to few seconds... - it is NOT an imaginary difference that does not require ability to hear beyond 20 kHz.
  
 And forget "it must be a different master" mantra, so popular over here - derived from my own DSD original recording, the only manipulation being bouncing down to whatever format.


----------



## RRod

What does a quiet string entry have to do with rise time? Are we talking frequency response or dynamic range (the same thing in weird DSD world, I know)?


----------



## analogsurviver

lamode said:


> Well now you are talking about the specifics of D/A conversion, not the format itself. Yes, different low pass filters will limit the rise time of a DAC's output, but it is irrelevant because the rise time is enough for all audible sounds and more.


 
 According to the above, there would be no craving for moving coil phono cartridges. Yet we all know in real life it is not so. Because it SOUNDS different. Ask any seasoned analog vinyl record listener.
  
 Despite the fact that I am proponent of moving magnet carts; with a twist - managed to make them at least comparably fast to the finest MCs. While retaining their lower distortion and greater dynamic range ...


----------



## lamode

> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> It will allow, for example, the audibility of string section on quiet entry a few moments before on DSD than on PCM - depending on resolution of each, this time can take from few tenths to few seconds... - it is NOT an imaginary difference that does not require ability to hear beyond 20 kHz.


 
  
 DSD has no advantage over PCM, only disadvantages. And high res audio is only good for recording sounds too quiet to hear or ultrasonic frequencies. It's as simple as that.


----------



## lamode

analogsurviver said:


> According to the above, there would be no craving for moving coil phono cartridges. Yet we all know in real life it is not so. Because it SOUNDS different. Ask any seasoned analog vinyl record listener.
> 
> Despite the fact that I am proponent of moving magnet carts; with a twist - managed to make them at least comparably fast to the finest MCs. While retaining their lower distortion and greater dynamic range ...


 
  
 Every time I point out your mistakes you conveniently jump to a different topic. What do MC cartridges have to do with 24 bit digital audio!?


----------



## manbear

analogsurviver said:


> According to the above, there would be no craving for moving coil phono cartridges. Yet we all know in real life it is not so. Because it SOUNDS different. Ask any seasoned analog vinyl record listener.
> 
> Despite the fact that I am proponent of moving magnet carts; with a twist - managed to make them at least comparably fast to the finest MCs. While retaining their lower distortion and greater dynamic range ...




There are any number of reasons different cartridge designs could sound different. It's not clear why you single out rise times.


----------



## manbear

analogsurviver said:


> I WISH it was that simple.
> 
> Please go trough last few posts - the improved pulse response DOES allow much finer gradations of what is audible and what not. It will allow, for example, the audibility of string section on quiet entry a few moments before on DSD than on PCM - depending on resolution of each, this time can take from few tenths to few seconds... - it is NOT an imaginary difference that does not require ability to hear beyond 20 kHz.
> 
> And forget "it must be a different master" mantra, so popular over here - derived from my own DSD original recording, the only manipulation being bouncing down to whatever format.




How does an impulse response improvement in microseconds become an audible improvement on the order of tenths to a few seconds?


----------



## lamode

manbear said:


> How does an impulse response improvement in microseconds become an audible improvement on the order of tenths to a few seconds?


 
  
 It doesn't. Analoguesurvivor makes this up as he goes along.


----------



## StanD

analogsurviver said:


> I WISH it was that simple.
> 
> Please go trough last few posts - the improved pulse response DOES allow much finer gradations of what is audible and what not. It will allow, for example, the audibility of string section on quiet entry a few moments before on DSD than on PCM - depending on resolution of each, this time can take from few tenths to few seconds... - it is NOT an imaginary difference that does not require ability to hear beyond 20 kHz.
> 
> And forget "it must be a different master" mantra, so popular over here - derived from my own DSD original recording, the only manipulation being bouncing down to whatever format.


 
 So now we're back into time travel. Are you deliberately trying to entertain us or do you really believe this stuff?


----------



## analogsurviver

lamode said:


> DSD has no advantage over PCM, only disadvantages. And high res audio is only good for recording sounds too quiet to hear or ultrasonic frequencies. It's as simple as that.


 
 If the audibility of the quiet entrance of -say- string sections at the precise time, without lag to the original feed from the microphone, is not an audible advantage - then I do not know what it is. 
  
 I can vividly remember the rehearsal of the Vienna Philharmonic Orchestra in 2008 ( Bartok Concerto for Orchestra ) with Ivan Fischer at the helm; in the large concert Gallus Hall in Cankarjev Dom , Ljubljana, Slovenia

  

  
 during the opening bars of one of the movements, where EVERYBODY were playing pppp, he quietly spoke : Leise (quiet) - and that quietly spoken word , in a LARGE hall, was louder than all the members of the VPO playing at the same time (!). I have that recorded to the Sony HiMD ( 44.1/16 uncompressed redbook ) - and it would have been one hell of a lot better recorded with the DSD ( which I obtained september/october 2009 ).
  
 So much for the advantages of DSD over PCM. They ARE REAL - and do not require hearing above 20 kHz.
  
 But I agree, it is MUCH harder to work with DSD than PCM - specially if lots of editing/mastering is required.


----------



## manbear

lamode said:


> It doesn't. Analoguesurvivor makes this up as he goes along.




Yes but I'm curious all the same.


----------



## analogsurviver

lamode said:


> Every time I point out your mistakes you conveniently jump to a different topic. What do MC cartridges have to do with 24 bit digital audio!?


 
 As a group, "normal" MC cartridges are faster than , again as a group, MM cartridges. VERY broadly are MM carts comparable in bandwidth to CD (still more extended/with less sharp filtering than CD ) - while MCs are, again broadly, comparable to frequency response of (good) hirez.
  
 There are exceptions to this rule, the fastest known production cartridge was Technics EPC 100CMK4 ( or P-Mount version, EPC P100CMK4) with bandwidth over 120 kHz - and it is a MM cartridge.


----------



## RazorJack




----------



## StanD

razorjack said:


>


 
 Does popcorn sound better when on DSD?


----------



## analogsurviver

manbear said:


> There are any number of reasons different cartridge designs could sound different. It's not clear why you single out rise times.


 
 Because it is the single most important factor which distinguishes cartridges one from another.
  
 I had a crazy luck to be able to play/hear the prototype for what later emerged as Benz Ruby cartridge. It was not practical, producing only 0.03 mV/5cm/sec output voltage - getting anything approaching "normal" signal to noise ratio, hum supresion included, was (next to, even with most advanced electronics imaginable) impossible. The rest of the system included Swiss Physics preamp and heat pipe cooled real 100 W Class A amp (>> 100 kHz ) driving tall Magneplanars with full height ribbon (>40 kHz) . TT was Thorens, arm SME V. So, the entire system was limited by the performance of the ribbon tweeter of Magneplanar.
  
 This cart sounded compared to regular MCs about the same as does FM sound to AM. Everything else measured more or less similar to the regular MCs -  except for the frequency response and, consequently, rise time.
  
 Measured with the CBS STR 112 test record : rise and fall times < 3 microseconds (!)


----------



## analogsurviver

manbear said:


> How does an impulse response improvement in microseconds become an audible improvement on the order of tenths to a few seconds?


 
 Please see the link by lindberg above; CD can capture mere 16 % of the real amplitude of pulse - and, according to the playing, it takes so much more time for the CD to reach enough amplitude to become audible. If the playing is really fine at pppp, and if it prolongs for a relatively long time, CD could NEVER produce an audible sound.
 This is audible when listening to live mike feed - it starts when it should, CD (44.1/16) is lagging. Better PCM with higher sampling rates approaches DSD better - but even DXD at 384/24 only can capture 84% of the original pulse.
  
 This sounds cruel - but sorry, it is the truth. And is _raison d'etre _behind DSD.


----------



## RRod

analogsurviver said:


> If the audibility of the quiet entrance of -say- string sections at the precise time, without lag to the original feed from the microphone, is not an audible advantage - then I do not know what it is.
> 
> I can vividly remember the rehearsal of the Vienna Philharmonic Orchestra in 2008 ( Bartok Concerto for Orchestra ) with Ivan Fischer at the helm; in the large concert Gallus Hall in Cankarjev Dom , Ljubljana, Slovenia
> 
> ...


 
  
 How was it better? You could actually hear it? Were the recording levels just different?


----------



## analogsurviver

lamode said:


> It doesn't. Analoguesurvivor makes this up as he goes along.


 
 This is not the case.


----------



## analogsurviver

stand said:


> Does popcorn sound better when on DSD?


 
 If popping while being fried is what you mean, probably yes. Do not make popcorn at home.
  
 But I did record frying an egg in different formats - no kidding. It is a part of my standard "obstacle course" whenever testing new or modified recording gear.


----------



## lamode

analogsurviver said:


> then I do not know what it is.


 
  
 Nonsense, perhaps?


----------



## lamode

analogsurviver said:


> Please see the link by lindberg above; CD can capture mere 16 % of the real amplitude of pulse - and, according to the playing, it takes so much more time for the CD to reach enough amplitude to become audible. If the playing is really fine at pppp, and if it prolongs for a relatively long time, CD could NEVER produce an audible sound.
> This is audible when listening to live mike feed - it starts when it should, CD (44.1/16) is lagging. Better PCM with higher sampling rates approaches DSD better - but even DXD at 384/24 only can capture 84% of the original pulse.
> 
> This sounds cruel - but sorry, it is the truth. And is _raison d'etre _behind DSD.


 
  
 More nonsense. Seriously, this is trolling, plain and simple.


----------



## analogsurviver

rrod said:


> How was it better? You could actually hear it? Were the recording levels just different?


 
 Recording levels ARE always set equally. Same recorder, same recording level setting - just recording format selected differently.
  
 You get the same result by bouncing DSD128 to say CD. Original DSD128 will play as it should, CD would be lacking/delaying in low levels. It is the nature how DSD and PCM work.
  
 And it does not require ability to hear beyond 20 kHz.


----------



## analogsurviver

lamode said:


> More nonsense. Seriously, this is trolling, plain and simple.


 
 Explain that to musicians. Who have been practicing for umpteen "yet another mile".
  
 Specially the ones who stated, loud and clear, that they have never heard their playing/instrument better recorded.


----------



## Tuco1965

analogsurviver said:


> Explain that to musicians. Who have been practicing for umpteen "yet another mile".
> 
> Specially the ones who stated, loud and clear, that they have never heard their playing/instrument better recorded.


 
  
 Is there a master list of musicians or something?


----------



## analogsurviver

stand said:


> So now we're back into time travel. Are you deliberately trying to entertain us or do you really believe this stuff?


 
 There is a thread regarding DSD recording of a group that was raising funding on Kickstarter - somewhere here on head-fi. It was brought up by other head-fiers, I am no way alone in making this claim of DSD being superior to PCM in what you term "time travel". It IS.
  
 I do not want to entertain anybody, I have more on my plate than required already - and should be doing my "chores". But since I really do believe what I wrote - every word of it - I feel important to try to present this belief, founded on listening, to others.


----------



## analogsurviver

rrod said:


> It's not like higher res formats get better rise times from magic; they add in frequency content, at frequencies that we can't hear.


 
 Of course there is no magic involved.
  
 But please see the difference in pulse response in the lindberg link :  CD can reach only about 14 % of the actual pulse - where DSD is at 98+ %. That is one hell of a lot difference in LEVEL - and should be easily audible.


----------



## StanD

I found the perfect listener.


----------



## lamode

analogsurviver said:


> Explain that to musicians. Who have been practicing for umpteen "yet another mile".
> 
> Specially the ones who stated, loud and clear, that they have never heard their playing/instrument better recorded.


 
  
 Is that supposed to be a scientific, reasoned conclusion? They would be just as happy with the recording when downsampled to 16/44.


----------



## lamode

analogsurviver said:


> But please see the difference in pulse response in the lindberg link :  CD can reach only about 14 % of the actual pulse - where DSD is at 98+ %. That is one hell of a lot difference in LEVEL - and should be easily audible.


 
  
 What link?


----------



## lamode

I think this sums up today's posts...


----------



## Tuco1965

LMMFAO


----------



## StanD

Smack me and I'll bite you.


----------



## bigshot

I can hear hyper sonic frequencies from outer space!


----------



## analogsurviver

lamode said:


> Is that supposed to be a scientific, reasoned conclusion? They would be just as happy with the recording when downsampled to 16/44.


 
 In a way, yes. Musicians being as satisfied, that is.
  
 But each time, when a musician wanted to get out of the recording hall with the "I will hear it on CD you'll send to me" attitude - AND - upon my insisting to listen to the real deal, he/she was grateful. Some went as far as listening to the entire concert - all over again !
  
 And few have actually commented that downsampled to CD , once they received it ( regardless done to the best of my abilities, using methods also disproved and disbelieved in these pages ), the music no longer sounded as good as from the DSD master. 
  
 Musician is perhaps the last "animal" you are likely to get to listen to his/hers recording right after the recording. Most will refuse it - understandable, mentally they are too exhausted and certainly NOT eager to hear their own mistakes in full glory. 101% for sure you are not going to get ANY musician to listen to ABX of DSD vs CD bounce - and it takes approx 1/6th of a real time length of the recording to bounce it down to CD. that is approx 20-25 minutes waiting after the concert of duration of 2 hours or so - by that time, all musicians are in the inn, relaxing over dinner and adult beverages ... 
  
 That is REAL life - and "science" can think of it as it pleases.


----------



## analogsurviver

lamode said:


> What link?


 
 http://www.lindberg.no/english/collection/004.pdf


----------



## cjl

analogsurviver said:


> http://www.lindberg.no/english/collection/004.pdf


 
 You do realize that the pulse response is only lower amplitude for redbook because the pulse contains extremely high quantities of >20kHz content, and in the case of any real music (where >99% of the sound energy is <20kHz), redbook will not attenuate at all, right?


----------



## lamode

analogsurviver said:


> And few have actually commented that downsampled to CD , once they received it ( regardless done to the best of my abilities, using methods also disproved and disbelieved in these pages ), the music no longer sounded as good as from the DSD master.


 
  
 Again, a flaw in your argument so big that you could drive a Mack truck through it. Their laptop computer, or whatever they use to listen to the CD, is very unlikely to compare to your system.


----------



## analogsurviver

stand said:


> Smack me and I'll bite you.


 
 There are - or is - at least one VERY grateful bat : 
  
 http://www.wimp.com/batnursed/


----------



## lamode

cjl said:


> You do realize that the pulse response is only lower amplitude for redbook because the pulse contains extremely high quantities of >20kHz content, and in the case of any real music (where >99% of the sound energy is <20kHz), redbook will not attenuate at all, right?


 
  
 Exactly right.
  
 The argument uses ultrasonic frequencies which are irrelevant.


----------



## StanD

stand said:


> Smack me and I'll bite you.


 
  
  


analogsurviver said:


> There are - or is - at least one VERY grateful bat :
> 
> http://www.wimp.com/batnursed/


 
 Is he DSD ready?


----------



## analogsurviver

lamode said:


> Again, a flaw in your argument so big that you could drive a Mack truck through it. Their laptop computer, or whatever they use to listen to the CD, is very unlikely to compare to your system.


 
 Unfortunately, in most cases, you are right. Musicians up to approx 50 years of age ALWAYS save for another, yet better instrument - in case they are not singers. And can not/do not want to invest in audio.
 And most, before they reach about 50, maybe even 60 years, have time to listen to music mainly in - car. No nagging partner, no children requesting this or that, while travelling from A to B they finally have time to listen to some RECORDED music. This is also real life...
  
 But there is an extremely small minority of musicians, who are also audiophiles. And these tend to run around with the likes of HD-800, DSD capable DACs, analog turntables, you name it - if not higher. Naturally, I cherish their remarks, opinions, etc. They tell it as they perceive it - it is as good as it gets. But they are also humans, just much more sensitive to music related matter than most other people - they are not beyond making mistakes.


----------



## analogsurviver

stand said:


> Is he DSD ready?


 
 Bats are by default ready for approx 40 kHz and up sound - depends on exact species.


----------



## analogsurviver

tuco1965 said:


> Is there a master list of musicians or something?


 
 No. 
  
 But when people/musicians approach me for the first time with "...yet another recording engineeer....
	

	
	
		
		

		
			





..." look and , upon after hearing the recording light up with excitement, describing what they just heard with passion and sparks in their eyes - then I must be doing at least "something" right.


----------



## StanD

analogsurviver said:


> Bats are by default ready for approx 40 kHz and up sound - depends on exact species.


 
 You can keep company with the bats, I'm not prepared to lose the extended sub bass that I enjoy with my Mag Planar cans.
 Don't tell me, they can hear from DC on upwards. Oops, DC has not time component.


----------



## analogsurviver

stand said:


> You can keep company with the bats, I'm not prepared to lose the extended sub bass that I enjoy with my Mag Planar cans.
> Don't tell me, they can hear from DC on upwards. Oops, DC has not time component.


 
 As far as I know, bats have lower limit around 1 kHz and high limit around 200 kHz http://en.wikipedia.org/wiki/Hearing_range#Bats
  
 I like bass, too. And DC capable, or as near DC capable equipment as humanly possible to produce.


----------



## RRod

analogsurviver said:


> Recording levels ARE always set equally. Same recorder, same recording level setting - just recording format selected differently.
> 
> You get the same result by bouncing DSD128 to say CD. Original DSD128 will play as it should, CD would be lacking/delaying in low levels. It is the nature how DSD and PCM work.
> 
> And it does not require ability to hear beyond 20 kHz.


 
  
 When you say low levels I think "dynamic range," which the CD should more than suffice for the Concerto for Orchestra. You also mentioned that peak levels are lower for a pulse, but as others have said if a large part of the energy content of the pulse is hi-frequency, then it's energy we can't hear and the lower pulse from CD is thus audibly indistinguishable.


----------



## analogsurviver

rrod said:


> When you say low levels I think "dynamic range," which the CD should more than suffice for the Concerto for Orchestra. You also mentioned that peak levels are lower for a pulse, but as others have said if a large part of the energy content of the pulse is hi-frequency, then it's energy we can't hear and the lower pulse from CD is thus audibly indistinguishable.


 
 It is much the same as attitude "all Japanese airplanes are biplanes and all Japanese pilots wear glasses" at the outbreak of WWII. It held true - up to the first encounter with Zeros.
  
 You would be surprised how quietly the Vienna Philharmonic Orchestra can play - it is their trademark, of which they are immensely proud - quite justifiably so. And that IS below reasonable intellegibility of CD redbook. No CD you ever could have possibly heard can do this kind of mastery of playing really quietly justice. CD produces at these really low levels AND high frequencies from multiple instruments (approx 90 members, all playing ) a decisively homogenized porridge out of real thing - this is NOT as easy as playing -100 dB level 1 kHz sine wave tone - which can be properly played back with CD redbook using appropriate dithering.
  
 Sorry, but being conservative, even if it is overkill, usually does produce better results. Having "just" the capability, under the best possible of conditions vs having sufficient overkill for even the worst case scenario - which one would you choose, provided your own scalp is at stake ?


----------



## bigshot

John Cage could play 4'33'' completely quiet!


----------



## analogsurviver

bigshot said:


> John Cage could play 4'33'' completely quiet!


 
 True.
  
 But when I heard him playing it in 1990 in Switzerland, it was actually quite a loud rendition 
	

	
	
		
		

		
		
	


	




 !
  
 http://en.wikipedia.org/wiki/4%E2%80%B233%E2%80%B3


----------



## RRod

analogsurviver said:


> It is much the same as attitude "all Japanese airplanes are biplanes and all Japanese pilots wear glasses" at the outbreak of WWII. It held true - up to the first encounter with Zeros.
> 
> You would be surprised how quietly the Vienna Philharmonic Orchestra can play - it is their trademark, of which they are immensely proud - quite justifiably so. And that IS below reasonable intellegibility of CD redbook. No CD you ever could have possibly heard can do this kind of mastery of playing really quietly justice. CD produces at these really low levels AND high frequencies from multiple instruments (approx 90 members, all playing ) a decisively homogenized porridge out of real thing - this is NOT as easy as playing -100 dB level 1 kHz sine wave tone - which can be properly played back with CD redbook using appropriate dithering.
> 
> Sorry, but being conservative, even if it is overkill, usually does produce better results. Having "just" the capability, under the best possible of conditions vs having sufficient overkill for even the worst case scenario - which one would you choose, provided your own scalp is at stake ?


 
  
 I don't anyone has a problem with people doing the recording being conservative, but that's a different argument than audibility. And you're simultaneously talking about low volume playing AND high frequencies, which to me is exactly the textbook example of things that are inaudible. It's not like strings playing ppp are dishing out huge energies at frequencies above 22kHz.


----------



## lamode

analogsurviver said:


> You would be surprised how quietly the Vienna Philharmonic Orchestra can play - it is their trademark, of which they are immensely proud - quite justifiably so. And that IS below reasonable intellegibility of CD redbook.


 
  
 If it's audible, then it's not "below reasonable intellegibility of CD redbook."


----------



## analogsurviver

lamode said:


> If it's audible, then it's not "below reasonable intellegibility of CD redbook."


 
 Try that in practice - and see how it ends up ...


----------



## analogsurviver

rrod said:


> I don't anyone has a problem with people doing the recording being conservative, but that's a different argument than audibility. And you're simultaneously talking about low volume playing AND high frequencies, which to me is exactly the textbook example of things that are inaudible. It's not like strings playing ppp are dishing out huge energies at frequencies above 22kHz.


 
 Of course not - but what highs is there, gets recorded with reduced amplitude and delayed in time - smearing the original.


----------



## dazzerfong

analogsurviver said:


> Of course not - but what highs is there, gets recorded with reduced amplitude and delayed in time - smearing the original.


 
 This is getting into the recording side of business. No-one's fussed about the recording being greater than Red Book: we're talking about the product that comes to our hands now. Only an idiot would claim that recording in 44/16 is enough for production. That's not what we're talking about here though.


----------



## analogsurviver

dazzerfong said:


> This is getting into the recording side of business. No-one's fussed about the recording being greater than Red Book: we're talking about the product that comes to our hands now. Only an idiot would claim that recording in 44/16 is enough for production. That's not what we're talking about here though.


 
 No.
  
 It has the same effect if >44,1/16 is bounced down to 44.1/16. Particularly true if original recording is DSD128 or even higher.


----------



## dazzerfong

analogsurviver said:


> No.
> 
> It has the same effect if >44,1/16 is bounced down to 44.1/16. Particularly true if original recording is DSD128 or even higher.


 

 And what effect would that be? Aliasing?


----------



## StanD

analogsurviver said:


> No.
> 
> It has the same effect if >44,1/16 is bounced down to 44.1/16. Particularly true if original recording is DSD128 or even higher.


 
  
  


dazzerfong said:


> And what effect would that be? Aliasing?


 
 Dreaming.


----------



## Don Hills

analogsurviver said:


> I am VERY aware of the fact that it does not sound unusually loud - yet it requires 120 dB + SPL capability on playback. It also means that any "average" music that would otherwise get recorded close to 0 dB is by now at -20 to - 30 dB down in level - decreasing the bit depth available for "music without rimshots" by 4 or 5 bit - which IS objectionable - and is a case for 24 bit recording and not only extended frequency response.


 
  
 I agree that it makes sense to capture at 24 bit - it gives you the headroom to make sure you don't clip the ADC - but playing those peaks at 120+ dB SPL still puts your "average" level in the 90 to 100 dB range which is louder than most people listen on a sustained basis. Regularly hitting 120+ dB will quickly do enough hearing damage to ensure that capturing signal above 20 KHz becomes moot. Ask any drummer...
 And "decreasing the bit depth" is not objectionable, It does not reduce the resolution, it just raises the noise floor. There is the potential for the noise floor to become audible, but you have to get pathological with your dynamic range and playback levels to hit that.


----------



## StanD

don hills said:


> I agree that it makes sense to capture at 24 bit - it gives you the headroom to make sure you don't clip the ADC - but playing those peaks at 120+ dB SPL still puts your "average" level in the 90 to 100 dB range which is louder than most people listen on a sustained basis. Regularly hitting 120+ dB will quickly do enough hearing damage to ensure that capturing signal above 20 KHz becomes moot. Ask any drummer...
> And "decreasing the bit depth" is not objectionable, It does not reduce the resolution, it just raises the noise floor. There is the potential for the noise floor to become audible, but you have to get pathological with your dynamic range and playback levels to hit that.


 
 +1 Our friend will never admit to the practical realities of life on Earth.


----------



## dazzerfong

stand said:


> Dreaming.


 

 Here's the way I think about it: in reality, if you use a low-pass filter to chop off the higher-band stuff when recording, you're never going to anti-alias perfectly because filters are not perfectly cutoff. But, when you use a digital system to lop off the extra frequency range, it's perfect, because it's digital. If we're, say, going from 192 to 48, you're skipping (in the digital world) to every 4th sample. Unless I'm completely mistaken in this.


----------



## castleofargh

well analogsurvy does make the sound science section very much alive. you rarely see so many posts in one day.


----------



## analogsurviver

don hills said:


> I agree that it makes sense to capture at 24 bit - it gives you the headroom to make sure you don't clip the ADC - but playing those peaks at 120+ dB SPL still puts your "average" level in the 90 to 100 dB range which is louder than most people listen on a sustained basis. Regularly hitting 120+ dB will quickly do enough hearing damage to ensure that capturing signal above 20 KHz becomes moot. Ask any drummer...
> And "decreasing the bit depth" is not objectionable, It does not reduce the resolution, it just raises the noise floor. There is the potential for the noise floor to become audible, but you have to get pathological with your dynamic range and playback levels to hit that.


 
 Fair enough. I merely wanted to present case where 16 bit does run out of steam - but it is an extreme rarely encountered.
  
 A big part of my inclination towards absolute desire to record at or very close to 0 dB stems from the use of Marantz DR-6000 CD-R recorder. That thing is _*seriously *_affected in SQ if its recording level is lowered by - a single dB, let alone - horror of horrors - 20. I will look into its service manual/circuit, my hunch is that it has digital recording level volume control - and we all know what that means. But make no mistake - each and every one of my friends I used to test this audibility of 0 vs -1 dB, -1 vs -2 dB etc thing , ALWAYS, invariably, responded to the difference : WHAT did you do - now it is so much better (or worse ) than the first sample (depending on what was played first, higher or lower gain recording ). Of course, playback level was adjusted for that 1 dB difference, it was NOT playback level cheating.
  
 And guess what, it turned out that, although to a lesser degree, this decision to record as close to 0 dB as possible turned out to sound better with DSD recorders as well.  Bob Katz and his recording level etc notwithstanding. Yes, nice - in case digital recorders did not lose resolution at lower levels. But, in real life - they do.  
  
 What I do agree some DR/playback scheme, something like Replay Gain in Foobar2000 _*SHOULD *_be provided with each and every recording. Those of you who have downloaded Linn's 24Bits of Christmas will be familiar with the problem; classical uncompressed tracks, if replay gain set to normal listening level, produce downright deafening playback levels on compressed pop-ish tracks - and vice versa, if volume set to be correct even for VERY loud playback of compessed pop-ish, the uncompressed classical stuff is next to inaudible.


----------



## castleofargh

analogsurviver said:


> What I do agree some DR/playback scheme, something like Replay Gain in Foobar2000 _*SHOULD *_be provided with each and every recording. Those of you who have downloaded Linn's 24Bits of Christmas will be familiar with the problem; classical uncompressed tracks, if replay gain set to normal listening level, produce downright deafening playback levels on compressed pop-ish tracks - and vice versa, if volume set to be correct even for VERY loud playback of compessed pop-ish, the uncompressed classical stuff is next to inaudible.


 
 did you try something that use the R128 norm I find that it deals pretty well with the perceived loudness.


----------



## analogsurviver

castleofargh said:


> did you try something that use the R128 norm I find that it deals pretty well with the perceived loudness.


 
 No, not yet. But I will look into it, thank you for mentioning it.


----------



## kraken2109

Can't believe this thread is still going.


----------



## Opportunist

It's living on borrowed time...


----------



## James-uk

It's completely mental! Head and brick wall come to mind! I realise reading survivors posts that it's no wonder the world is so messed up! At least his bizarre sense of reality is hopefully limited to audio, imagine if it was important stuff and he was in a position of power! Its very worrying!


----------



## MacacoDoSom

analogsurviver said:


> What is truly embarassing ?
> 
> Once musicians, who listen ( mainly, mostly, with very few exceptions far in between ) without technical knowledge, WILL start to "thank" for "the perfect sound forever" - not only privately, but publicly so.
> 
> ...


 
 Simply place the Ultima HD Mat on top of an optical disc, with the gold side against the disc in a transport drawer or top loading mechanism. A micro-thin metallic coating dissipates static build-up, while a proprietary composite material made of an embedded carbon nano-tube structure dramatically reduces the level of disc vibration. Subtle resonances which were previously impossible to effectively control will now be completely eliminated, 16-bit/44.1K recordings will resemble 24-bit/192K recordings - instrumental timbres, harmonic density, and minimal ambient information will demand your attention in ways you never thought possible. ($239)
  
 uauuuuuuuuuuuuuuuu......... this must be what every one is looking for...


----------



## MacacoDoSom

analogsurviver said:


> It is much the same as attitude "all Japanese airplanes are biplanes and all Japanese pilots wear glasses" at the outbreak of WWII. It held true - up to the first encounter with Zeros.
> 
> You would be surprised how quietly the Vienna Philharmonic Orchestra can play - it is their trademark, of which they are immensely proud - quite justifiably so. And that IS below reasonable intellegibility of CD redbook. No CD you ever could have possibly heard can do this kind of mastery of playing really quietly justice. CD produces at these really low levels AND high frequencies from multiple instruments (approx 90 members, all playing ) a decisively homogenized porridge out of real thing - this is NOT as easy as playing -100 dB level 1 kHz sine wave tone - which can be properly played back with CD redbook using appropriate dithering.
> 
> Sorry, but being conservative, even if it is overkill, usually does produce better results. Having "just" the capability, under the best possible of conditions vs having sufficient overkill for even the worst case scenario - which one would you choose, provided your own scalp is at stake ?


 

 ...have you tried a CD mat?


----------



## analogsurviver

macacodosom said:


> Simply place the Ultima HD Mat on top of an optical disc, with the gold side against the disc in a transport drawer or top loading mechanism. A micro-thin metallic coating dissipates static build-up, while a proprietary composite material made of an embedded carbon nano-tube structure dramatically reduces the level of disc vibration. Subtle resonances which were previously impossible to effectively control will now be completely eliminated, 16-bit/44.1K recordings will resemble 24-bit/192K recordings - instrumental timbres, harmonic density, and minimal ambient information will demand your attention in ways you never thought possible. ($239)
> 
> uauuuuuuuuuuuuuuuu......... this must be what every one is looking for...


 






 nice try ... - but if it were all true of your imaginary Ultima HD Mat, for a mere measly $239, it would still be a bargain, while used with a shopping mall DVD transport beating that $5000 and up CD player used without it - wouldn't it ?
  
 Seriously, the real world mat I linked costs < 50 EUR and CAN pull the above trick (providing a decent DAC is used with that shopping mall DVD transport ). 
  
 Errata corrige: I really thought you were pulling my leg - but googling did produce this : http://www.musicdirect.com/p-109390-marigo-audio-labs-ultima-hd-signature-cd-mat.aspx
  
 And, yes, it does adress the weaknesess of "plain vanilla" carbon disc - particularly the static buildup. To anyone intending to spend similar kind of money on WHATEVER cable while still using silver discs (even if for ripping only ), listen to this Ultima HD Mat first - you may well thank you me for the suggestion.


----------



## analogsurviver

macacodosom said:


> ...have you tried a CD mat?


 
 I use it - make it THEM - for the last 7 or so years. ALWAYS - except in car CD player and when I do not want to show it and effect it can produce.
  
 For those wearing glasses - it is akin to cleaning your glasses properly with a micro fibre cloth (with CD mat ) and going about with the glasses not cleaned for a day or two (without CD mat ). 
  
 You can see/listen in both cases - question is HOW.


----------



## lamode

don hills said:


> I agree that it makes sense to capture at 24 bit - it gives you the headroom to make sure you don't clip the ADC - but playing those peaks at 120+ dB SPL still puts your "average" level in the 90 to 100 dB range which is louder than most people listen on a sustained basis.


 
  
 Yep, drums can hit 120+dB at 1 metre, but drums were never intended to be listened to from 1m.
  
 At 4 metres, that 120dB peak drops to 108dB, so that's about as loud as your peaks ever need to be at the listening position. And if that's the peak, the average should be 80-90dB, which is safe enough.


----------



## analogsurviver

lamode said:


> Yep, drums can hit 120+dB at 1 metre, but drums were never intended to be listened to from 1m.
> 
> At 4 metres, that 120dB peak drops to 108dB, so that's about as loud as your peaks ever need to be at the listening position. And if that's the peak, the average should be 80-90dB, which is safe enough.


 
 Dream on... - during the climax of "some" Shostakovich or Prokofiev, at a recent concert, SPL app on a smartphone from the friend registered solid  113 dB peak. 
  
 1st balcony of the hall in pics a few posts back, at least 20m metres from the edge of the stage, some 30 metres away from the "kitchen department" (percussion section). 
  
 It did not sound particularly loud - not at all. But two young ladies, with combined age approx equalling mine, sitting a few seats from me, jolted the whole row of seats, such was their reaction to what was written in the scores.
  
 No doubt "pampered" by tame/polite recordings 
	

	
	
		
		

		
		
	


	




.
  
 P.S: There were also moments when mere breathing of audience was on par in loudness with the playing ...


----------



## dazzerfong

Quote:


analogsurviver said:


> Dream on... - during the climax of "some" Shostakovich or Prokofiev, at a recent concert, SPL app on a smartphone from the friend registered solid  113 dB peak.
> 
> 1st balcony of the hall in pics a few posts back, at least 20m metres from the edge of the stage, some 30 metres away from the "kitchen department" (percussion section).
> 
> ...


 
 Does that mean, for the performers, assuming that sound drops 6dB every time you double the distance (hence, doubles every time you halve the distance), if you were 1 m away, the loudness would be ~140 dB. Coz if so, if your SPL measurement was right, I hope the orchestra has some damn good health insurance in place.


----------



## Krutsch

analogsurviver said:


> nice try ... - but if it were all true of your imaginary Ultima HD Mat, for a mere measly $239, it would still be a bargain, while used with a shopping mall DVD transport beating that $5000 and up CD player used without it - wouldn't it ?


 
  
 I am confident that my shopping-mall-grade, bottom-of-the-line Pioneer spinner reads with the same or better accuracy than any CD/BD/DVD transport made today or in the past, under normal conditions, even without the "Ultima HD Mat" (which I would expect to potentially damage my player).


----------



## analogsurviver

dazzerfong said:


> Quote:
> Does that mean, for the performers, assuming that sound drops 6dB every time you double the distance (hence, doubles every time you halve the distance), if you were 1 m away, the loudness would be ~140 dB. Coz if so, if your SPL measurement was right, I hope the orchestra has some damn good health insurance in place.


 
 No, I did not check the accuracy of the calibration of the friend's SPL app. But it was the same level which is the maximum Stax Lambda Pro/STM1MK2 can cleanly reproduce - 113 dB - and with which I am of course familiar. At the very least, it was not far off the true SPL value.
  
 Assumptions are the mother of all f.....s . It is true that sound drops 6dB every time the distance is doubled - in open free space. 
  
 Concert halls are made so to at least try to maintain, within reason, the "same" loudness for all. Although this can not be ever achieved, all the echoes etc pretty much achieve this.
  
 During one of the rehearsals of the same orchestra in the same hall, I asked if I can seat in within the orchestra - of course I am curious how members of the orchestra hear it. I can assure you that it is LESS noisy than in the audience - provided you do not seat in front of the brass section. They do use sound deflectors between brass section and poor souls seating in front of them.
  
 Remember, that 113 dB SPL peak reached lasted probably no more than 2 seconds - in a 2 hour 2 part concert.  With dynamic range > 80 dB (if we disregard buzzing from dimmed lights - lots and lots of them do add up to audible "Call From The Grave" by Nikola Tesla ). Average levels were well within safe limits regarding hearing damage.


----------



## analogsurviver

krutsch said:


> I am confident that my shopping-mall-grade, bottom-of-the-line Pioneer spinner reads with the same or better accuracy than any CD/BD/DVD transport made today or in the past, under normal conditions, even without the "Ultima HD Mat" (which I would expect to potentially damage my player).


 
 If that shopping-mall-grade Pioneer happens to have turntable like platter, on which you put the CD with data/pits facing up, being read by the laser from above and being supported over entire CD surface, than go to the shopping mall you bought it from and thank them by purchasing more from them. In this case, not only you do not need CD mat, you can not even use it. You already own the best possible transport - at least from the disc vibration point of view.
  
 Any other transport, which clamps the disc around the hole area only (99+% of everything available ) should benefit from the use of the mat. It is true that tolerances of the spindle that fits the CD disk can be a bit tricky to acommodate the additional 0.3 or so mm thick mat atop the regular CD - it may mean _VERY SLIGHT _enlargement of the hole in the mat by _*gently *_sanding it with very fine grit sandpaper might be required; it may be that the CD mat will not be always 100% centered on each CD/DVD/etc disc played - but the mass and eccentricity usually achieved are negligible at 1x playing speed. That eccentricity does not affect the damping action in any appreciable way. It will not damage your player.
  
 It is another story with CD-ROM drives etc. There, you MUST make sure, with draconian measures if required, that your drive does not switch, even for a moment, above 10x speed - for anything. I always check for this by NOT using CD mat - whatever action I want my PC to perform, I try a normal CD without CD mat first - PCs have super nasty tendency to always set the highest speed possible as default, particularly after switching the PC on. You have to tick ALL the boxes in the software you use for your CD-ROM drive in order to prevent spinning it faster than 10x. There ARE ways of writing a CD that absolutely do not allow you to set speed low enough - BEWARE. It is not idiot proof, but with some reasonable care it is manageable. But never attempt this in a hurry - it WILL backfire.
  
 As schiit happens, I can assure you it is not that catastrophic. I did destroy a few CDs and CD-Rs this way; they were scratched beyond polishing. It does leave mark on the CD mat - which does not affect anything but cosmetics . But it never damaged the CD burner (Yamaha CRW-F1 - I have both external USB and desktop versions ).


----------



## RRod

Sometimes I feel like Sound Science needs its own Snopes page.


----------



## analogsurviver

rrod said:


> Sometimes I feel like Sound Science needs its own Snopes page.


 
 Not being from States, I had to google what Snopes is....
	

	
	
		
		

		
		
	


	




.
  
 There is a fly in this ointment - unless you try the mat (and do not have disc transport that does not need it/precludes its use ) - you will never hear your CDs at their best.
  
 It is the second best tweak in audio - after ultrasonic cleaning of vinyl records.
  
 And one of the least expensive to boot.


----------



## RRod

analogsurviver said:


> Not being from States, I had to google what Snopes is....
> 
> 
> 
> ...


 
  
 Seeing as how I can rip my CDs twice and get null difference, I think I'll be alright.


----------



## lamode

analogsurviver said:


> Dream on... - during the climax of "some" Shostakovich or Prokofiev, at a recent concert, SPL app on a smartphone from the friend registered solid  113 dB peak.
> 
> 1st balcony of the hall in pics a few posts back, at least 20m metres from the edge of the stage, some 30 metres away from the "kitchen department" (percussion section).
> 
> ...


 
  
 Yet again, you are changing the goal line to try and win an argument. You should really see someone about that.
  
 I was talking about a snare drum rim shot, and suddenly you want to compare this with an orchestra, measured on an iPhone? LOL. Desperation has no shame, it seems.


----------



## analogsurviver

rrod said:


> Seeing as how I can rip my CDs twice and get null difference, I think I'll be alright.


 
 Twice getting it equally wrong does not make it right either.


----------



## lamode

krutsch said:


> I am confident that my shopping-mall-grade, bottom-of-the-line Pioneer spinner reads with the same or better accuracy than any CD/BD/DVD transport made today or in the past, under normal conditions, even without the "Ultima HD Mat" (which I would expect to potentially damage my player).


 
 You're missing the point!!! Your CD player only retrieves 100% of the data!! Analoguesurvivor's magic disk made of pixie dust can help you to retrieve 145%!! Yes, that's right - you can get more data than what's actually stored on the disk!!! Pixies are truly amazing creatures.


----------



## lamode

analogsurviver said:


> During one of the rehearsals of the same orchestra in the same hall, I asked if I can seat in within the orchestra - of course I am curious how members of the orchestra hear it. I can assure you that it is LESS noisy than in the audience...


 
  
 As someone who has played in an orchestra, I have to disagree, and the very notion of sound becoming louder with distance is absurd.


----------



## Krutsch

lamode said:


> As someone who has played in an orchestra, I have to disagree, and the very notion of sound becoming louder with distance is absurd.


 

 You don't need to be a musician to know that... we (well, many of us) learned that in freshman physics class.


----------



## lamode

analogsurviver said:


> Remember, that 113 dB SPL peak reached lasted probably no more than 2 seconds - in a 2 hour 2 part concert.  With dynamic range > 80 dB (if we disregard buzzing from dimmed lights - lots and lots of them do add up to audible "Call From The Grave" by Nikola Tesla ). Average levels were well within safe limits regarding hearing damage.


 
  
 For the love of all things holy... stop trolling!
  
 113dB is FAR from being "well within safe limits regarding hearing damage"


----------



## analogsurviver

lamode said:


> Yet again, you are changing the goal line to try and win an argument. You should really see someone about that.
> 
> I was talking about a snare drum rim shot, and suddenly you want to compare this with an orchestra, measured on an iPhone? LOL. Desperation has no shame, it seems.


 
 Ok, I will get an calibrated SPL meter and will ask a friend drummer to record some rim shot readings at precisely measured distance(s). And will report the readings. 
  
 I was not trying to win an argument - just saying music live can and does sound loud. Most of my audiphile friends are shocked how loud it can be for real - since they are acustomed to compressed recordings played back on equipment that can not support full dynamic range. Hence 
	

	
	
		
		

		
			





 expression on their faces - when 5X or so kg soprano pull out all the stops, for example... - for real, this  is NOT Norah Jones dreamy soothing lullalaby...


----------



## lamode

analogsurviver said:


> If that shopping-mall-grade Pioneer happens to have turntable like platter, on which you put the CD with data/pits facing up, being read by the laser from above and being supported over entire CD surface, than go to the shopping mall you bought it from and thank them by purchasing more from them. In this case, not only you do not need CD mat, you can not even use it. You already own the best possible transport - at least from the disc vibration point of view.


 
  
 Wrong again, you can't improve on 100% data retrieval.


----------



## RRod

lamode said:


> For the love of all things holy... stop trolling!
> 
> 113dB is FAR from being "well within safe limits regarding hearing damage"


 
  
 I have to defend him here: he said the *average* levels were well within safe limits. A 113dB peak is believable for the type of pieces he mentioned but probably won't last for more than a second or two.


----------



## analogsurviver

lamode said:


> As someone who has played in an orchestra, I have to disagree, and the very notion of sound becoming louder with distance is absurd.


 
 It does depend on the venue. And which section of the orchestra you are in. 
  
 I have a very interesting project in mind - wish me luck with it. It should answer such questions once and for all.


----------



## analogsurviver

lamode said:


> For the love of all things holy... stop trolling!
> 
> 113dB is FAR from being "well within safe limits regarding hearing damage"


 
 It is exposure OVER TIME - average, not peak - that are dangerous for hearing damage. And peaks or exceeding the threshold of pain, that is around 120 dB. 7 dB or more than twice below that. For more level vs time, please see : http://www.head-fi.org/t/723464/hearing-safety-and-ear-health-thread-a-diary-of-a-ear-health-noob
  
 Slamming/closing the door in a car produces such peaks on regular basis - but as it is a few times a day lasting together for perhaps a second, it will leave no permanent damage. 
  
 I did write AVERAGE level were well within safe limits - why are you now  trying to win the argument by twisting the words or taking things out of context  ?
  
 I guess I should now pre-load an app to smartphone with the composition to be played - and put mufflers the minute SPL prescribed by XY is exceeded by 1 dB ? LOL


----------



## wakibaki

analogsurviver said:


> It is exposure OVER TIME - average, not peak - that are dangerous for hearing damage. And peaks or exceeding the threshold of pain, that is around 120 dB. 7 dB or more than twice below that. For more level vs time, please see : http://www.head-fi.org/t/723464/hearing-safety-and-ear-health-thread-a-diary-of-a-ear-health-noob
> 
> Slamming/closing the door in a car produces such peaks on regular basis - but as it is a few times a day lasting together for perhaps a second, it will leave no permanent damage.
> 
> ...


 
  
 Rubbish.
  
 It's well known that a sufficiently loud explosion can perforate your eardrums. Instantaneously.


----------



## StanD

wakibaki said:


> Rubbish.
> 
> It's well known that a sufficiently loud explosion can perforate your eardrums. Instantaneously.


 
 That's probably why he thinks he can hear above 20 kHz, it's really tinnitus.


----------



## lamode

analogsurviver said:


> It is exposure OVER TIME - average, not peak - that are dangerous for hearing damage. And peaks or exceeding the threshold of pain, that is around 120 dB. 7 dB or more than twice below that. For more level vs time, please see : http://www.head-fi.org/t/723464/hearing-safety-and-ear-health-thread-a-diary-of-a-ear-health-noob


 

 I'm well aware of the relationship between hearing loss and SPL over time.
  
 If the orchestra was really producing 113dB for several seconds from 30m away, then I would expect hearing damage among the musicians, where the SPL levels would be 120dB+ (and who are rehearsing this over and over).
  
 Frankly, no-one should be taking a smartphone's SPL reading seriously, but you seem to.
  


analogsurviver said:


> Slamming/closing the door in a car produces such peaks on regular basis - but as it is a few times a day lasting together for perhaps a second, it will leave no permanent damage.


 
  
 Wow, more nonsense. Have you ever bothered to measure a car door slam? These guys have, and it measured 92dB: http://www.thehills.nsw.gov.au/files/sharedassets/public/ecm-website-documents/page-documents/major-plans-on-exhibition/160-162-excelsior-ave-castle-hill-planning-proposal-32015plp/1-acoustic_report.pdf


----------



## analogsurviver

lamode said:


> Wrong again, you can't improve on 100% data retrieval.


 
 A disc player, playing in real time,  will never tell you it did not succeed to read all the data. What will be missed will be replaced by for such a case "preloaded" error correction system - different for CD, different for CD-R - and further variations from make to make and model to model. 
  
 CD mat makes sure this error correction is triggered as few times as possible - so that what is "100% retrieved data" is actually approaching 100% and not ( 100 - X )%, that X which could not be read properly being an average , based on the last succession of the properly read 0s and 1s, supplied by one type or another of error correction system INSTEAD of the proper data. It is this where the improvement by using a CD mat is stemming from.
  
 Take any CD and put it on any of your fingers that support it with one finger alone. Than flicker the edge of the CD with a fingernail of the other hand. LISTEN to the resonance this produces. You will need quiet environment for this.
  
 The mechanical amplitude of this resonance is greater than is the pit size. If the laser optics has to constantly "hunt" for focus - the errors are so to speak prescribed.
 By playing the disc in as resonance free mode as possible, optics has MUCH easier time, produces less data dropouts and thus injects less "guessed" replacement data supplied by error correction. 
  
 Is that so hard to understand ?


----------



## analogsurviver

wakibaki said:


> Rubbish.
> 
> It's well known that a sufficiently loud explosion can perforate your eardrums. Instantaneously.


 
 Sure. 
  
 But explosions that can do that are WAY over 120 dB.
  
 http://listverse.com/2007/11/30/top-10-loudest-noises/
  
 http://www.bksv.com/NewsEvents/measurement-moments


----------



## lamode

analogsurviver said:


> A disc player, playing in real time,  will never tell you it did not succeed to read all the data. What will be missed will be replaced by for such a case "preloaded" error correction system - different for CD, different for CD-R - and further variations from make to make and model to model.
> 
> CD mat makes sure this error correction is triggered as few times as possible - so that what is "100% retrieved data" is actually approaching 100% and not ( 100 - X )%, that X which could not be read properly being an average , based on the last succession of the properly read 0s and 1s, supplied by one type or another of error correction system INSTEAD of the proper data. It is this where the improvement by using a CD mat is stemming from.
> 
> ...


 
  
 No, it's easy to understand - it's uninformed conjecture.


----------



## analogsurviver

lamode said:


> No, it's easy to understand - it's uninformed conjecture.


 
 If it did not work, I would not be using it.
  
 And I do not  paint CDs with green marker, do not demagnetize them, etc. I use what proved itself, time and time again, with my equipment and everybody else's where we tried this CD mat. Or DVD mat, if you are more of a video guy.
  
 YMMV - but only after you try it, not a priori rejecting it as an audio myth.


----------



## StanD

analogsurviver said:


> If it did not work, I would not be using it.
> 
> And I do not  paint CDs with green marker, do not demagnetize them, etc. I use what proved itself, time and time again, with my equipment and everybody else's where we tried this CD mat. Or DVD mat, if you are more of a video guy.
> 
> YMMV - but only after you try it, not a priori rejecting it as an audio myth.


 
 This is like me telling you to flap your arms and you will be able to fly. YMMV.


----------



## dazzerfong

analogsurviver said:


> Ok, I will get an calibrated SPL meter and will ask a friend drummer to record some rim shot readings at precisely measured distance(s). And will report the readings.
> 
> I was not trying to win an argument - just saying music live can and does sound loud. Most of my audiphile friends are shocked how loud it can be for real - since they are acustomed to compressed recordings played back on equipment that can not support full dynamic range. Hence
> 
> ...


 

 Obviously it sounds loud. Rock concerts frequently hit 120 dB+. The point is the dynamic range: your prescribed range of 80 dB is, frankly, bollocks. Not to mention an orchestra hitting that loud.


----------



## wakibaki

analogsurviver said:


> CD mat...


 
  
 Sorry, I didn't realise your symptoms were so florid. So are there any audio devices that you reject as scams? How 'bout the Lessloss Blackbody? Do you believe that bit-identical files sound different if ripped on a regular computer or one with a low-noise PSU? Or if the destination medium was solid-state or rotating  platter disk?
  
 This information will help us to reach a more accurate diagnosis.


----------



## OddE

analogsurviver said:


> Twice getting it equally wrong does not make it right either.


 
  
 -True, but the odds of that happening would be astronomical - and then some. (The read errors being repeatable would strongly suggest that the errors originate from the CD itself, rather than from the playback mechanism - which would, if I've understood correctly what the Magic Mat(tm) allegedly does, render it useless.)
  
 That being said, for the record I approach the mat issue with a healthy dose of skepticism. And then some.


----------



## analogsurviver

dazzerfong said:


> Obviously it sounds loud. Rock concerts frequently hit 120 dB+. The point is the dynamic range: your prescribed range of 80 dB is, frankly, bollocks. Not to mention an orchestra hitting that loud.


 
 Obviously, I will have to post some screenshot from Adobe Audition 1.5 ( which I use for determining the ultimate gain at which any DSD recording is being converted to anything else ) - for you all to see how a truly dynamic recording looks like. It is the exact opposite of those compressed files that fill almost entire area possible - peaks, peaks and more peaks...- with most of the available "space" - blank.
  
 During the recent recording of the rehearsal of Bach's Johannes Passion, a single soprano could hit only 2.5 dB less loud peaks than the whole shebang in full cry. And I call her "deceitful" soprano - because, to a listener, it does not create an impression of being loud - at all !
  
 I did not realize those things before I starting recording either - but grasped the problem very soon after. Why on earth do you think Technics SH-9020 http://www.kenrockwell.com/audio/technics/sh-9020.htm is THE most sought after and priciest component from the entire Technics Professional Series 90X0 ? For analog recordings, it is completely inescapable. Any meters on ANY R2R, pro or consumer, are, compared to 9020, there only for the show...
  
 All it takes for it to show the TRUE peak is one halfwave at 10 kHz or below - higher frequencies are NEVER louder than those below 10 kHz ( and probably even this 10 kHz is a VERY generous overkill ).
  
 It is Technics at its absolute best. Only Toshiba produced a similar unit - but that is so rare I never saw one for sale, let alone in person.


----------



## analogsurviver

odde said:


> -True, but the odds of that happening would be astronomical - and then some. (The read errors being repeatable would strongly suggest that the errors originate from the CD itself, rather than from the playback mechanism - which would, if I've understood correctly what the Magic Mat(tm) allegedly does, render it useless.)
> 
> That being said, for the record I approach the mat issue with a healthy dose of skepticism. And then some.


 
 All it takes is a listen.
  
 A very respected mastering engineer tested it with his Plextor TOTL "burner" - and whatever test used, it did NOT show any significantly different reading for errors etc.
  
 Playing 2-3 different CD (R)s , either from his recent masterings or what he considers reference recordings, made his wallet to open - 10 minutes max. And other persons present then also expressed interest to buy - I only had one spare sample at the time.


----------



## sonitus mirus

analogsurviver said:


> All it takes is a listen.


 
  
 This is not reliable, at all.


----------



## cjl

analogsurviver said:


> Try that in practice - and see how it ends up ...


 
 I've heard very quiet passages mastered with redbook. They were perfectly fine. Until you can show a double blind test showing otherwise, you really can't expect us to take your claims seriously either. The burden of proof is on the one making the new claim, and proof is not merely a collection of anecdotes.


----------



## StanD

analogsurviver said:


> All it takes is a listen.
> 
> A very respected mastering engineer tested it with his Plextor TOTL "burner" - and whatever test used, it did NOT show any significantly different reading for errors etc.
> 
> Playing 2-3 different CD (R)s , either from his recent masterings or what he considers reference recordings, made his wallet to open - 10 minutes max. And other persons present then also expressed interest to buy - I only had one spare sample at the time.


 
 Ah, a traveling salesman with only one on hand for sale. You must stock up better, think of all the lost sales volume.


----------



## HWTest

odde said:


> -True, but the odds of that happening would be astronomical - and then some. (The read errors being repeatable would strongly suggest that the errors originate from the CD itself, rather than from the playback mechanism - which would, if I've understood correctly what the Magic Mat(tm) allegedly does, render it useless.)


 

  And then, there is the AccurateRip database ...


----------



## analogsurviver

sonitus mirus said:


> This is not reliable, at all.


 
 Sorry, if the clear preference for mat of a recording/mastering engineer, bassist and sax player who play on that recording, put eventually to CD , are not enough to at least give it a shadow of a doubt, then I do not know what could convince you. 
  
 I could list many, almost countless testimonies from musicians regarding "CD"s - I used to give them copies of their playing recorded to CD-Rs with and without the mat, marked so that they could possibly not know which is which ( I kept a small notebook ) - and asked them which copy they liked better. It was way over 90 % votes for the one made with the mat - despite they all played it at first of course without the mat. Only the notebook knew whether "A" denotes CD-R recorded with and "B" without the CD mat - or vice-versa.
  
 I think it is enough on the CD mat theme. Info is there, I am NOT affiliated with ANY of the sellers, I am NOT a manufacturer, occasionally I would order either for me or for friends, with a sample or two in addition for "stock" or reserve should I break any of my workhorse CD mats. 
  
 HOUG !


----------



## analogsurviver

hwtest said:


> And then, there is the AccurateRip database ...


 
 Of commercially available recordings ...


----------



## Krutsch

analogsurviver said:


> Sorry, if the clear preference for mat of a recording/mastering engineer, bassist and sax player who play on that recording, put eventually to CD , are not enough to at least give it a shadow of a doubt, then I do not know what could convince you.
> 
> I could list many, almost countless testimonies from musicians regarding "CD"s - I used to give them copies of their playing recorded to CD-Rs with and without the mat, marked so that they could possibly not know which is which ( I kept a small notebook ) - and asked them which copy they liked better. It was way over 90 % votes for the one made with the mat - despite they all played it at first of course without the mat. Only the notebook knew whether "A" denotes CD-R recorded with and "B" without the CD mat - or vice-versa.
> 
> ...


 
  
 No, I think there is much more to discuss on the benefits of your CD Mat. Head-Fi has a thread where you can discuss tweaks like your CD Mat and Shakti Stones and no discussion of ABX is allowed. That's the place for you, my friend, and there are many enthusiasts there waiting with open wallets to learn more.


----------



## analogsurviver

stand said:


> Ah, a traveling salesman with only one on hand for sale. You must stock up better, think of all the lost sales volume.


 
 Yes, I could have done better.
  
 MUCH better infact.
  
 People still ask from time to time .


----------



## StanD

stand said:


> Ah, a traveling salesman with only one on hand for sale. You must stock up better, think of all the lost sales volume.


 
  
  


analogsurviver said:


> Yes, I could have done better.
> 
> MUCH better infact.
> 
> People still ask from time to time .


 
 You might throw in a free Pet Rock with each sale.


----------



## analogsurviver

krutsch said:


> No, I think there is much more to discuss on the benefits of your CD Mat. Head-Fi has a thread where you can discuss tweaks like your CD Mat and Shakti Stones and no discussion of ABX is allowed. That's the place for you, my friend, and there are many enthusiasts there waiting with open wallets to learn more.


 
 OK - would dbt using two same type cd transports , playing two equally made CD-R copies or originally pressed CDs, one with and one without the mat, played simoultaneously for A/B trough a single DAC, played over speakers to a panel of say three conductors/musicians, who perform on these CD(R)s - be good enough for you ?
  
 I - or better yet - a third party person, knowing only "A" and "B" (and how to note the results correctly) would record the results. It can not be ABX under Foobar2000, which can only compare two files from PCs disk and not external digital input(s). At least, I do not have such a PC that accepts two S/PDIF inputs.


----------



## sonitus mirus

analogsurviver said:


> OK - would dbt using two same type cd transports , playing two equally made CD-R copies or originally pressed CDs, one with and one without the mat, played simoultaneously for A/B trough a single DAC, played over speakers to a panel of say three conductors/musicians, who perform on these CD(R)s - be good enough for you ?
> 
> I - or better yet - a third party person, knowing only "A" and "B" (and how to note the results correctly) would record the results. It can not be ABX under Foobar2000, which can only compare two files from PCs disk and not external digital input(s). At least, I do not have such a PC that accepts two S/PDIF inputs.


 
  
 Why not just have somebody put one of your favorite CDs in the player, randomly with or without the mat, and then listen to the music as long as you like and try and determine if the mat is in place or not?  As long as you have no idea whether the mat is being used or not, that should be sufficient to remove bias.  
  
 If you guess correctly 8 times in a row, I'd be impressed, and I would be satisfied that the mat is making a difference that you can identify.


----------



## HWTest

analogsurviver said:


> Of commercially available recordings ...


 
  
 And? Is it a problem to rip and submit a (not commercially available) CD on two or more drives?


----------



## analogsurviver

sonitus mirus said:


> Why not just have somebody put one of your favorite CDs in the player, randomly with or without the mat, and then listen to the music as long as you like and try and determine if the mat is in place or not?  As long as you have no idea whether the mat is being used or not, that should be sufficient to remove bias.
> 
> If you guess correctly 8 times in a row, I'd be impressed, and I would be satisfied that the mat is making a difference that you can identify.


 
 That is easier to do than what I proposed.
  
 I will try to make it happen at dealer's ASAP - he tests me, I test him. This makes for double the result. I jost spoke with him on the phone and we will do it ASAP.
  
 He told me another thing: ripping the CD with or without the mat to a file on PC etc produces different results. I could do a few and load them to some cloud , so that everyone can compare the files at the leisure of his/hers home.
  
 How long can these excerpts be, due to copyright reasons ? IIRC, 30 seconds are allowed - I do not want to have any trouble regarding this.


----------



## analogsurviver

hwtest said:


> And? Is it a problem to rip and submit a (not commercially available) CD on two or more drives?


 
 No.


----------



## bigshot

When I first got a CD burner, I used it for making vital backups as part of my job as a media archivist. I would do checksum verification on every burn I made. If I was creating an audio CD, I would first master it as a disk image and burn it from that just so I could do bit for bit verification. As part of this job, my volunteers and I burned well over 10,000 disks, three at a time using the internal drives of three iMacs, using Taiyo Yudin stock, both CDs and DVDs. We never got a single verification error.
  
 I ran out of Taiyo Yudin one day, so I sent a volunteer to Frys to pick up a batch of blanks to hold us over until the next shipment arrived from Monoprice. He came back with a spindle of Memorex. I got an error on the very first burn. I burned a few more and realized that it was like pulling teeth to get bit for bit verification with the Memorex blanks. They went straight in the trash. By the way, every one of those burns I made on Taiyo Yudin years and years ago are still good.
  
 Error free burning has everything to do with the quality of the stock you use, not magical felt stickers or green pens.


----------



## lamode

analogsurviver said:


> YMMV - but only after you try it, not a priori rejecting it as an audio myth.


 
  
 When a $20 CD-ROM drive can retrieve 100% data from a disc, this becomes pointless. You can't improve on 100%.
  
 Besides, why keep going on about CDs? That become an obsolete playback medium 15 years ago, as soon as lossless files and ripping took over.


----------



## lamode

> Originally Posted by *analogsurviver* /img/forum/go_quote.gif
> 
> Why on earth do you think Technics SH-9020 http://www.kenrockwell.com/audio/technics/sh-9020.htm is THE most sought after and priciest component from the entire Technics Professional Series 90X0 ?


 
  
 ... because people are crazy about retro gear. A good digital meter would run circles around this unit.


----------



## analogsurviver

bigshot said:


> When I first got a CD burner, I used it for making vital backups as part of my job as a media archivist. I would do checksum verification on every burn I made. If I was creating an audio CD, I would first master it as a disk image and burn it from that just so I could do bit for bit verification. As part of this job, my volunteers and I burned well over 10,000 disks, three at a time using the internal drives of three iMacs, using Taiyo Yudin stock, both CDs and DVDs. We never got a single verification error.
> 
> I ran out of Taiyo Yudin one day, so I sent a volunteer to Frys to pick up a batch of blanks to hold us over until the next shipment arrived from Monoprice. He came back with a spindle of Memorex. I got an error on the very first burn. I burned a few more and realized that it was like pulling teeth to get bit for bit verification with the Memorex blanks. They went straight in the trash. By the way, every one of those burns I made on Taiyo Yudin years and years ago are still good.
> 
> Error free burning has everything to do with the quality of the stock you use, not magical felt stickers or green pens.


 
 I certainly agree discs themselves are important. Taiyo Yuden (proper name) is a well respected top notch manufacturer.
  
 Even with the best of discs, CD mat will help. All it takes is a good listen. If you need,you can compare the results of ripping, or/and playback. This is part of the reason why a CD-R copy can be made to sound better than the original pressed CD - by using CD mat for both, reading and burning at as slow speed as possible.


----------



## lamode

analogsurviver said:


> Sorry, if the clear preference for mat of a recording/mastering engineer, bassist and sax player who play on that recording, put eventually to CD , are not enough to at least give it a shadow of a doubt, then I do not know what could convince you.


 
 You can't convince us that you can achieve more than 100% accuracy, sorry.


----------



## lamode

analogsurviver said:


> Of commercially available recordings ...


 

 You are (deliberately) missing the point. AccurateRip can confirm that people are achieving 100% error-free data retrieval with their drives. No amount of pixie dust or carbon fibre power plugs can make a difference.


----------



## StanD

lamode said:


> You can't convince us that you can achieve more than 100% accuracy, sorry.


 
 Not even 101%? Bummer, my life is ruined.


----------



## analogsurviver

lamode said:


> ... because people are crazy about retro gear. A good digital meter would run circles around this unit.


 
 Please read the specs before making such claim ... - and verification of the accuracy in the review of the stock, non refurbished unit - more than 30 years old.
  
 For starters, anything digital that could aspire to achieve this performance would have to use >> than CD redbook to begin with... - its frequency response is extended, enough so even for the fastest currently commercially available microphones: 8 Hz to 93 kHz at +-1 dB.
  
 This can not be had with 192/24 ... - it would not be so flat to 93 kHz in real life, and fall off as a brick beyond 96 kHz, where analog meter rolls of gently above 100 kHz.


----------



## Krutsch

I think at this point, it's very clear that @analogsurviver is trolling and seeing how long he get all of us to continue to respond to his insane commentary.
  
 EDIT: and so far, he's put on an epic performance.


----------



## HWTest

analogsurviver said:


> He told me another thing: ripping the CD with or without the mat to a file on PC etc produces different results.


 
  
 There is also a very nice software called "CD Vergleich", it compares CDs in two drives or a CD with a ripped image.


----------



## analogsurviver

krutsch said:


> I think at this point, it's very clear that @analogsurviver is trolling and seeing how long he get all of us to continue to respond to his insane commentary.


 
 Wrong.


----------



## StanD

krutsch said:


> I think at this point, it's very clear that @analogsurviver is trolling and seeing how long he get all of us to continue to respond to his insane commentary.
> 
> EDIT: and so far, he's put on an epic performance.


 
  
  


analogsurviver said:


> Wrong.


 
 You're enjoying the attention.


----------



## lamode

analogsurviver said:


> Please read the specs before making such claim ... - and verification of the accuracy in the review of the stock, non refurbished unit - more than 30 years old.
> 
> For starters, anything digital that could aspire to achieve this performance would have to use >> than CD redbook to begin with... - its frequency response is extended, enough so even for the fastest currently commercially available microphones: 8 Hz to 93 kHz at +-1 dB.
> 
> This can not be had with 192/24 ... - it would not be so flat to 93 kHz in real life, and fall off as a brick beyond 96 kHz, where analog meter rolls of gently above 100 kHz.


----------



## analogsurviver

stand said:


> You're enjoying the attention.


 
 No. I feel sorry for people who can not or wish not to think outside established and safe limitations.
  
 I will upload some rips with and without mats to some cloud, and am really interested what the results will be.
 And collect the results and generate statistics etc.
  
 Beyond that, I will be minding my own business from now on - if all that comes back - well, almost all - is negative, why bother ?


----------



## analogsurviver

hwtest said:


> There is also a very nice software called "CD Vergleich", it compares CDs in two drives or a CD with a ripped image.


 
 Thank you for that ! Will try it tomorrow.


----------



## headdict

I have learned three things from the last 50 or so pages on this thread.
  
 1. How to use the block feature.
 2. There is no way to block someone's posts along with all replies to those posts.
 3. How pointless it is to block a troll as long as everybody else keeps feeding him.
  
 Maybe the best thing that can happen to this thread is being locked. Or maybe I'm too impatient and should wait and see what the next 50 pages will be like.


----------



## StanD

headdict said:


> I have learned three things from the last 50 or so pages on this thread.
> 
> 1. How to use the block feature.
> 2. There is no way to block someone's posts along with all replies to those posts.
> ...


 
 I think he's taking a vacation from this thread so we can commence with less nonsense.


----------



## bigshot

I think I have a solution to the blather and attention grabbing. Let's go back to the beginning and start out right again!
  
 Quote:


gregorio said:


> It seems to me that there is a lot of misunderstanding regarding what bit depth is and how it works in digital audio. This misunderstanding exists not only in the consumer and audiophile worlds but also in some education establishments and even some professionals. This misunderstanding comes from supposition of how digital audio works rather than how it actually works. It's easy to see in a photograph the difference between a low bit depth image and one with a higher bit depth, so it's logical to suppose that higher bit depths in audio also means better quality. This supposition is further enforced by the fact that the term 'resolution' is often applied to bit depth and obviously more resolution means higher quality. So 24bit is Hi-Rez audio and 24bit contains more data, therefore higher resolution and better quality. All completely logical supposition but I'm afraid this supposition is not entirely in line with the actual facts of how digital audio works. I'll try to explain:
> 
> When recording, an Analogue to Digital Converter (ADC) reads the incoming analogue waveform and measures it so many times a second (1*). In the case of CD there are 44,100 measurements made per second (the sampling frequency). These measurements are stored in the digital domain in the form of computer bits. The more bits we use, the more accurately we can measure the analogue waveform. This is because each bit can only store two values (0 or 1), to get more values we do the same with bits as we do in normal counting. IE. Once we get to 9, we have to add another column (the tens column) and we can keep adding columns add infinitum for 100s, 1000s, 10000s, etc. The exact same is true for bits but because we only have two values per bit (rather than 10) we need more columns, each column (or additional bit) doubles the number of vaules we have available. IE. 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024 .... If these numbers appear a little familiar it is because all computer technology is based on bits so these numbers crop up all over the place. In the case of 16bit we have roughly 65,000 different values available. The problem is that an analogue waveform is constantly varying. No matter how many times a second we measure the waveform or how many bits we use to store the measurement, there are always going to be errors. These errors in quantifying the value of a constantly changing waveform are called quantisation errors. Quantisation errors are bad, they cause distortion in the waveform when we convert back to analogue and listen to it.
> 
> ...


----------



## RazorJack

tl;dr version:
  
 - CD is more than capable of providing the best required sound quality in stereo for humans. 
  
 - There is no evidence to support the contrary.
  
 - Some people are great at fooling themselves.
  
 - Some of them are really good at wasting time of others.


----------



## Roly1650

Smart post, canny move....


----------



## StanD

Back on track/reality.


----------



## RRod

Back in reality, I can't pick up a 2kHz sine wave in my listening room at much under -85dBFS. As even my most dynamic music clocks in at a minimum RMS level of -65dBFS, this doesn't really bother me. It does make me wonder why people make such a huge deal about dither/shaping when comparing dynamic range of 16 vs 24bit. Fact is I could probably convert most of my stuff to 15bits without dither and still not hear a difference.


----------



## StanD

rrod said:


> Back in reality, I can't pick up a 2kHz sine wave in my listening room at much under -85dBFS. As even my most dynamic music clocks in at a minimum RMS level of -65dBFS, this doesn't really bother me. It does make me wonder why people make such a huge deal about dither/shaping when comparing dynamic range of 16 vs 24bit. Fact is I could probably convert most of my stuff to 15bits without dither and still not hear a difference.


 
 Because they need something to yak about. What about the ambient noise level in a room when open back cans are used? People ignore that as well. I can hear myself breathing, that's got to cut into the ambient noise level. Though I don't think I'm willing to drop to 15 bits.
 I believe the background of a quiet bedroom at night is 30 dBSPL.


----------



## MacacoDoSom

analogsurviver said:


> Sorry, if the clear preference for mat of a recording/mastering engineer, bassist and sax player who play on that recording, put eventually to CD , are not enough to at least give it a shadow of a doubt, then I do not know what could convince you.
> 
> I could list many, almost countless testimonies from musicians regarding "CD"s - I used to give them copies of their playing recorded to CD-Rs with and without the mat, marked so that they could possibly not know which is which ( I kept a small notebook ) - and asked them which copy they liked better. It was way over 90 % votes for the one made with the mat - despite they all played it at first of course without the mat. Only the notebook knew whether "A" denotes CD-R recorded with and "B" without the CD mat - or vice-versa.
> 
> ...


 

 ...Neil Young (and some other musicians) claim that listening a CD is like listening under water compared to 192/24 or 96/24...
     -and that, doesn't make it a reality...


----------



## analogsurviver

macacodosom said:


> ...Neil Young (and some other musicians) claim that listening a CD is like listening under water compared to 192/24 or 96/24...
> -and that, doesn't make it a reality...


 
 That is a clearly exagerrated description of the real relationship.
  
 But it can sound like that if one concentrates on the differences only - which is not in context of the music. Music should be first, SQ second.
  
 But what is missing in SQ from CD redbook can significantly contribute to the realism of the reproduced sound.


----------



## ChrisIsAwesome

Sound......Science........Forum.

Back up your statements using evidence and facts not anecdotal claims. 

Just a friendly reminder to all. This thread reads like something out of cables/tweaks.


----------



## spruce music

rrod said:


> Back in reality, I can't pick up a 2kHz sine wave in my listening room at much under -85dBFS. As even my most dynamic music clocks in at a minimum RMS level of -65dBFS, this doesn't really bother me. It does make me wonder why people make such a huge deal about dither/shaping when comparing dynamic range of 16 vs 24bit. Fact is I could probably convert most of my stuff to 15bits without dither and still not hear a difference.


 
 http://audiosciencereview.com/forum/index.php?threads/at-what-level-is-noise-heard-in-your-system.1013/
  
 You could try the above signals.  Steps down in level 10 db at a time until you can't hear the noise mixed in with some music.


----------



## RRod

spruce music said:


> http://audiosciencereview.com/forum/index.php?threads/at-what-level-is-noise-heard-in-your-system.1013/
> 
> You could try the above signals.  Steps down in level 10 db at a time until you can't hear the noise mixed in with some music.


 
  
 Will give a try. Trying things with pseudo-white/pink noise of course means you have to jack up the volume even more, so I took something like a 2k/4k sine/square as a kind of "worst case" for truncation distortion without dither. In a 35dBA room, the pot simply has to be really high to hear anything, any anything other than my most dynamic classical stuff is simply ear destroying at that level.


----------



## upstateguy

The same song at  24/192, 16/96, and 16/44.1   Looks like a lot of nothing much up there except taking up a lot of space.
  
 click to make larger


----------



## icebear

upstateguy said:


> The same song at  *24/192*, 16/96, and 16/44.1   Looks like a lot of nothing much up there except taking up a lot of space.
> 
> click to make larger


 
  
 THERE is the proof ... clearly more black background above 75khz in the 24/192khz file.
 Blacker black in this region is like audophile gold, even when it seems just black
	

	
	
		
		

		
			





 
 And don't forget the all important headroom for the ultra soncis.
	

	
	
		
		

		
		
	


	








  <--- and btw this is 24bit audiophile grade popcorn, not just some cinema grade stuff!


----------



## castleofargh

I love spectrograms in 24bits.


----------



## StanD

But the latest fad is R2R DACs and they're only available in 16 bits. That means 24 bit DACs are out of fashion.


----------



## watchnerd

chrisisawesome said:


> Sound......Science........Forum.
> 
> Back up your statements using evidence and facts not anecdotal claims.
> 
> Just a friendly reminder to all. This thread reads like something out of cables/tweaks.


 
  
 Quite the necro post you did there.


----------



## analogsurviver

rrod said:


> Will give a try. Trying things with pseudo-white/pink noise of course means you have to jack up the volume even more, so I took something like a 2k/4k sine/square as a kind of "worst case" for truncation distortion without dither. In a 35dBA room, the pot simply has to be really high to hear anything, any anything other than my most dynamic classical stuff is simply ear destroying at that level.


 
 Yes, it does take uncompressed recordings for any noise issues to be picked up by the ear. And that, most usually, will be classical music.
  
 One can flip the coin into another direction; on some smartphones, the output power for headphones is barely enough for compressed pop/rock stuff - at least with headphones that do not have a well above average efficiency/sensitivity. Any high dynamic range recording would simply be too quiet even at the maximum volume control setting.
  
 I have found sometimes 10 dB+ "headrooom" ( ability to set the overall gain of the playback system while still remaining below clipping point ) is required for my recordings - even when compared with the requirement for the commercially available audiophile label recordings - for the same perceived loudness. There are many speaker setups that will simply run out of steam under such scenario.
  
 The sheer amount of abuse of compression used nowadays is amply highlighted by the clear preference for the (original first press ) vinyl vs CD version of the same recording. Although I will never accept RBCD to be "enough", if used correctly, it can and does exceed the dynamic range of vinyl - yet recent practices in mastering show that this potential is not only not explored, but being suppressed by any means at disposal.
  
 Please note I am looking for truth in recording - as unhampered by the limits of technology as humanly possible.  At no point in time has technology been offering better capabilities to capture the sound in its full measure as today - yet most of the time technology is being used to "squeeze" the sound within the capabilities of the medium and/or playback equipment.
  
 That 10dB+ capability required does not come without the price tag...


----------



## watchnerd

stand said:


> But the latest fad is R2R DACs and they're only available in 16 bits. That means 24 bit DACs are out of fashion.


 
  
 While I'm not a believer in the alleged inherent superiority of R2R, that's not actually factually correct.  For example, the AD5791 used in the Schiit Yggy is a 20 bit chip (+1 LSB).


----------



## gregorio

I notice that you didn't do as requested and backed-up your statements, instead you deflected. Unfortunately, your deflection is just as nonsensical as the nonsensical statements you're refusing to back up?!
  
 Quote:


analogsurviver said:


> That 10dB+ capability required does not come without the price tag...


 
  
 Actually, it comes with two price tags and both of them are exorbitant! Let's take the example from the post to which you were responding, a room with a noise floor of 35dBA. The digital noise of 16bit is at -96dBFS, so that -96dBFS level has to at least equal the 35dBA noise level of the room to be audible. If -96dBFS = 35dBA then the peak level of the 16bit recording (0dBFS) is 96dB above 35dBA, which is 131dB. Add your 10dB headroom and that's 141dB. So price tag #1. A system capable of 141dB peak output, while it's own self noise is below the 35dB noise floor of the room. How much would such as system cost? Are there any systems capable of that performance at any price? Price tag #2. 131dB is above the threshold of pain and would rapidly lead to permanent hearing damage. Deafness is a pretty hefty price, wouldn't you say?!
  
 Quote:


analogsurviver said:


> [1] Please note I am looking for truth in recording - as unhampered by the limits of technology as humanly possible.  ... [2] yet most of the time technology is being used to "squeeze" the sound within the capabilities of the medium and/or playback equipment.


 
  
 1. A full symphony orchestra has a dynamic range of roughly 55dB or so, maybe around 70dB if we include extreme transient peaks but even the 70dB figure is more than ten times less dynamic range than 16bit offers. So what "truth" are you hoping to find of which CD/16bit is incapable?
  
 2. What? You think record labels should distribute recordings beyond the capabilities of playback equipment? Recordings that are not truthful, where you can't actually hear all of it and if you could, would cause pain and injury?
  
 Doesn't any of this sound at least a tad nonsensical to you?
  
 G


----------



## goodvibes

There should be dither (intentional low level noise) added when rendered back so that can cause a very minor difference but for all intents and purposes that's correct, especially on less revealing 16 bit​.
	

	
	
		
		

		
		
	


	




​


----------



## analogsurviver

gregorio said:


> 1. A full symphony orchestra has a dynamic range of roughly 55dB or so, maybe around 70dB if we include extreme transient peaks but even the 70dB figure is more than ten times less dynamic range than 16bit offers. So what "truth" are you hoping to find of which CD/16bit is incapable?
> 
> 2. What? You think record labels should distribute recordings beyond the capabilities of playback equipment? Recordings that are not truthful, where you can't actually hear all of it and if you could, would cause pain and injury?
> 
> ...


 
 Just a quick reply.
  
 1. I have not been necesirilly referreing to the 35dB (or so ) noise floor of a quiet home room. This is head-fi, with reasonable headphones this noise floor can be at least 10 dB lower - which brings us to the grand total of maximum 131 dB ( and correspondingly less, given better isolation in headphones ). PLEASE do note that those 131 dBs would have been reached for a grand total of less than one second for the duration of a concert - say 2 hours - and the average level from an uncompressed recording would certainly have been LESS than most of us are familiar with from the commercially available recordings. There is VERY little (acoustical )  music that exceeds the loudness of a standing ovation applause ( really big symphonic works, with percussion and organ etc as finale, like Mahler*s 2nd ) - and as far as I am aware, nobody yet did suffer pain, let alone an injury or permanent hearing damage from an applause heard from within the audience.
  
 2. Yes, I do think record labels should distribute truthful recordings. With the necessary warning/caution CLEARLY written up to which SPL/frequency range the equipment should be capable of reproducing any given recording.  Dynamic range is an essential part of the music - and squashed orchestras and opera singers grander than these orchestras are in no way doing justice to reality . I agree that some form of "replay gain" is required - to keep things under control and reproduced at as nearly exact level as intended by the artist(s) - to prevent playback of say more than 3 dB above should be value.
  
 In analogue days, it was next to inescapable to cut records with lower than optimum amplitude/velocity - because the majority of record buyers do not posses SOTA analog playback equipment adjusted to max performance. Such well reasoned limitations no longer apply with digital, where even the most inexpensive digital devices should trouble free reproduce up to 0 dBFS . 
  
 The limitations of the rear end ( amps, speakers, headphones ) remain the same - and can be further compounded by really high quality digital recordings, which can capture all the bass and all the loudnes the recording source is capable of producing. There USED TO BE testing of loudspeakers for the maximum SPL vs frequency ( at given max distortion, say 10 % ) back in the day, there used to be testing of dynamic linearity of loudspeakers ( much the same way digital players are being tesed ) - yet, "somehow" all of these measurements, which, once for a change, DID tell something how we can expect any given tested loudspeaker to perform in real life, disappeared from the reviews. Sorry, I do not care much for a loudspeaker that has very low linear deviations ( flat frequency response ) - but can not take any real dynamics. Or one that has to be "turned up" to have decent definition - where played at normal level for any given genre of music such speaker sounds muffled and lifeless. The speakers I am (unfortunately.. - I do not own them and they are illegal in Europe) familiar with that sport the widest and most true to life dynamics are Beveridge 2SW - and they can not play above say level required for a (larger) Mozart symphony. The problem of residual noise floor of real rooms , unfortunately, remains - and do not allow such extraordinarily good transducers to fully reveal what they are actually capable of. Yet given a chance to get Bevs for permanent, I would not flinch for a femtosecond !
  
 I interrupted "chopping to sections" my binaural recording of Bach*s Johannes Passion to partially answer your reply - and one single soprano is dominating the whole bunch of other soloists, orchestra and chorus - her peaks float some 10 dB above what male soloists could muster. On commercially available recording using multimiking etc, such a difference in level is usually "mastered out" - along with realism... NO, THANK YOU !
  
 I do not try to get the things out of context - and most certainly do mind my hearing. I found I was capable of reducing the monitoring level slightly BELOW what is being heard live - and that is usually much farther from the musicians and conductor position. Therefore I am exposed to less SPLs tha say a conductor. And conductors , by default, have to keep their hearing acute - up to a VERY ripe age.
  
 But, it *can* be a *bit* embarrasing to have on the floor equipment worth 100K+ - which can not reproduce even an uncompressed recording of a grand piano, let alone a full symphonic orchestra. 
  
 I do realize that not all the cars on the streets can have the performance similar to Ferraris - and for the same reason, similar goes for recordings. Yet, as this IS head-fi, putting together a headphone system capable of say clean 120dB SPL above say 30 Hz  with reasonable isolation ( to allow for dynamic range of say 80 dB ) should today not cost an arm and a leg. And recordings that do such a headphone rig justice IMO have their place under the sun. Again, with users clearly aware how to use such a equipment not only to get maximum enjoyment from listening, but taking precautions regarding hearing safety first.
  
 No one is limitting cars - there are "any" cars on the street, in Germany there is at least one registered Porsche 917 on the streets ( ! ) (and to my knowledge, it is STILL in one piece and driving in regular traffic...) - so why limit the recordings, if all some of us want is a realistic representation of a live musical event ?
  
 The reason behind all the opposition are commercial interests for the things to keep the  status quo, not technical or scientific.  Science can be used for progress - or for keeping the progress as slow as possible.  
  
 For bussines, as low size file formats as still "usable" is the best. One can not charge proportionally per file size - because Hi Rez PCM and/or DSD would reach prohibitive prices in no time compared to MP3s & similar.


----------



## gregorio

analogsurviver said:


> Just a quick reply.
> 
> 1.  There is VERY little (acoustical )  music that exceeds the loudness of a standing ovation applause ( really big symphonic works, with percussion and organ etc as finale, like Mahler*s 2nd ) - and as far as I am aware, nobody yet did suffer pain, let alone an injury or permanent hearing damage from an applause heard from within the audience.
> 
> ...


*

  
 1. Of course, because the actual ("truthful") level of a standing ovation is much lower than the level you would have to replay the recording in order to hear the digital noise floor of 16bit. Did you actually read what you are responding to?
  
 2a. OK, that's nonsensical. You think record labels should make recordings for just the say 0.01% of consumers, that tiny number of extreme audiophiles? You do realise that record labels are commercial businesses and not audiophile charities?
 2b. We're not talking about squashing an orchestra, we're talking about reducing transient peaks which, as you say, last for tiny fractions of a second and it's inaudible if they are reduced somewhat. Why is reducing these transients a mastering practice? To make the music sound realistic!! There's nothing "realistic" about not being able to hear quiet sections of the music, so what you're saying is the absolute reverse of reality. And, even if we don't reduce those transients, 16bit still has more than 10 times the dynamic range required. So what you're talking about is the aims of mastering, NOT any supposed limitations of 16bit!
 2c. Replay gain has nothing whatsoever to do with replaying at the level intended by the artist, so this statement is also nonsense!
  
 3. No it's not, that's nonsense. Firstly, you now appear to be confusing transient peaks with musical dynamic peaks. Secondly, not even a student mastering engineer would "master out" a 10dB musical peak in classical music, let alone a practising professional!
  
 4. Unfortunately though you've failed, as you are taking it out of context. Maximum musical dynamic at the conductor's position is roughly up to about 110dB or so. A concert hall has a noise floor of probably at least 50dB. The dynamic range is therefore about 60dB, that's about 36dB less than what 16bit is capable of!! If you're talking about a "truthful" recording then it HAS to include the noise floor of the performance venue, which is way, way higher than the digital noise floor.
  
 5. That depends. Many cinema sound systems cost $100k or more and can only just manage a dynamic range of 60dB.
  
 6a. Maybe not an arm and a leg, just a pair of ears!
 6b. I realise that extreme audiophiles have either no concern for the realities and limitations of human hearing or simply no understanding of them but it would be incompetent and negligent for a record label to release a recording where to hear the quiet sections means replay the recording at dangerous levels.
 6c. The instructions would be simple: If you want to continue to hear normal conversations, NEVER set your headphones to peak level (0dBFS) equal to 120dBSPL!!
  
 7. For hundreds of years, since the very dawn of modern science, science has had absolutely zero affect on the limitations of human hearing. 16bit is both beyond the ability of technology to fully reproduce but more importantly, beyond the limitations of human hearing.
  
 Please, no more nonsense and especially no more nonsense which advocates injury!!!
  
 G
*


----------



## goodvibes

The idea that you can't hear below a noise floor is a poor one. The possible advantage of HiDef is more about timing than acoustic bandwidth or measured noise floor and probably why I prefer it. Remember thresholds of what is considered audible was done with listening tests of single tones before solid state. The same ones the stated 1db is the threshold of difference when we know 1/2 db used as EQ in something good is repeatably distinguishable. I've done it. What happens in music is much more demanding and complex and it's the perception of that complexity that makes our hearing sense quite special. 
  
 Here's an interesting take on it: http://slidegur.com/doc/180896/hugo-dac-technical-master-class Consider that if in complex material as opposed to simple measurements, not everything times perfectly when your ears are actually this sensitive to time, which they are. It would give the sound a flatter, hashier perspective. Scratch your fingers together in front of the bridge of your nose. Now move it a fraction of an inch one way or the other. Very easy to hear.


----------



## spruce music

goodvibes said:


> The idea that you can't hear below a noise floor is a poor one. The possible advantage of HiDef is more about timing than acoustic bandwidth or measured noise floor and probably why I prefer it. Remember thresholds of what is considered audible was done with listening tests of single tones before solid state. The same ones the stated 1db is the threshold of difference when we know 1/2 db used as EQ in something good is repeatably distinguishable. I've done it. What happens in music is much more demanding and complex and it's the perception of that complexity that makes our hearing sense quite special.
> 
> Here's an interesting take on it: http://slidegur.com/doc/180896/hugo-dac-technical-master-class Consider that if in complex material as opposed to simple measurements, not everything times perfectly when your ears are actually this sensitive to time, which they are. It would give the sound a flatter, hashier perspective. Scratch your fingers together in front of the bridge of your nose. Now move it a fraction of an inch one way or the other. Very easy to hear.


 

 Oh NO! The garbage ideas that won't die.  Timing of undithered 16 bit redbook.  Accurate to 56 picoseconds.  Less than that with proper dither. So 4 microseconds is no sweat.  Other garbage myths that won't die.  Transients.  Transients perceptibility are determined by frequency response.  For the human ear steady state, complex signal any kind of signal it does not extend to the point you need extremely high sample rates.  The stair step output sample.  So sorry, a myth.  Looking at 20 khz from a CD will show as smooth sine wave, no stair steps. Increased sample rate 20 khz sine waves....same smoothness.  No increased smoothness or accuracy from the higher sample rates.  Sorry, you were taken in by advertising hype and believed it.


----------



## watchnerd

goodvibes said:


> The possible advantage of HiDef is more about timing


 
  
 Huh?
  
 Please explain this in terms of Nyquist theorem.
  
 Because it makes no sense at first glance how increased bit depth or higher sampling rates affect timing, which is a clock function and where errors lead to jitter.


----------



## StanD

stand said:


> But the latest fad is R2R DACs and they're only available in 16 bits. That means 24 bit DACs are out of fashion.


 
  
  


watchnerd said:


> While I'm not a believer in the alleged inherent superiority of R2R, that's not actually factually correct.  For example, the AD5791 used in the Schiit Yggy is a 20 bit chip (+1 LSB).


 

 I checked the spec sheet, only the 14 LSBs are R2R and the 6 MSBs are decoded using 63 matched resistors and 63 switches. Weird,


----------



## watchnerd

stand said:


> I checked the spec sheet, only the 14 LSBs are R2R and the 6 MSBs are decoded using 63 matched resistors and 63 switches. Weird,


 
  
 Wow...that is weird...I've never heard of that, nor sure why one would build it that way...


----------



## StanD

watchnerd said:


> Wow...that is weird...I've never heard of that, nor sure why one would build it that way...


 
 Just imagine how many resistors and switches would be required if the entire DAC used the same method as the upper MSBs? The chip would be the size of a car.  I wonder what the R2R worshipers would think of this? We could start a new fad.


----------



## watchnerd

stand said:


> Just imagine how many resistors and switches would be required if the entire DAC used the same method as the upper MSBs? The chip would be the size of a car.  I wonder what the R2R worshipers would think of this? We could start a new fad.


 
  
 Of course, forgot about the physical size issues, and that they're not transistors.
  
 But I think you're hit on something with the car-DAC.


----------



## gregorio

goodvibes said:


> [1] The idea that you can't hear below a noise floor is a poor one.
> [2] The possible advantage of HiDef is more about timing than acoustic bandwidth or measured noise floor and probably why I prefer it.
> [3] Remember thresholds of what is considered audible was done with listening tests of single tones before solid state.
> [4] Here's an interesting take on it: http://slidegur.com/doc/180896/hugo-dac-technical-master-class


 
  
 1. I never said one can't hear below a noise floor! In certain circumstances, say a pure tone in the critical hearing band and a digital noise floor which has been "shaped" away from the critical hearing band, then it is possible to hear many dB, even many tens of dB below the noise floor. However, you've unfortunately missed the point because I wasn't talking about tones or shaped noise, I was talking about white (or close to white) noise. The close to white noise floor of the recording, the close to white noise floor of the listening environment and the white digital noise floor of 16bit. It is impossible to distinguish/hear a white noise floor beneath another white noise floor which is many times higher in level. This is both common sense and something you can easily test yourself, so just saying it's "a poor idea" is not even close to being enough here in the science forum. If you're going to make such an extraordinary claim, you're going to have to provide some fairly extraordinary evidence, otherwise, you're just going to come across as someone who doesn't even have a grasp of basic common sense!
  
 2. What makes you think HiDef has any more timing accuracy than 16bit or that human awareness/perception of timing accuracy exceeds the accuracy achieved by 16bit? Is it just that marketing material to which you linked and presumably some other marketing material which suggests roughly the same thing? If so, the obvious implication of your statement is that you prefer Hidef because of marketing material! Just to put what Spruce Music stated into context, in case you didn't understand it: The 4us threshold (used in the marketing material) = 4,000ns = 4,000,000ps. At 56ps, the humble CD is therefore capable of a level of timing accuracy which is roughly 70,000 times below the threshold of audibility quoted in the marketing material! This fact obviously contradicts their marketing aim, so they hide it by not mentioning the timing accuracy of CD and try to hoodwink gullible audiophiles by confusing sampling period with timing accuracy.
  
 3. True but if you're going to mention this then you need to be honest and also "remember" that listening tests of audible thresholds did not just stop at the advent of solid state but continued up to the present day and that these tests did not only use single tones, they also tested with music and with signals even more complex than music, noise for example.
  
 4. Interesting, yes, but only from the point of view of seeing just how far audiophile marketing material is willing to pervert science in order to con gullible audiophiles! The solution to this common problem is quite simple, don't be so gullible! The majority of the time being less gullible really isn't very hard to achieve. In this particular case, just a modicum of common sense and a few basic facts would quickly expose the linked document as just marketing BS rather than the "technical master-class" it's pretending to be. For example, the first few pages go on about science's lack of understanding of hearing perception, that psychoacoustic models are therefore effectively useless and that the solution to all this is "lots of carefully designed listening tests" (p.5). One simple fact proves this is BS; psychoacoustic models are all actually based on "lots of carefully designed listening tests"!! And, science's tests are almost guaranteed to be more "carefully designed" than Chord's tests, plus science has definitely done "lots" more of them. The people at Chord apparently understand more than the whole rest of the world of science and have created chips which can apparently do what "no computer yet designed can do". If this were true, how come those people at Chord are not splashed all over the news being showered with Nobel prizes for advancing scientific understanding AND for advancing computer design? There are a number of other examples which don't need more than a bit of common sense and a basic fact or two to debunk, look at the two last points on page 12 for example. And with a bit more knowledge, most of the rest of the document is also easily recognisable as nothing more than marketing BS. The document should be titled: Hugo-DAC-Marketing-Master-Class! Although to be honest, I've seen better marketing BS than this example.
  
 While I can see how some of the more technical BS could have slipped past you, how did you manage to miss the far more obvious BS which only needed a bit of common sense to recognise?
  
 G


----------



## goodvibes

spruce music said:


> Oh NO! The garbage ideas that won't die.  Timing of undithered 16 bit redbook.  Accurate to 56 picoseconds.  Less than that with proper dither. So 4 microseconds is no sweat.  Other garbage myths that won't die.  Transients.  Transients perceptibility are determined by frequency response.  For the human ear steady state, complex signal any kind of signal it does not extend to the point you need extremely high sample rates.  The stair step output sample.  So sorry, a myth.  Looking at 20 khz from a CD will show as smooth sinke wave, no stair steps. Increased sample rate 20 khz sine waves....same smoothness.  No increased smoothness or accuracy from the higher sample rates.  Sorry, you were taken in by advertising hype and believed it.


Could be but I absolutely hear it and passed blindtests convincingly. Perhaps I'm trying to find a reason but it's not a bad idea. Just adding to the discussion. Feel freeto be rude. I've nothing to prove. You guys can have your ball back and toss it around the circle. I also think EVERY USB dac is worthless for hearing significant differences. Fine for making the noise from a pic more palatable but thanks t's about it. You can now hate on me some more. Ciao.


----------



## analogsurviver

gregorio said:


> 1. Of course, because the actual ("truthful") level of a standing ovation is much lower than the level you would have to replay the recording in order to hear the digital noise floor of 16bit. Did you actually read what you are responding to?
> 
> 2a. OK, that's nonsensical. You think record labels should make recordings for just the say 0.01% of consumers, that tiny number of extreme audiophiles? You do realise that record labels are commercial businesses and not audiophile charities?
> 2b. We're not talking about squashing an orchestra, we're talking about reducing transient peaks which, as you say, last for tiny fractions of a second and it's inaudible if they are reduced somewhat. Why is reducing these transients a mastering practice? To make the music sound realistic!! There's nothing "realistic" about not being able to hear quiet sections of the music, so what you're saying is the absolute reverse of reality. And, even if we don't reduce those transients, 16bit still has more than 10 times the dynamic range required. So what you're talking about is the aims of mastering, NOT any supposed limitations of 16bit!
> ...


 
 Oh dear... do I REALLY have to go to a CD store, grab - say - 10 of the recently released classical recordings at random , endure the chore of ripping them - and then post the result on audition or whichever PCM editor ? I started because of  rejecting the SQ of CDs recorded around the turn of the millenium onwards - the newer they were, the less realistic they sounded ! 
  
 Now, TBH - how many of headphones , even those considered to be high end, CAN in fact play back an uncompressed recording of a piano - or even worse, live microphone feed - at realistic SPL ? Say the same SPL as measured in say 5th row of parter ? And then, how many loudspeaker systems can do the same ? 
  
 Because record companies are releasing what they are, the percentage of equipment actually capable of doing it may well be in the 0.01%ish  range mentioned above.  And that needs to change - no more massive bass reductions ( which is the very first thing to sacrifice ) - or, releasing two versions of the same recording; one truthful, another stated as being compressed/EQed so that lesser equipment can play it with satisfactory result.
  
 I agree regarding the noise floor of the venues ; the newer they are, the noisier they are. Lighting alone can be enough, in case where "climatic" devices add their drone in the LF, it can be unbearable for any serious recording. Actually, I found that recording with lower resolution in such troublesome environment actually produces better result - sometimes all the way to using MP3 as an original master medium ! But, I will go to any length to avoid to have to record in noisy environment - and then will use whatever maximum resolution system available.
  
 16 bit may or may not be enough for _*playback *_- but it is nowhere good enough for recording. I did record a few CDs using CD-R, which, according to its specs, was recording with 14 bits. And, I found that pushing the recording level as far as it would go without clipping was extremely desirable - because a single dB reduction of recording level had drastic reduction of SQ.  Now, it did give me the skill to work much in the same way using better recording devices, including DSD - where it keeps the level of the recorded sound ratio to ultrasonic noise of the DSD as high as possible. In PCM, by going to 24 dB, one can have headroom of more than 10 dB while still having better resolution than 16bit recording driven close to 0dBFS. RBCD also pushes under the carpet any ultrasonic noise of ADCs and DACs - everything above 22.1K gets chopped off - end of "problems". 
  
 However, I DID find out that converting DSD128 to PCM 192kHz/32bit floating point can bring an uncannyly low noise floor - which I have yet to hear on any other medium. Compared to that is RBCD - noisegenerator. I particularly remember one organ recording; during the pauses, the sheer sensation of the acoustics of the church was great on master DSD128, unbelievable on 192/32 - and, compared to the former two, poor on RBCD. 
  
 Now, RBCD may well sound fine if one never gets the exposure to the former two ... - or enough live music. 
  
 On purpose I did not mention response above 20 kHz as being mandatory for quality recording. Regardless whether we can or can not hear pure sine waves above certain frequency - we can "perceive" "somehow" these frequencies ( there were studies and more are needed to bring down the myth "20 kHz is enough" - for good ) - without them it just is not realistic enough. It may be timing, it may be I-do-not-know-what - but I know I regretted with all my heart the recently released harpsichord recording could not yet be mastered in DSD ( Pyramax DAW, it goes from 1 bit to 8 bit only very close to the cut, meaning >99% of the resulting master recording will still be native DSD ) but in 192/32 and then released on CD. It is still very good - but the true charm of the harpsichord that has output (on this recording ) up to approximately 45-50 kHz ( one has to convert DSD into 192/24 or 32, in order to be able to use spectrum analyzer up to half that frequency, 96 kHz ) ,  is diminished. Next time, I hope DSD DAW to be fully operational - so that true DSD downloads can be made available. 
  
 If you have read the above correctly, there was nowhere any mention of any loud sounds. True, organ can get loud - but harpsichord is a quiet instrument, even close up, even in full cry. Going above RBCD 16 bit 44.1kHz limitations brings MUCH more believable reproduction - of both.
  
 Yes, those > 20 kHz are WAY, WAY down in level - but they DO matter. Yes, I know it not only sounds, but IS expensive - but centuries of musical instrument making and development have produced what we have today and it is my firm belief that if the technology allows for it, it is not excusable not to use it to preserve as much of the original sound as possible.
  
 Back in 50s, they used whatewer best they could ( including optical tape ) hold their hands on; that is why we are still discovering, using ever more sophisticated vinyl playback devices, what is truly lurking in those grooves. In 50 years time, no one could possibly claim the same for RBCD . While for sheer reproduction 16 bits ( after all the work in the studio with greater bit depth ) may be enough,  44.1kHz sampling most definitely is not.


----------



## spruce music

goodvibes said:


> couldn be but I absolutely hear it and passed blindtests convincingly. Perhaps I'm trying to find a reason but it's not a bad idea. Just adding to the discussion. Feel freeto be rude. I've nothing to prove. You guys can have your ball back and toss it around the circle. I also think EVERY USB dac is worthless for hearing significant differences. You can now hate on me some more. Fine for making the noise from a pic more palatable but thanks t's about it. Ciao.


 

 What kind of blind tests?  You hear it (what?).  Do you mean a difference in timing? Sorry for being rude.  I have seen this misguided idea about the length of sample periods being the limit of timing with digital thousands of times.  It was wrong the first time and still is.  If you hear something you think is due to timing that is one thing.  I can tell you however, whatever it is you are hearing is not due to timing limitations of even CD quality levels. 
  
 Same goes for those stair step illustrations.  Amazing that one still gets used.  Amazing that people believe it.  If you have never watched it, watch this video.
 https://xiph.org/video/vid2.shtml
 Here using quality analog signal generators and analog oscopes and spectrum analyzers they show what an old cheap AD/DA converter does to high frequencies.  The result is they look just like the analog signal they put into it at the output. So watch it and learn.  Unless you think they faked the video, next time some marketing BS masquerading as a white paper or master class tells you about the uneveness of sine waves from low rez digital and shows you stairsteps, you will know they are lying to you.
  
 And yes, your complaint about USB being worthless, let me guess, just by the marketing around such, it is about noise from the PC leaking into and corrupting the DAC right?  Yeah, right.  You really need to get better sources for technical subjects.


----------



## gregorio

goodvibes said:


> [1] couldn be but I absolutely hear it and passed blindtests convincingly. [2] Perhaps I'm trying to find a reason but it's not a bad idea. Just adding to the discussion. [3] Feel freeto be rude. I've nothing to prove.


 
  
 1. Are you really saying your hearing is 70,000 times more acute than the article you posted says is the threshold of human hearing? Does that mean you are claiming not to be human or some sort of super-human? 
 2. What, are you saying it's not a bad idea to seriously tell the world that you're super-human?
 3. Of course you've got something to prove, this is the science forum. If you're going to make outrageous claims then you need to back them up (and not with marketing BS!!). You come here and are rude to this forum's members by posting BS marketing and making BS claims and then you get all b*tthurt when the forum's members are rude in return? Do you have the worldwide monopoly on rudeness, only you are allowed to be rude?
  


analogsurviver said:


> [1] Oh dear... do I REALLY have to go to a CD store, grab - say - 10 of the recently released classical recordings at random , endure the chore of ripping them - and then post the result on audition or whichever PCM editor ?
> 
> [2]16 bit may or may not be enough for _*playback *_- but it is nowhere good enough for recording. [2a] And, I found that pushing the recording level as far as it would go without clipping was extremely desirable - because a single dB reduction of recording level had drastic reduction of SQ.
> [3] Now, it did give me the skill to work much in the same way using better recording devices, including DSD - where it keeps the level of the recorded sound ratio to ultrasonic noise of the DSD as high as possible


 
  
 1. You could try but how would the result in a "pcm editor" support all your claims? There is in fact a much better alternative, STOP spouting your nonsense in the first place, it's getting to the point of trolling!
  
 2. Name any music recording mics which have a dynamic range greater than 96dB.
 2a. This is an example of not only being wrong but of being the exact opposite of correct. Making a recording as hot as possible does not improve SQ, it reduces it. That's one of the first lessons any recording student is taught.
  
 3. Another example of not just being wrong but being as far from correct as imaginable. DSD does the exact opposite of what you suggest, it has an absolutely terrible signal to ultrasonic noise ratio, due to the severe noise shaping which is essential to making DSD work. You would know this if you had even a newbie level of understanding of how DSD works!
  
 I could carry on explaining why the rest of the points in your post are utter nonsense but why bother? All you'll do is just is just ignore the explanations and post another whole bunch of nonsense. You obviously have not even read the OP, let alone understood any of it. If you had, you would realise there is no resolution difference between 16bit and 24bit in terms of SQ, only in terms of an inaudible noise floor. If you disagree then fine, present your argument WITH some corroborating evidence but DO NOT just keep posting your off-topic suppositions, especially as they're all utter nonsense anyway! What you are doing is trolling, SO STOP!
  
 G


----------



## goodvibes

spruce music said:


> What kind of blind tests?  You hear it (what?).  Do you mean a difference in timing? Sorry for being rude.  I have seen this misguided idea about the length of sample periods being the limit of timing with digital thousands of times.  It was wrong the first time and still is.  If you hear something you think is due to timing that is one thing.  I can tell you however, whatever it is you are hearing is not due to timing limitations of even CD quality levels.
> 
> Same goes for those stair step illustrations.  Amazing that one still gets used.  Amazing that people believe it.  If you have never watched it, watch this video.
> https://xiph.org/video/vid2.shtml
> ...


 
  You can disagree but your assumptions of my experience are completely erroneous. 
  
 Tried dozens of USB DACs including DAVE and DCS via asio or wasapi and Wavelab, Foobar, J River and some Hi-End memory buffer type players. Found Wavelab setup correctly sounds the best to me though not really friendly as a library player. None sounded as good via USB as they did with a Firewire interface of a Weiss INT 202 or Konnekt interface (either linear supplied)(Dave wouldn't interface with the Weiss for some reason but we didn't troubleshoot very long. Those don't sound as good as what I can get streaming from a selected dedicated server and steamer. I'm plenty experienced so chill. I work with an award winning recording engineer and constantly hear transfers of both analog and digital sources and have been at the venues during the process. Not our main thing of which includes some tech repair. Mostly simple mic'd 2 track acoustic recordings in real space. We are always trying to improve the PC interface to have a better presentation when editing. Quality or price of equipment isn't an issue. Difficult to prove audibility on the interwebs. Measurements are always lacking IMO. Never told me much about the sound of anything. That we can disagree on and perhaps it's the bridge you can never cross but in a science forum, I think it would be better to find out why something exists that to simple dismiss its existence because it cannot be explained to your satisfaction. That existence is based on experience as you stated and ours are obviously different. I can't change you mind without a demonstrations so perhaps we could agree to disagree but I suspect you can't accept that. I truly believe you are as misguided as you think I am.
  
 I know how this goes, you don't accept anything by anyone that doesn't want to play by your specific rules and continue to attack to get someone defending himself when it shouldn't be part of a discussion. I'm out. Have at it and remember that when you stick your head in the sand, those ears need a good cleaning.


----------



## analogsurviver

gregorio said:


> 1. Are you really saying your hearing is 70,000 times more acute than the article you posted says is the threshold of human hearing? Does that mean you are claiming not to be human or some sort of super-human?
> 2. What, are you saying it's not a bad idea to seriously tell the world that you're super-human?
> 3. Of course you've got something to prove, this is the science forum. If you're going to make outrageous claims then you need to back them up (and not with marketing BS!!). You come here and are rude to this forum's members by posting BS marketing and making BS claims and then you get all b*tthurt when the forum's members are rude in return? Do you have the worldwide monopoly on rudeness, only you are allowed to be rude?
> 
> ...


 
 I am a regular human being with particular bias on sound quality. 
  
 No, I do not have monopoly on anything - and would react as I did only if treated like I have been by the members on sound science thread(s). 
  
 1. PCM editor would go to show how compressed are commercial recordings ( classical, from premiunm brands, most of the time including audiophile labels ) compared to something that was allowed to be left intact. It would have been visible from across the room !
  
 2. I will have to look up for the exact model #, but there is a Neumann mike with noise low enough to be on the same order of magnitude as air molecules impigning on its membrane. It can not get better than that.  It is covered in the Neumann book ( Jubilee X0 years ? ) from approx 5 years or so ago. No, I do not have it or have seen it in flesh so far - but it does exist. This means it has a good chance of exceeding the 96 dB dynamic range. There might be others as well - but I am not familiar with them.
  
 2a. Not true in all cases. Certainly not in the one cited. I am aware if and when pushing the level is detrimental to SQ - and act accordingly. It usually has to do with less than optimal analogue stages; ADC and DAC work just fine, only to "highlight" the cost cutting measures of the analogue parts of the recording chain.
  
 3. I know VERY well how DSD works. By recording as hot as it goes the CONSTANT ULTRASONIC NOISE FLOOR is kept as low as possible - that much you should understand. Ultimately, it will take DSD256 - or even DSD512 - to allow for > 100 dB S/N up to at least 100 kHz ; with DSD64 and less so, DSD128, the ultrasonic noise can quickly become a problem if the recording level is low(er) than it could be. IIRC, you get 6 dB lower ultrasonic noise floor for each doubling of the sampling frequency, which also starts twice higher in frequency compared to half the sampling rate frequency. It is a tradeoff . Korg recorders I use (with TI ADCs ) are noise -wise decent to approx 50 kHz, then the noise starts going up - regardless if it is DSD or PCM mode(s). With faster DSD ( better processors ) , the need to push levels should get reduced. I am eyeing Mytek Brooklyn ADC ( a recorder ) at the moment; DSD256. But will only go for it if I get a spectral analysis of its actual performance up to at least 100 khz. Last resort is Mytek*s 30 days return policy - but I would prefer to know this important spec/fact in advance.
  
 I think that terminology used as well as the fact that English is not my native language can at least partly be blamed for so heated a debate. I guesss if we could get together in a GOOD listening room with GOOD equipment, you could hear for yourself what I am "trolling" about.  The trouble is that really good equipment is really scarce - one is almost forced to beg/borrow/steal/modify/design/whatever by him/herself. No amount of real or virtual ink spent can replace a few seconds of audio that goes beyond the "officially accepted standard to be good". I do not get positive feedback from musicians and listeners alike for nothing. And I have been honest enough to post a recording I was clearly not satisfied with - and stating how and why it happened. You have latched on this one as if it were my best recording. You can disagree with me, you can call me crazy - that is OK - but I AM sincere and most definitely NOT trolling.
  
 The same goes for the moderators - ANY, each and every one of them.


----------



## StanD

So what's next a new audio high speed optical link for audio? Is anyone claiming to be able to hear RF yet? Not TOS, that as slow as pudd'in.


----------



## castleofargh

goodvibes said:


> spruce music said:
> 
> 
> > Oh NO! The garbage ideas that won't die.  Timing of undithered 16 bit redbook.  Accurate to 56 picoseconds.  Less than that with proper dither. So 4 microseconds is no sweat.  Other garbage myths that won't die.  Transients.  Transients perceptibility are determined by frequency response.  For the human ear steady state, complex signal any kind of signal it does not extend to the point you need extremely high sample rates.  The stair step output sample.  So sorry, a myth.  Looking at 20 khz from a CD will show as smooth sinke wave, no stair steps. Increased sample rate 20 khz sine waves....same smoothness.  No increased smoothness or accuracy from the higher sample rates.  Sorry, you were taken in by advertising hype and believed it.
> ...


 

 I'm not trying to blame you for getting a little mad, as you're answering a post that wasn't exactly gentle. so while I'd like it to stay civilized, because TOS, IMO it's 1-1 and all is good. but I just want to say that the "I've got nothing to prove" really doesn't belong to this subsection.
 we're discussing technical stuff and audibility, not ego and charisma.
 so instead of getting mad, if you really care about finding out why you're getting differences that others fail to notice, explain your setup,  and maybe your settings for the blind test.
 about USB, I'm limited to my own reading on the matter, but I've seen numerous times mentioned that async USB was usually the best option for timing. so unless all the stuff I've read was wrong, you might need to reconsider your argument about timing being the strong point of highres and what makes the differences audible. or stop trash talking usb, but both at the same time doesn't seem logical.
  
 edit:
 ps: @goodvibes weren't you the one saying that panning wasn't a problem for you on IEMs in the er4 topic when I mentioned that I couldn't stand listening to anything mastered for speakers without at least some matter of crossfeed? does that make sense to you to notice and favor devices and streaming solutions for stupidly small timing reasons, and favor highres while finding that there is no need to get a proper ITD and proper panning in general? did I get confused somewhere about the idea of fidelity?


----------



## gregorio

analogsurviver said:


> 1. PCM editor would go to show how compressed are commercial recordings ( classical, from premiunm brands, most of the time including audiophile labels ) compared to something that was allowed to be left intact. It would have been visible from across the room !
> 
> 2. I will have to look up for the exact model #, but there is a Neumann mike with noise low enough to be on the same order of magnitude as air molecules impigning on its membrane. It can not get better than that.  It is covered in the Neumann book ( Jubilee X0 years ? ) from approx 5 years or so ago. No, I do not have it or have seen it in flesh so far - but it does exist. This means it has a good chance of exceeding the 96 dB dynamic range. There might be others as well - but I am not familiar with them.
> 
> ...


 
 At least you're actually addressing the points now. Unfortunately we're still only half way there though, because you're addressing them with nonsense. 
	

	
	
		
		

		
			




  
 Let's get this out of the way quickly:
  
 1. Audio editors such as Audition cannot measure the amount of compression applied.
 2. So that's a "no" then, you don't know any mics which are capable of a dynamic range greater than 16bit. Why then, for a "truthful" recording, do you need more than 16bit when no mic can capture a "truthful" recording with greater than 16bit?
 2a. You act accordingly when you become aware that your levels are detrimental? By that time it's already too late! And, if it's due to say analogue stages and not ADC/DAC then it's obviously off-topic!
 3. Huh? It's a constant digital noise floor, it doesn't matter how hot your recording is, it won't affect the digital noise floor in any way, it's "CONSTANT"! The rest of your point is nonsense as well because at 100kHz there is nothing but noise! No music mic goes anywhere near 100kHz so even if there were some music content up there (which there isn't) it can't be recorded anyway.
  
 It really is quite impressive how you manage to pile nonsense on top of nonsense, apparently ad infinitum. It's quite a skill, even if I were trying deliberately, I don't think I could keep it up for so long and certainly not while maintaining the illusion that I was being serious. And of course it is an illusion right, it's all deliberate? I mean, it's inconceivable that you could be so absolutely wrong about virtually everything and actually believe it all. Even just by chance you'd have to at least occasionally come across some actual facts which you found believable. How is it even possible to ONLY believe nonsense? 
	

	
	
		
		

		
		
	


	



  
 G


----------



## goodvibes

Just doesn't seem to be an issue for me and the recordings I refer to are mostly coincidence so don't need mono type panning which we don't find as natural. I still seem to get good lateral positioning with good IEMs. I fell like crossfeed just mucks it up more for me than it helps. I get great localization from my jh13s but we are all bit different when it comes to in ears. I agree that I'd always rather hear it on speakers but I don't really think there's a great fix to your need so it comes down to preference and how your brain works the sound. We do have a natural sort of EQ/perspective engine at work where we acclimate. I suspect that certain aspects of that are more prominent in some than others. Psychoacoustics probably doesn't belong here either so I'l just say that for me, that issue falls into the no biggie range and what works for you is the way to go. 
	

	
	
		
		

		
		
	


	



  
 I don't know if it's timing but I do suspect that's a reasonable possibility. I do prefer the general tempo when I think things are right but I also know that's not the really same thing. May just be noise, tracking, clock or jitter. I really don't know but I do tend to prefer it when the clock is close to the dac and the supply is quiet and stiff.  Stupidly? Only if it's proven to not matter.
	

	
	
		
		

		
		
	


	




 If you're ever in Chicago. PM me and we'll spend an afternoon having some good spirited fun listening. 
  
 I'm really out this time. It genuinely seems I should stay away from this forum, LOL. The invite was sincere. I've got things like quad panels etc that I could pull apart for you and some great kit to hear.


----------



## analogsurviver

gregorio said:


> At least you're actually addressing the points now. Unfortunately we're still only half way there though, because you're addressing them with nonsense.
> 
> 
> 
> ...


 
 Your constant dismisal is equally impressive.
  
 How nowhere near 100 kHz goes a music mike you can check here :  
 http://www.sanken-mic.com/en/product/product.cfm/3.1000400
  
 Not quite so high go some of modified Bruel & Kjaer mikes, otherwise meant for measurements but found to be of low enough noise to be used for music.
  
 Up to "only" 50 kHz ( mightily flat .... ) go Eartworks* mikes
 http://www.earthworksaudio.com/microphones/qtc-series-2/qtc50/
 http://www.earthworksaudio.com/microphones/m-series/m50/
  
 I will dig up the EXACT Model/number of that low noise Neumann mike in a day or two - no idea where exactly is the book at the moment, other than it is at my place. Just because I did not instantly provide the model # of the Neumann mike, it does NOT mean it does not exist. Actually, I believe people will ask Neumann more for a "as low noise mike you have in programme" than for "XYZ-561NR ( or whatever might the true mike be called ). And, yes, at self noise around or below 15dB(A), maximum SPL at least 120 dB, that means it has at very least more than 96 dB dynamic range.
  
 OK, found something online - interpret it as you wish, a 7dB(A) microphone has dynamic range of 96 dB or more if the source has 103 dB  SPL or more. Therefore, at least one microphone that under entirely realistic conditions exceeds RBCD dynamic range :
 http://www.neumann.com/homestudio/en/what-is-self-noise-or-equivalent-noise-level
  
 Has it ever occured to you that somebody else might actually be right in his/hers claims - regardless how ludicrous they might appear to you at first ?


----------



## spruce music

goodvibes said:


> You can disagree but your assumptions of my experience are completely erroneous.
> 
> Tried dozens of USB DACs including DAVE and DCS via asio or wasapi and Wavelab, Foobar, J River and some Hi-End memory buffer type players. Found Wavelab setup correctly sounds the best to me though not really friendly as a library player. None sounded as good via USB as they did with a Firewire interface of a Weiss INT 202 or Konnekt interface (either linear supplied)(Dave wouldn't interface with the Weiss for some reason but we didn't troubleshoot very long. Those don't sound as good as what I can get streaming from a selected dedicated server and steamer. I'm plenty experienced so chill. I work with an award winning recording engineer and constantly hear transfers of both analog and digital sources and have been at the venues during the process. Not our main thing of which includes some tech repair. Mostly simple mic'd 2 track acoustic recordings in real space. We are always trying to improve the PC interface to have a better presentation when editing. Quality or price of equipment isn't an issue. Difficult to prove audibility on the interwebs. Measurements are always lacking IMO. Never told me much about the sound of anything. That we can disagree on and perhaps it's the bridge you can never cross but in a science forum, I think it would be better to find out why something exists that to simple dismiss its existence because it cannot be explained to your satisfaction. That existence is based on experience as you stated and ours are obviously different. I can't change you mind without a demonstrations so perhaps we could agree to disagree but I suspect you can't accept that. I truly believe you are as misguided as you think I am.
> 
> I know how this goes, you don't accept anything by anyone that doesn't want to play by your specific rules and continue to attack to get someone defending himself when it shouldn't be part of a discussion. I'm out. Have at it and remember that when you stick your head in the sand, those ears need a good cleaning.


 

 This highlights why you are getting a poor reception here.  You tell me of all the high dollar gear you have tried.  All the hands on experience.  As if that amounts to credentials to your views. Well in this forum that is not much of a benefit in convincing people.  In this forum we prefer for some careful testing with unsighted listening, or some measurements of the pertinent signals or some explanation that fits with how things work electrically, perceptibly and psychologically.  You offer that if we drop by you could play your system for us and that is going to illustrate your views and perhaps convince us of your opinions about things.  Sorry, that is the conventional audiophile approach, and not one that holds much sway in the Sound Science forum.
  
 I listen to electrostat panel speakers and might describe the sound as fast, quick, catches transients.  Yet I would understand transient ability and actual physical speed is not the reason for what I am hearing. You listen to CD and say the timing sounds insufficient (without fleshing out why you say that) and want to proceed as if you have solid data that CD has insufficient timing. Sorry, we know that isn't the case.  Though you may be hearing something real the timing isn't the problem.  Jumping to other formats at higher sample rates from the influence of marketing and saying timing is better is not convincing in this forum either.  Yes we have different rules to play by which is the whole reason for this forum separate from the others.
  
 You say measurements are always lacking in your opinion and expect to get a great reception based upon such a premise in this sub-forum?  Really, your feelings are hurt and it is our fault?
  
 You say USB is lacking, yet asyncrhonous USB lets the clock sitting right at the DAC to do the clocking.  This is the lowest jitter, best timing possible for playback and you dislike it.  Maybe what you think is timing isn't timing at all.  Yes, measurements are for real and do show this to be the case.  If your ears say one thing and the timing of the DAC says another your ears aren't going to be believed as an accurate measurement of timing. 
  
 Sorry, but the problem isn't a lack of open mindedness on this forum, it is you insisting on an ears first primacy which according to science has been shown to be false.


----------



## goodvibes

One more because it was my error.
 I wasn't trying to convince anyone and just relaying my thoughts. The list of kit and credentials was a mistaken response to this.
  
 "You really need to get better sources for technical subjects."
  
 I now understand that you meant written sources as opposed to listening sources so my response was not to the point you intended. Apologies. Just trying to show I'm not a novice in reply to inexperience which was not your intent but really, I think anyone can hear these things when presented properly so feel free to ignore it and to all the agnostics here:
  
 Have a great holiday!


----------



## gregorio

analogsurviver said:


> [A] Your constant dismisal is equally impressive.
> 
> How nowhere near 100 kHz goes a music mike you can check here :
> http://www.sanken-mic.com/en/product/product.cfm/3.1000400
> ...


*

  
 A. Not really, anyone with a moderate knowledge/experience of digital audio and recording could do the same.
  
 Oh dear, we seem to be going backwards!
  
 1. You didn't address this at at.
 2. Addressed this point but unfortunately with nonsense! The TLM 103 has an SNR, quoted by Sennheiser themselves, as 76.5dB. That's roughly ten times less than the dynamic range of 16bit.
 2a. Not addressed.
 3. Not addressed.
 3a. You've not addressed my point that there's nothing at 100kHz except noise! You have found a single example of a music mic which extends to 100kHz, which means my statement that there are none was inaccurate. I should have said that almost no music recording mics go anywhere near 100kHz. However, until you have (successfully) addressed my salient points here (massive digital noise levels and virtually no music related level at 100kHz), then you're still spouting nonsense. Even then you'd still be spouting nonsense unless you can demonstrate some evidence that 100kHz is audible.
  
 B. If it sounds ludicrous, conflicts with accepted scientific knowledge and is backed up with nothing but anecdotes then "no",  it occurs to me that there is very little or no chance that somebody else "might actually be right". And, if someone does that repeatedly, across numerous "ludicrous" unsupported claims, then the chances of them being right reduces to around infinity!!
  
 G
*


----------



## spruce music

Well DNR specs on mics are self noise referenced to one Pascal ( 94 db). So given that mics can typically work to 130 to 155 db sound levels most are capable under some conditions of exceeding 96 db DNR.


----------



## analogsurviver

gregorio said:


> A. Not really, anyone with a moderate knowledge/experience of digital audio and recording could do the same.
> 
> Oh dear, we seem to be going backwards!
> 
> ...


 
  I am sure to have posted this before - but since you obviously have to have it delivered on the silver platter : http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
  
 Seriously, if you do not know what to listen for the (in)audibility of the response beyond 20 kHz, it can only mean:
  
 1. The time between exposure to live music is measured in weeks if not months
 2. The equipment you use is purposedly limited to approx 20 kHz - one single "filter" called RBCD is more than enough to ruin the realism, let alone the microphone, preamp, power amp, any ADCs and DACs in the chain,  headphones, speakers, etc, each and every one limited to approx 20 kHz,  in series. FYI - around 1980 Technics had ENTIRE audio chain ( except the microphone ) where each and every piece of equipment, including the phono cartridge, has been capable beyond 100 kHz.  Around that time, microphones with MHz ( yes, you read it right, megahertz  ) bandwidth have been under research and development - and still are. Sooner or later, someone will suceed at making it a viable device for music recording.
 3. Because of the above , you can not capture nor reproduce the sound beyond 20 kHz on your recordings - a perfect "excuse" a la "if it is not audible on MY masters, it does not matter in the slightest". 
  
 It takes one single instrument to demonstrate the requirement to reproduce way  beyond 20 kHz, and certainly WAY beyond RBCD.
  
 It is called the double bass; pizzicato, plucked and/or slap playing produce VERY healthy output at least to 50 kHz ( which overall recording chain I am using can clearly reproduce ) ; how much above 50 kHz the double bass is capable of playing, will have to wait till I get DSD256 rekorder and mics that go, at the very least, to 100 kHz. Both are now reality, it is only a question of time I will have the financial means available to get them. It also means doubling the storage, computer performance, etc - in short, it is anything but inexpensive. But required if ultimately realistic recording is the goal.
  
 Earthworks, once upon a time, has been sending demo CDs featuring their microphones; the most effective demo ever has been recording of double bass, simoultaneously recorded by their mics flat to 25, 30, 40 and 50 kHz respectively. It was a CLEARLY audible difference - even on RBCD.
  
 Now think how much more this difference is audible if  the entire chain - from microphone to the final transducer ( headphone or speaker ) - supports the overall bandwidth at least 40 kHz. Then 100 kHz. The 40 kHz overall capable systems are quite realistic - since they have been around for decades.
  
 And, no, after hearing a good double bass playing live , then recording(s) of it in various "grades" ( from acceptable down to the RBCD ), played on good equipment that does support at least 40 kHz "flat overall", one can not say there is no difference. You would have to be deaf or dead - or both.
  
 For those insisting on A/Bing on computer no matter what: no, I have not been capable of stuffing a live double bass player into  the internet.


----------



## castleofargh

should we ask video recording and video playback systems to provide UV content because there were some in the scene that was filmed?


----------



## spruce music

analogsurviver said:


> I am sure to have posted this before - but since you obviously have to have it delivered on the silver platter : http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
> 
> Seriously, if you do not know what to listen for the (in)audibility of the response beyond 20 kHz, it can only mean:
> 
> ...


 

 Must be the holidays or something.
  
 You have heard it and it is so.  Audiophile bilge again.  Yes instruments put out some content above 20 khz.  You still can't hear it. 
  
 1 mhz microphones.  
	

	
	
		
		

		
		
	


	




 Some audiophiles will no doubt hear the difference.  Never mind that 1 mhz in air is absorbed at 160 db/meter.  I suppose if we develop 2 mhz PCM and 1 mhz capable speakers we can do away with the output filtering and just let the air do it for speakers.  Might not work for headphones.  So probably need 4 mhz response for that.
  
 http://www.ktu.lt/ultra/journal/pdf_50_1/50-2004-Vol.1_09-A.Vladisauskas.pdf  Here is info on air absorption of ultrasonics.
  
 To repeat, just because the sound is there at some frequency does not mean you can hear it.


----------



## gregorio

analogsurviver said:


> [A] but since you obviously have to have it delivered on the silver platter : http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
> 
> * Seriously, if you do not know what to listen for the (in)audibility of the response beyond 20 kHz, it can only mean: ...
> 
> ...


*

  
 A. It's not called "delivered on a silver platter", it's called supporting your statements and is required in science. Unfortunately though, you still haven't quite got the hang of it. The idea is to present evidence which supports your argument, not evidence which contradicts your argument and supports mine!! Oh dear.
  
 The paper you linked to shows that even those instruments which do produce content above 20kHz, produce exceedingly little. A trumpet for example, in the worst case scenario of being muted, produces 0.5 - 2% of it's power above 20kHz. How much do you think it's producing at 100kHz? According to the published frequency plot, far less again. The other tuned instruments tested produce significantly less power above 20kHz than a Harmon muted trumpet. The piano (which you seem fond of quoting) was just 0.02% above 20kHz! An entire symphony orchestra is therefore producing exceedingly little above 20kHz, let alone at 100kHz! The one exception with any potential significance was the crash cymbal, although it's percentage of power above 20kHz is still only 40% and a crash cymbal produces noise rather than defined tones/harmonics, noise which is close to while in nature, which brings us back to the inability to differential noise from noise. And that's the problem we have with SACD/DSD, massive amounts of unavoidable noise at around 100kHz, noise which is hundreds of times greater than what any of the instruments are producing (with the exception of the cymbal), a problem which you continually avoid addressing! To punch a couple more significant holes in the evidence you're attempting to use to support your argument: In an attempt to avoid room acoustics affecting the result, the measurements were effectively only of the transients (or just slightly longer than transients in the case of tuned instruments) and also the measurements were taken very close to the instrument, which obviously does not allow for the absorption of ultrasonic content by air, which would significantly lower the power output at 100kHz even further, at a typical listening position! Your "delivered on a silver platter" evidence is therefore more supportive of my argument than of yours and that's even if humans could hear 100kHz!
  
 B. It can only mean that I'm a Homo Sapien rather than a dog, bat, dolphin or alien!
  
 C. That's nonsense. All it demonstrates is that some playing techniques on string instruments produce transient noise above 20kHz. It does not demonstrate a "requirement", it does not even demonstrate that >20kHz transient noise is desirable!
  
 D. As a CD contains no frequencies above 22kHz, any audible difference MUST therefore be due to frequency content below 22kHz and have absolutely nothing to do with the mics' response above 25kHz!
  
 E. As the human ear does not contain and structures designed to respond to frequencies that high and this has been repeatedly confirmed by the fact that no one has ever demonstrated the ability to hear 40kHz, let alone 100kHz. The only logical answer is that there is no audible difference, let alone more of a difference!
  
 F. So what you're saying is that: 1. All the countless thousands of people who have been formally and informally tested over the last 90 years or so were all deaf, dead or both? B. As far as the evidence is concerned, you are very possibly the only human being on the planet who is not deaf, dead or both? or that C. You are actually a bat, dolphin or alien? I discount the possibility that you are a dog because even dogs can't hear that high! Even if you have a series of very significant genetic mutations which enable you to hear up to 100kHz, then still your response is nonsense because the maximum dynamic range on DSD/SACD is only around 6dB, due to the massive amounts of shaped dither noise. In other words, on a symphony orchestra SACD recording, pretty much the entire performance, with the exception of mainly just transients, would be below the digital noise floor! That must drive you nuts, how can you even bare to listen to SACDs?
  
 Again, you have failed to respond to ANY of the points, except for just one sub-point of point #3. Lastly, what you have responded to is off-topic any way! Frequency bandwidth is a function of the sampling rate, NOT the bit depth. Surely we must be approaching even the theoretical limits of nonsense by now? 
	

	
	
		
		

		
		
	


	



  
 G
*


----------



## StanD

Oh Lord, now we're venturing back into one of Analog's favorite topics, the OOB (Out Of Band) audio experience. He will always find another thread to invoke this theory so don't get too worked up over this.


----------



## goodvibes

Just to be clear, the idea in that link is that there is a difference between time and hearing out of band as stated. I couldn't less if an amp or DAC was filter rolled at 20k. In fact there can be advantages in being able to so with good phase character. Again, not claiming anything or that this is the mechanism why it's preferred, just looking for a reason. Not arguing here, just clarifying.


----------



## zareliman

Does anyone know a music recording that actually has more than 96dB of effective dynamic range ?


----------



## watchnerd

analogsurviver said:


> I am sure to have posted this before - but since you obviously have to have it delivered on the silver platter : http://www.cco.caltech.edu/~boyk/spectra/spectra.htm


 
  
 Yes, there are ultrasonic sounds in the world.
  
 But we don't design our audio systems to replay the dog whistle spectrum, because we can't hear them.
  
 Isn't it just easier to accept the truth than to find tortured rationales for an existing position?
  
 I honestly don't get it, unless you're just trolling.


----------



## StanD

goodvibes said:


> Just to be clear, the idea in that link is that there is a difference between time and hearing out of band as stated. I couldn't less if an amp or DAC was filter rolled at 20k. In fact there can be advantages in being able to so with good phase character. Again, not claiming anything or that this is the mechanism why it's preferred, just looking for a reason. Not arguing here, just clarifying.


 

 When you stated, "I do prefer the general tempo when I think things are right but..." what did you mean by tempo? Do you think that a DAC will change the tempo of music? Please clarify, thanks.


----------



## RRod

zareliman said:


> Does anyone know a music recording that actually has more than 96dB of effective dynamic range ?


 
  
 On a released album? Probably not. There was one track in the BAS experiment where a listener could hear the effect of the 16-bit loop at high volume, but their listening room was 19dBA. I have that disc: the RMS is about -34dBFS, and there are literally a handful of samples above -6dBFS.


----------



## goodvibes

stand said:


> When you stated, "I do prefer the general tempo when I think things are right but..." what did you mean by tempo? Do you think that a DAC will change the tempo of music? Please clarify, thanks.


 
 Didn't mean tempo in the technical sense. I doubt you guys like PRAT either. Happy holidays.


----------



## gregorio

zareliman said:


> Does anyone know a music recording that actually has more than 96dB of effective dynamic range ?


 
  
 On a commercial music recording the most I have seen is about 60dB. I've heard reports that there is at least one with just over 70dB. If you think about it, no one would release a commercial recording with more, you'd just annoy virtually all your customers. Many/most would be annoyed with a 60dB range, let alone one with say 30 times more!
  
 G


----------



## watchnerd

goodvibes said:


> Didn't mean tempo in the technical sense. I doubt you guys like PRAT either. Happy holidays.


 
  
 PRAT, cute Linn-marketing term aside, is basically a measure of transient response and system Q (ringing / overhang).
  
 The analog portion of a DAC could have some influence on this if it has a crappy power supply or crappy capacitors.


----------



## StanD

goodvibes said:


> Didn't mean tempo in the technical sense. I doubt you guys like PRAT either. Happy holidays.


 
 I don't think that the misuse of language by audiophiles is a good way to hide behind myth. PRAT is another good one, especially when someone says that it makes music faster. Merry XMAS to all.


----------



## goodvibes

Man, you guys are aggressive.


----------



## StanD

goodvibes said:


> Man, you guys are aggressive.


 

 All in the holiday spirit.


----------



## castleofargh

sadly we've been over all this a number of times(230+ pages of it just for this topic).
 when the recording used mics that didn't roll off too much in the ultrasounds, when the instruments had loud ultrasonic content, when nothing in the studio emitted loud ultrasounds, when no DSP whatsoever downconverted or low passed the signal, when the sound engineers decided to keep high energy ultrasounds as is even though they probably know it might F up some devices, when our playback device is ok with that, when we bought the expensive highres version and it was the same master, when our headphone can actually handle the transient response and the distortions aren't just as loud or louder in the ultrasounds as the actual music, when our ears can notice all that stuff wile listening to actual music, when the moon is aligned with mars, then the sound is somehow objectively "richer" in content compared to redbook and maybe for a small minority of people it makes a small audible difference.
 "takes a deep breath in french"
  
 that how far I'm ready to go to keep an open mind about this. a possibility under ideal conditions for some people. if it was more than that, I assume we would have seen loads of sponsored studies demonstrating it by now. because it's been an eternal concern for audiophiles and because the industry has wet dreams about having clear proof of the audible superiority of highres and all the money they would make if they could advertised it without having to constantly play hide and seek with the law about the terms they use.
  
 now mess with some parts of my requirement list and you can jump into no audible change at all, or into some changes that are actually making the highres file worst than the redbook one. so for all I know, several of the people saying there is a night and day difference are telling the truth about what they hear, but may be wrong about which is the best signal getting to their ears. how many people went to control that? if I count only with my fingers I expect it will be enough for headfi.
 the other element is to find out if just upsampling the redbook version to avoid the 44khz filter of the DAC may or may not create an audible difference. if it does, then there is a need to check correctly that it's not all the difference heard with highres music. to make sure that we're paying for content and not for just a sampling number.
  
 making claims about what we heard requires not only to demonstrate that we really heard things and not just "saw" them, but also the need to confirm the actual origin of the change before assuming that it's the highres content that made it all better. failure to provide all that is to me failure to demonstrate the hypothesis "music with it's original ultrasound content sounds better". and I'll then take the usual skeptical stand on the matter. not proved 100% impossibru, but certainly not proved to be true. and at this point I care about 2 things, how I feel for my music, and how blind tests in general answer to that question. personally I really don't care if something is highres or not and fail to identify anything in abx using the same format, suggesting that the ultrasonic content doesn't means shiit to me. I've had better luck upsampling redbook on some devices, so to me the low pass filter of the DAC is way more important than the ultrasonic content of the file. in general, most trials show guessing stats or super super close to it, even for the sponsored researches with an agenda to show ultrasounds are important.
 until I see enough evidence of the contrary, I'll stick to that and let marketing talk to my hand.
  
  
  
  
  
 off topic:
 I do have a lot of biases against PRAT. I know that's really not "objective" of me, but I usually move on to something else anytime I see a review using that term. and I think to myself "the brain jury will disregard previous readings".


----------



## goodvibes

Things are either inconceivable or conceivable. There's no kind of about it. There's a reason I didn't use P.R.A.T here but that sure didn't stop you guys from continuing to act out when fed.


----------



## StanD

goodvibes said:


> and why I didn't use it to begin with but without some measurement based term and not subjective, you guys wont accept it. That part is fine as it's your ball and game here but try being adults.
> 
> I treat posting as if I were sitting across from others. I don't know if some of you lack social skills or are simply compensating. I gave you prat and even said why I wouldn't use it to see what you'd do. You did not disappoint.


 
 The feeling might be mutual as we face the ranting of forum flash mobs going on about unprovable fairy tales with the gusto of a lynch mob. Yes at times they use R rated language or hateful phrases. You might try minding your own social skills, as to many of us you are visiting the last bastion of audio reality.


----------



## goodvibes

I recanted some of that as I'm letting it go. It's only personal when directed at someone and made persona. Someone else's reality shouldn't threaten your own. I just don't think we are aware of all that goes on under dynamic conditions as opposed to tests. You and others do and that's great but the attacks are a bit much.


----------



## castleofargh

stand said:


> The feeling might be mutual as we face the ranting of forum flash mobs going on about unprovable fairy tales with the gusto of a lynch mob. Yes at times they use R rated language or hateful phrases. You might try minding your own social skills, as to many of us you are visiting the last bastion of audio reality.


 
 that's really not a good reason. aren't objectivists supposed, and I really mean supposed, to be the better men with the more rational brains?
  
 TOS clearly says no personal attacks, it's really hard to argue without any sense of personal argument, but please try to make some efforts. you guys know all too well how it ends.
 attack the claims not the people!


----------



## gregorio

goodvibes said:


> [1] Someone else's reality shouldn't threaten your own.
> 
> [2] I treat posting as if I were sitting across from others. I don't know if some of you lack social skills or are simply compensating.
> 
> [3] without some measurement based term and not subjective, you guys wont accept it. [3a] That part is fine as it's your ball and game here but try being adults.


 
  
 1. You seem to have missed the whole point of why science exists! Science SPECIFICALLY attempts to avoid any individual's notion/s of reality and get to the actual facts.
  
 2. Well there's your problem! "Social skills" are defined by the society and as science exists to avoid superstition and get to the actual facts, pushing your own unsubstantiated idea of reality is effectively an attempted perversion of science and is therefore about as rude as it gets, as far as the scientific community/society is concerned!
  
 3. Subjective opinions are perfectly acceptable on this forum, provided they are NOT used as the sole basis for some stated fact. If this were not the case, it wouldn't be science and this couldn't honesty be called the sound science forum!
 3a. If by our "ball game" you mean science, then yes, it's our ball game and if you wish to participate YOU should try to be an adult, as an adult is expected to have at least a basic understanding of what science is, and therefore at least some idea of how to state facts without actually insulting science!!
  
 G


----------



## goodvibes

Science is observation, attempting to explain it by setting up a hypothesis and then trying to prove it with repeatable results. Even this can be flawed by interpretation of those results but that doesn't mean it's not science. That someone has done earlier portions of this without completing the process doesn't mean they are wrong or railing against science. It's often happened in science that an observation and hypothesis has been brought to the community and been laughed off as silly and later proven correct. Even Hubble doubted red shift though he was the one that collected the data. Einstein never accepted an expanding universe. Knowing the mechanism of a function is also not required or even gravity does not exist. As those that railed against past discovery, it's not that difficult to be blinded by science. I fully acknowledge that this does not give every crackpot idea credence but like most things observed by humans regardless of how critical and careful, we're not perfect; in our testing, our observations or our interpretations of same. I understand the need for blinders to strive for that perfection but it does narrow one's view.
  
 I began with my personal observation and gave an example of another's hypothesis. The idea was to discuss and look for other possible hypothesis. I wasn't trying to prove anything or attack anyone. The rest has been a waste of all our time. This can be discussed without resentment or derision or simply ignored unless you feel mockery is scientific. I suspect others, possibly very smart people, may have an idea or 2 but are hesitant to join in because of the type of responses I received so would rather not be bothered. We all suffer. If this were only about knowns, this forum need not exist and those pertinent facts could just be posted with simply question/answer pages applied. I guess science is not about asking questions or changing facts because, you know, that never happens in science.
  
 I promise I'll leave, never to return, once this has passed.


----------



## watchnerd

goodvibes said:


> If this were only about knowns, this forum need not exist and those pertinent facts could just be posted with simply question/answer pages applied.


 
  
 On the contrary:
  
 1. The "knowns" don't appear to be widely known or believed among placebophiles.
  
 2. Snake oil, lies, and vodoo are common amongst vendors as a way to separate the naive from their money
  
 3. Because of #2, much money and time is wasted on things that actually don't advance the state of the art at all in terms of genuine innovations
  
 This thread topic itself is evidence of all 3.


----------



## gregorio

goodvibes said:


> Science is observation, attempting to explain it by setting up a hypothesis and then trying to prove it with repeatable results.


 
  
 That's patently untrue, even more untrue when talking about audio equipment. Have you actually thought about what you wrote? You think someone observed a DAC in nature (say growing on a tree), then attempted to explain it with a hypothesis and then tried to prove the hypothesis with repeatable results? Of course not. In the case of digital audio, first came a hypothesis, then came a mathematical proof of that hypothesis (thereby turning it into a theorem) and then many years later engineers attempted to create a device which implemented that theorem. Personal observation of a DAC had nothing to do with it and any contrary hypothesis you care to come up with is invalid unless it can somehow disprove that which has already been proven.
  
 Even if we're talking about observation based science (rather than technology), after a hypothesis has been formed, experiments are designed to support that hypothesis and to be valid, those experiments have to be repeatable. At this stage, the hypothesis is still effectively worthless scientifically! The results then have to be compiled into a paper, peer reviewed and published in a respected scientific journal. Only then does it have any scientific validity! An observation, hypothesis and repeatable results prove nothing by themselves. Let's take an example and your definition of science: Let's say I observe god (maybe in a dream or hallucination), I then hypothesise that there is a god and, this hypothesis is obviously repeatable because many people in human history claim to have seen god. By your definition, are you therefore saying that the existence of god has been proven scientifically?
  
 Again, the purpose of science is to eliminate personal bias, personal ideas of reality and thereby differentiate science from superstition. You were the one who brought up "being an adult" and yet you don't appear to have an adult's basic understanding of what science is!
  
 Quote:


goodvibes said:


> I began with my personal observation and gave an example of another's hypothesis. The idea was to discuss and look for other possible hypothesis. I wasn't trying to prove anything or attack anyone. The rest has been a waste of all our time.


 
  
 Except for the last sentence, that's a lie! The offending post (#3435) began with "_The idea that you can't hear below a noise floor is a poor one._". Calling an "idea" a "poor one" is obviously an attack. Attacks on hypotheses or ideas are perfectly acceptable here and actually a requirement of science, providing, obviously, your attack/argument has some scientific validity. As explained above, say with a link to peer reviewed, published paper. We can be a little more forgiving here than in the formal world of science and allow links to some other publications/supporting evidence, providing it's a reputable source and doesn't obviously fly in the face of the accepted science. Your supporting evidence was marketing material, which is about as far from a reputable source as it's possible to get, and flew in the face of the known science numerous times! In fact, you supported your attack with a document which itself was an apparently deliberate perversion of the science! If one were trying to be an adult and had a basic understanding of science, how could one not see that as anything but insulting?
  
 G


----------



## StanD

I think we should ignore this and just move on, rather than keep flogging it over again and to no purpoose giving it unnecessary attention.


----------



## voxie

stand said:


> I think we should ignore this and just move on, rather than keep flogging it over again and to no purpoose giving it unnecessary attention.


 
 Thank God for that!!


----------



## watchnerd

stand said:


> I think we should ignore this and just move on, rather than keep flogging it over again and to no purpoose giving it unnecessary attention.


 
  
 So lock the thread with a final post saying?
  
 "Conclusion: 24bit audio, all else being equal, is of no audible benefit for consumer replay."


----------



## StanD

watchnerd said:


> So lock the thread with a final post saying?
> 
> "Conclusion: 24bit audio, all else being equal, is of no audible benefit for consumer replay."


 
 No. I suggested not engaging in a tennis match with a pointless exchange that is not much on topic and some might think borders on trolling.


----------



## StanD

So other than a digitally controlled analog attenuator, what do you guys think about software or digitally controlled attenuation that might be subject to reduced resolution and quantization? How might that play out in 24 bit vs 16 bit DACs?


----------



## watchnerd

stand said:


> So other than a digitally controlled analog attenuator, what do you guys think about softwaret or digitally controlled attenuation that might be subject to reduced resolution and quantization? How might that play out in 24 bit vs 16 bit DACs?


 
  
 Digital volume control is often done by up-sampling / SRC to a higher bit depth first.
  
 This is how Roon does it.
  
 But you can do this with normal Redbook 16bit source files.


----------



## StanD

watchnerd said:


> Digital volume control is often done by up-sampling / SRC to a higher bit depth first.
> 
> This is how Roon does it.
> 
> But you can do this with normal Redbook 16bit source files.


 

 Is this done before or after the DAC? Lets think of the consequences of 16 bit source material. If one up-samples, attenuates and then brings it back down to 16 bits (R2R) for the DAC what happens? Any other scenarios that might be questionable as to the results?


----------



## watchnerd

stand said:


> Is this done before or after the DAC? Lets think of the consequences of 16 bit source material. If one up-samples, attenuates and then brings it back down to 16 bits (R2R) for the DAC what happens? Any other scenarios that might be questionable as to the results?


 
  
 It's done before the DAC.
  
 In a modern DS DAC, you don't need to bring it "back down to 16 bits" -- you just go straight to conversion.
  
 As for R2R DACs, they do they same thing they do if they're fed higher bitrate content of any other kind.


----------



## goodvibes

gregorio said:


> Except for the last sentence, that's a lie! The offending post (#3435) began with "_The idea that you can't hear below a noise floor is a poor one._". Calling an "idea" a "poor one" is obviously an attack. Attacks on hypotheses or ideas are perfectly acceptable here and actually a requirement of science, providing, obviously, your attack/argument has some scientific validity. As explained above, say with a link to peer reviewed, published paper. We can be a little more forgiving here than in the formal world of science and allow links to some other publications/supporting evidence, providing it's a reputable source and doesn't obviously fly in the face of the accepted science. Your supporting evidence was marketing material, which is about as far from a reputable source as it's possible to get, and flew in the face of the known science numerous times! In fact, you supported your attack with a document which itself was an apparently deliberate perversion of the science! If one were trying to be an adult and had a basic understanding of science, how could one not see that as anything but insulting?
> 
> G


 
 LOL It's the concept and not a person I addressed. It came from the suggestion in a few posts that dynamic range is an absolute so I wasn't really reading in. It was not my intent to insult anyone. Since you said that you agree that we can hear into the noise floor, that ended. No one had specifically said you can't hear into the noise floor before that so I don't see how that was targeting anyone because it wasn't. I've been a bit less cordial in other posts since but there was no malice there. You're digging deep here. People in glass houses shouldn't throw stones.


----------



## StanD

watchnerd said:


> It's done before the DAC.
> 
> In a modern DS DAC, you don't need to bring it "back down to 16 bits" -- you just go straight to conversion.
> 
> As for R2R DACs, they do they same thing they do if they're fed higher bitrate content of any other kind.


 

 If you up-scale the numbers before using a 16 bit R2R DAC, you still have to scale it back down to 16 bits. once attenuated the entire DR is across a smaller set of numbers that are still quantized. So if the attenuation is such that the max value is 1024, then there are only 1024 steps of quantized resolution. Bring it down to 256, then there are even less steps of resolution. The question becomes, what is the audible effect? This is not the case in an analog world where depending on the design one might be battling the noise floor not quantization.


----------



## zareliman

gregorio said:


> On a commercial music recording the most I have seen is about 60dB. I've heard reports that there is at least one with just over 70dB. If you think about it, no one would release a commercial recording with more, you'd just annoy virtually all your customers. Many/most would be annoyed with a 60dB range, let alone one with say 30 times more!
> 
> G


 

 Then 14 bit was good enough at some point.


----------



## StanD

zareliman said:


> Then 14 bit was good enough at some point.


 

 Those that demand 32 bit, let alone 24 bit will be sorely disappointed.


----------



## zareliman

stand said:


> Those that demand 32 bit, let alone 24 bit will be sorely disappointed.


 

 It's reasonable to demand more bits if you plan to engineer, mix, DSP, edit or do something with the tracks. If you only want to listen to them, then they're just placebophiles.


----------



## StanD

zareliman said:


> It's reasonable to demand more bits if you plan to engineer, mix, DSP, edit or do something with the tracks. If you only want to listen to them, then they're just placebophiles.


 
 These forums are about listening. But that won't stop an audiophile from seeking more than necessary.


----------



## watchnerd

stand said:


> The question becomes, what is the audible effect?


 
  
 Ask the guys who make R2R DACs.


----------



## StanD

watchnerd said:


> Ask the guys who make R2R DACs.


 

 Are you looking for an answer from the marketing department?


----------



## Don Hills

stand said:


> If you up-scale the numbers before using a 16 bit R2R DAC, you still have to scale it back down to 16 bits. once attenuated the entire DR is across a smaller set of numbers that are still quantized. So if the attenuation is such that the max value is 1024, then there are only 1024 steps of quantized resolution. Bring it down to 256, then there are even less steps of resolution. The question becomes, what is the audible effect? This is not the case in an analog world where depending on the design one might be battling the noise floor not quantization.


 
  
 You first have to properly gain-stage your system. If you have to turn down the digital attenuation so you're only using 8 to 10 bits (256 to 1024 levels) at your usual listening level, your system is set up wrong. "Full digital volume" (no attenuation), on the music sources you usually listen to, should be enough to just drive your amplifier into clipping or at least be louder than you would want to listen to. If it's far too loud, you will need analogue attenuation between the DAC and amp. If it's not loud enough, you will need a preamp with gain. Once you have that established, you can use digital attenuation without problems. By the time you attenuate down to the levels you're talking about, it'll be so quiet you won't hear any audible problems.


----------



## castleofargh

watchnerd said:


> So lock the thread with a final post saying?
> 
> "Conclusion: 24bit audio, all else being equal, is of no audible benefit for consumer replay."


 
 but then you know it's exactly that kind of post that will alienate everybody who had some anecdotal experience of something that looked, at least to them, like a counter example.
 it's the "all dacs sound the same" and "cables don't change the sound" kind of stuff. we're reaching out to people who can't tell the difference between an anecdote and global conclusive evidence, or between a lemon and a standard, so we can't possibly give conditional truth without conditions. when you do, some people are almost sure to misinterpret what you're saying:
 -"he says it can't happen, but it happened to me that one time". conclusion, the guy is full of crap and objectivists are idiots.(too much? ^_^)
 is what many people will think in the end.
  
 I agree with with all those general statements myself as what should be expected, instead of as claims of how things will always be. they should be taught to all audiophiles. the audiophile starting pack of knowledge and expectations. so that when it's different, the audiophile will think that something is wrong instead of thinking he discovered iteration 284 of the one and only real sound(where is the facepalm emoji when we need one!).
  
 when I say those stuff, it's like when I say usb is 5V,or that a short headphone cable will always be less than 1ohm. none of my USB devices are exactly 5V and I actually have a few short cables that are 1ohm or a little more.  so the statements are accepted standards, and expected to be true(within some manufacturing margin), but they're not claims about all USB power sources and all cables under all conditions. we have to make the distinction clear enough for people of all levels of knowledge and thinking. what is true under nominal condition vs what is true always.
  
 so right now if you explain nothing else but 


> "Conclusion: 24bit audio, all else being equal, is of no audible benefit for consumer replay."


 
 in my head I go : "burden of proof lalalah".


----------



## StanD

don hills said:


> You first have to properly gain-stage your system. If you have to turn down the digital attenuation so you're only using 8 to 10 bits (256 to 1024 levels) at your usual listening level, your system is set up wrong. "Full digital volume" (no attenuation), on the music sources you usually listen to, should be enough to just drive your amplifier into clipping or at least be louder than you would want to listen to. If it's far too loud, you will need analogue attenuation between the DAC and amp. If it's not loud enough, you will need a preamp with gain. Once you have that established, you can use digital attenuation without problems. By the time you attenuate down to the levels you're talking about, it'll be so quiet you won't hear any audible problems.


 

 Phones and many devices change gain by software or digital calculation as a means of volume control. Is one's phone or laptop properly gain staged for normal listening at 100% volume?  I don't think so. Who listens at 100% volume? Now for the fun, where is the proof that one cannot hear any audible problems when turning down the digital volume?


----------



## spruce music

stand said:


> Phones and many devices change gain by software or digital calculation as a means of volume control. Is one's phone or laptop properly gain staged for normal listening at 100% volume?  I don't think so. Who listens at 100% volume? Now for the fun, where is the proof that one cannot hear any audible problems when turning down the digital volume?


 

 You have it backwards.  Where is the proof you can hear audible problems.  You can't prove the negative.
  
 As DH says, properly gain staged digital volume is no problem.  24 bit on the playback end may give some leeway on that. 
  
 One can gain stage poorly and have issues with analog volume too. At one time quite a few tube pre's were rather high gain and could put out high voltage levels like 20 or even 40 volts.  This was because it reduced distortion.  Used with tube power amps that would drive to near clipping with only .775 volts you had a bad combination.  Your pre had to be turned well down.


----------



## StanD

spruce music said:


> You have it backwards.  Where is the proof you can hear audible problems.  You can't prove the negative.
> 
> As DH says, properly gain staged digital volume is no problem.  24 bit on the playback end may give some leeway on that.
> 
> One can gain stage poorly and have issues with analog volume too. At one time quite a few tube pre's were rather high gain and could put out high voltage levels like 20 or even 40 volts.  This was because it reduced distortion.  Used with tube power amps that would drive to near clipping with only .775 volts you had a bad combination.  Your pre had to be turned well down.


 

 Turing down the volume on an analog device is not the same as scaling a discontinuous stream of numbers. So who plugs IEMs into their laptop and turns up the volume all the way? Someone with ringing in their ears?


----------



## gregorio

zareliman said:


> It's reasonable to demand more bits if you plan to engineer, mix, DSP, edit or do something with the tracks. If you only want to listen to them, then they're just placebophiles.


 
  
 In most of those cases a higher bit depth audio file format makes no difference. In the case of editing, bit depth makes no difference. In the case of mixing/DSP it doesn't matter either, because DAWs create a virtual mix environment (commonly 64bit float) and whether you load 16bit audio files or 24bit audio files into that mix environment makes no practical difference. Where a 16bit or 24bit audio file depth does make a difference is in recording, where the increased dynamic range of 24bit is effectively employed as increased headroom, which is very useful because we can't predict before we start recording what the peak level/s are going to be. It's for this reason that pretty much all pro recording is done at 24bit.
  

 Quote:


stand said:


> Phones and many devices change gain by software or digital calculation as a means of volume control. Is one's phone or laptop properly gain staged for normal listening at 100% volume?  I don't think so. Who listens at 100% volume? Now for the fun, where is the proof that one cannot hear any audible problems when turning down the digital volume?


 
  
 If we take your example of reducing the digital volume by 6bits (to 10bits) what we're left with is effectively 60dB of dynamic range. As hardly any commercial recordings exceed 60dB dynamic range then the 6bits you're loosing is nothing more than 6bits of noise floor or digital silence. In this example, the noise floor of a recording with 60dB dynamic range would be equal to the digital noise floor. In theory, the result of this would be an increase in the total noise floor of 3dB (summing together two equal level white noise sources). This of course should not be audible because you've reduced the volume of everything to start with by 36dB. However, this depends on your gain staging, as Don Hills tried to explain. Let's take a more extreme example for the sake of demonstration: Let's say you reduce your digital volume by 56dB (to 7bits) and then increase the amplification by 56dB. What we now have is a digital noise floor which is only 40dB below peak level, which in many cases would be easily audible. BTW, when I say "increase the amplification" I don't necessarily just mean whacking up the dial on your amp, very sensitive/easily driven cans or IEMs might effectively achieve a similar result. However, both of these cases (high amp or cans sensitivity) are examples of very poor, incorrect gain staging and is IMHO a common reason why audiophiles can sometimes apparently easily hear that which should be inaudible.
  
 In the case of a phone, where you can only reduce digital volume rather than the amount of subsequent amplification, we can't prove it's not audible because we still have the variable of the headphone/IEM sensitivity. All we can say is that a phone with a competently designed output stage should allow for a fairly wide range of models of cans/IEMs without causing any audible problems. If you're significantly reducing your digital level to achieve a very quiet listening level then there absolutely should not be any audible issues but if you're reducing it by say 30dB or more in order to achieve a normal listening level, then you've got a gain staging problem and are entering the realm of audible issues.
  
 G


----------



## spruce music

stand said:


> Turing down the volume on an analog device is not the same as scaling a discontinuous stream of numbers. So who plugs IEMs into their laptop and turns up the volume all the way? Someone with ringing in their ears?


 

 Never said they were the same thing.  I said poor gain staging can cause problems in the analog side as well.  I said reasonable gain staging will mean digital volume control is a non-issue.  These ideas are true.
  
 Since I know of no laptops with analog volume control what do most people interested in quality sound do with super-sensitive IEM's?  They don't plug them directly into the laptop.  Which means whatever they use instead can be chosen correctly for the purpose.  As most modern laptops have a 24 bit capable sound card it won't be an issue even with laptops. Laptops do usually have a higher noise floor in their outputs.  That isn't a digital volume control issue, rather simply a noise issue.


----------



## HAWX

@gregorio, I want to ask you something. Thanks for the enormous information you provide as a real person in industry. My curuosity is wouldn't be there always errors when capturing audio, because there are not numbers in real life, sound's exact starting and ending time, it's exact frequency and it's exact desibell?
  
 My question is what is the minimal decimal of the exact frequency in a lossless in 44.1khz file? Does 44.1 kHz file tells us about the 400.000038 th Hz or it tells like 400,401,402nd frequency with a 1 hz of increment an so on? And does higher sampling rates like 96 kHz sampling increase  the accuracy or does it increases just the range with same accuracy? And If it does not increase it how can we increase it, or can't we increase in this sampling method?
  
 And same thing about the desibell of that frequency, how can we increase minimally like 1db, 2db or 1.5432db? I don't think this will be infinitely small on a finite file. And does 24bit audio increases just the range of what we can capture or does it also increases the minimal increament 1.5db so that the accuracy?
  
  
 THANKS!


----------



## castleofargh

not Greg but when asking a question you have to consider your starting reference and a working model. if we're talking about anything in the world, then we don't know that there is any limit in accuracy aside from the limits imposed by our tools. quantum mechanic theories could suggest that nothing is perfectly analog and instead everything has some preferred quantified values.
  but obviously we really couldn't care less about that for music when people think that the real analog sound is a nail rubbing on a turning piece of wobbly vinyl.
  
 so instead of asking about perfection, I suggest you start looking at the resolution of recorded music. the conditions, the tools, the post processing done on most albums when mastered.
 let's say you start with 24bit, the room has noise, do you need to record the noises in the room to perfection? of course not. then the microphones have noise and generates some matter of distortions. after all, a mechanical device will not transmit 100% of the energy without any loss and some distortions from momentum are likely to occur. do you know what the most used microphones have as SNR for example? do you think we need a format that goes a lot beyond that other than for practical post processing purposes?
 then to record a singer, you can't possibly have the recording set full scale, because what if the singer goes a little louder that one time take? so if you record at 24bit, you'll start a few db below that already. well that's not really a problem as no ADC or DAC can do 24bit anyway.
 those are the conditions set for real life albums.
  
 then we move on to playback, the DAC, the amp, all have some noises and distortions. the headphones/speakers have a lot of distortions, it's really not rare to have some at 60db below the signal and way higher in the low frequencies(and I'm being conservative here). so at the end of the playback chain, what resolution do you really need? does it matter if all the noises and distortions down at -100db aren't perfectly reproduced?
  
 last but certainly not least, the human ear. for blind tests we count a 0.1db variation to be inaudible. that implies that a 0.1db error or less at full volume level will elude us. dunno for you but that sure made me think the first time I read about it.
  
 when you start to look at real world examples, 24bit resolution playback is ironically excessive for a human being without an integrated µSD slot. everything is relative and even if you really wish to increase the resolution of your music, the file format is not the place where you should be looking.


----------



## HAWX

Thanks for the info, and I know mechanical devices has severe limitations. But as physics do, I asked my question regarding every other element of the chain is perfect, otherwise I should start from else as you said. But I just asked for the numbers, If accuracy is not increasing with 24 bit audio, only range is increasing, than there is no need for 24 bit for playback, as Greg says in the beginning of the forum. I just want to know limitations of todays digital format, even it is less important than the tempereture and moisture of the air while recording.


----------



## gregorio

hawx said:


> My curuosity is wouldn't be there always errors when capturing audio, because there are not numbers in real life, sound's exact starting and ending time, it's exact frequency and it's exact desibell?


 
  
 Yes, there would be errors. The important question is, where do those errors occur? They mostly occur when we convert from one form of energy to another, for example: When converting sound pressure waves travelling through the air into electrical energy (by a microphone), when converting electrical energy back into sound waves (headphones/speakers) and lastly, when converting sound waves back into electrical energy again (the human ear). Digital audio doesn't directly capture audio (sound waves), what digital audio does is to measure the voltage and frequency of that electrical energy and store it as digital data, which can then be used to reproduce that electrical signal. Digital audio is many times more accurate than any of the transducers (those devices just mentioned which convert one form of energy into another) in the recording/reproduction chain and that includes the human ear! This is essentially the same as what castleofargh has said, just phrased differently.
  


> Originally Posted by *HAWX* /img/forum/go_quote.gif
> 
> And does higher sampling rates like 96 kHz sampling increase  the accuracy or does it increases just the range with same accuracy? And If it does not increase it how can we increase it, or can't we increase in this sampling method?


 
  
 In effect it just increases the range with the same accuracy. Increasing the accuracy is rather pointless, the accuracy is already significantly greater than the ear can discern and don't forget that you are effectively asking how to increase the accuracy of measuring and reproducing an electrical signal which is relatively very inaccurate to start with.
  


hawx said:


> And does 24bit audio increases just the range of what we can capture or does it also increases the minimal increament 1.5db so that the accuracy?


 
  
 Again, it effectively increases the range not the accuracy. The accuracy is already (with 16bit) tiny fractions of a decibel, far greater than that of the human ear. The original post of this thread explains all this.
  
 If you are asking how do we improve the accuracy of capturing and reproducing audio, the answer lies in finding improvements the transducers (mics, headphones and speakers primarily). However, transducers are very well developed as they've been around for a century or so, and there hasn't been any really significant improvements for many years, just incremental improvements. And, until we find a way to improve (or maybe bypass?) the final transducer in the chain (the human ear), there is nothing to be gained by any increase in digital resolution.
  
 G


----------



## HAWX

Thanks for the asnwer they just increase the range! Ok, I understand, but It really goes sideways, I don't ask how can we improve the overall quality, really I guess I'm not spesific enough.
  
 I donwloaded from tone generator and from 35.7 to 35.1 hz and I can and most will hear the diffrence I guess. But does it like that indeed, or does my equipment is rounding off to 35 and 36 repectively? When it comes to speaker? I know even speaker will not play exactly 35 but there is an audible diffrence I'm questioning.
  
 Do you have the numbers regarding how minimal we can increse the frequency and desbiell in a lossless file? Again, I'm not arguing increasing it would gain an audible advantage, but just asking it.
  
 Is this correct in terms of minmal dB (or level) increment for 16 bit?
 96 dB/65,000= 0.00147692307692.. dB (or level)?


----------



## castleofargh

hawx said:


> Thanks for the asnwer they just increase the range! Ok, I understand, but It really goes sideways, I don't ask how can we improve the overall quality, really I guess I'm not spesific enough.
> 
> I donwloaded from tone generator and from 35.7 to 35.1 hz and I can and most will hear the diffrence I guess. But does it like that indeed, or does my equipment is rounding off to 35 and 36 repectively? When it comes to speaker? I know even speaker will not play exactly 35 but there is an audible diffrence I'm questioning.
> 
> Do you have the numbers regarding how minimal we can increse the frequency and desbiell in a lossless file? Again, I'm not arguing increasing it would gain an audible advantage, but just asking it.


 

 some devices won't play exactly the perfect right pitch(I'm trying to be strict here, because pretty much nobody cares to even control that, it's a non issue in general). I remember all the fuss about the sansa clip having a pitch variation some years back, so it was playing everything a tiny bit too fast or too slow I don't remember ^_^. but the variations as you mention between 2 tones would still be correct even on those old sansa clips with the old firmware. the clocks used even in the crappiest device is many times faster than those frequency values and will always be able to keep 35.7hz and 35.1hz as 2 different frequencies with about that difference(if a device can do 20000hz, it knows how to 35.1 ^_^). what you want is simply a sample rate at least twice the value of the frequency you play. look up Nyquist theorem or those so very informative and now classic videos https://www.xiph.org/video/
  
 about the speaker, in fact the frequency is the one thing it will always get right. it may add a few others, but the one in the signal will be good. worst case, if it's moving too slow, it won't be able to reach the amplitude(loudness) it should, but the oscillations will follow the speed of the signal. drivers mess up a lot of stuff but not that.


----------



## spruce music

16/44 would differentiate 1000 hz from 1000.00000000056 hz. Add dither and you can add some zeroes. That is if you have a noise floor below 96 db and rarely will be the case with anything not an artificial signal. 24 bit add more zeroes.

As for levels, assuming 2 volt max output you can differentiate steps of 30 millionths of a volt. Again even smaller if you dither.


----------



## HAWX

Oh thank you! I'm refering to artificial signal yes. And about 24 bit I don't get it. Does it add more zeroz and increase presicion like 1000.000000000(+000)56 hz? (I know it isn't really important in real life.)


----------



## Danz03

Interesting how ppl are worried about digital frequencies shift, it's quite minute compare to analogue tapes or vinyls and yet no one ever complained about those.


----------



## spruce music

hawx said:


> Oh thank you! I'm refering to artificial signal yes. And about 24 bit I don't get it. Does it add more zeroz and increase presicion like 1000.000000000(+000)56 hz? (I know it isn't really important in real life.)




Yes 24 bit makes it more precise. Add two zeroes and end with 22.


----------



## HAWX

Yeah I am asking especially for some kind of coding but in the future. For playback may be the imperfections make better and natural experience, don't know


----------



## TheoS53

Loving this thread. It's always good to see something being objectively tested and discussed rather than relying on pseudo BS.

 For a while I stuck with 192/24 files simply because my player has the capability to play it, so I figured "why the hell not?"

 Recently, I decided to test out and see what the best compromise actually would be. To do this I used a track from a 192/24 album that I purchased on HDTracks and converted it to various sampling rates and formats. What I wanted to *test* was how accurate the file remained after the conversions compared to the original. So what I did was to convert, then import the original and the converted one into Audacity, and invert one of the tracks. If they were identical, there should be no sound, right?

 What I found was that converted to 96/24 there was a minor amount of noise added, but it genuinely could not be heard. I turned the amp up as well as turning the gain all the way up in Audacity, and still, it simply could not be heard. 48/24 added a bit more noise, but again, no matter what I did, it simply could not be heard. 

 However, when converted to 44.1/16, there was a noticeable hiss. Granted, I strongly doubt that you're gonna hear this when playing music...but the whole point of my test was to see what the best compromise would be to have as small a file, with essentially zero degradation of the file (i.e nothing lost, nor added within the audible range).

 Lossy formats were a total joke though....there was quite a bit of detail lost. So much so, in fact, that I could still make out the lyrics.


----------



## RRod

theos53 said:


> Lossy formats were a total joke though....there was quite a bit of detail lost. So much so, in fact, that I could still make out the lyrics.


 
  
 That's because the point of lossy formats is to throw out material that you can't hear when other material is playing around it. Null tests on lossy files will almost always have easily audible material, sometimes even sounding like a softer version of the track. But try to hear that difference underneath the lossy file itself: at any decent bit rate, you'll find it a hard task. A better null test to try would be to do a difference on two rates of the same lossy format.


----------



## TheoS53

rrod said:


> That's because the point of lossy formats is to throw out material that you can't hear when other material is playing around it. Null tests on lossy files will almost always have easily audible material, sometimes even sounding like a softer version of the track. But try to hear that difference underneath the lossy file itself: at any decent bit rate, you'll find it a hard task. A better null test to try would be to do a difference on two rates of the same lossy format.


 
 Agreed. I actually plan on making a video of my findings as well as different rates for the lossy files as you suggested.


----------



## tigen

Did you never think that some people might enjoy playing music for dogs, bats, dolphins, or aliens? I think you're all just being selfish.


----------



## watchnerd

tigen said:


> Did you never think that some people might enjoy playing music for dogs, bats, dolphins, or aliens? I think you're all just being selfish.


 
  
 Aliens don't have ears.


----------



## uchihaitachi

theos53 said:


> Agreed. I actually plan on making a video of my findings as well as different rates for the lossy files as you suggested.


 
It's worth bearing in mind that high fidelity tracks are not quite neutral either. At a practical level, it's worse and it's been proven in listening tests that distortion due to ultrasonic content can indeed be audible.
 
Distortion tends to increase sharply at the lowest and highest frequencies. Say if the transducer reproduces ultrasonics along with audible content, any nonlinearity will shift some of the ultrasonic content down into the audible range as an uncontrolled spray of intermodulation distortion products covering the entire audible spectrum. Nonlinearity in a power amplifier will produce the same effect. The effect is very slight, but listening tests have confirmed that both effects can be audible.
  
  
 You can however offset it

 A dedicated ultrasonic-only speaker, amplifier, and crossover stage to separate and independently reproduce the ultrasonics you can't hear, just so they don't mess up the sounds you can. (BUT WHYYYYY?!?!?!?!?!? 
	

	
	
		
		

		
			





)
 Amplifiers and transducers designed for wider frequency reproduction, so ultrasonics don't cause audible intermodulation. This will still be at the cost of some performance degradation in the audible portion of the spectrum.
 Speakers and amplifiers carefully designed not to reproduce ultrasonics.
 Not encoding such a wide frequency range to begin with. You can't and won't have ultrasonic intermodulation distortion in the audible band if there's no ultrasonic content.


----------



## castleofargh

watchnerd said:


> tigen said:
> 
> 
> > Did you never think that some people might enjoy playing music for dogs, bats, dolphins, or aliens? I think you're all just being selfish.
> ...


 

 please provide evidence to your claim.

 this guy doesn't seem to have ears, or at least not where humans have them.
 but the next one clearly has what could be interpreted as ears:


 maybe it's a male/female variation, but do aliens have different sex? I find all this highly inconclusive.


----------



## watchnerd

castleofargh said:


> please provide evidence to your claim.
> 
> this guy doesn't seem to have ears, or at least not where humans have them.
> but the next one clearly has what could be interpreted as ears:
> ...


 
  
 Real aliens are not anthropomorphic.


----------



## limpidglitch

tigen said:


> Did you never think that some people might enjoy playing music for dogs, bats, dolphins, or aliens? I think you're all just being shellfish.


 
  
 FIFY


----------



## spruce music

watchnerd said:


> Real aliens are not anthropomorphic.


 

 Maybe that is what they say about us.


----------



## watchnerd

spruce music said:


> Maybe that is what they say about us.


 
  
 I'll ask next time I talk to them.


----------



## headdict

watchnerd said:


> Aliens don't have ears.


 
  
  


watchnerd said:


> Real aliens are not anthropomorphic.


 
  
  


watchnerd said:


> I'll ask next time I talk to them.


 
  
 So I understand they are ear-less and not anthropomorphic beings. What exactly is involved in "talking" to them? How will their answer be transmitted?


----------



## watchnerd

headdict said:


> So I understand they are ear-less and not anthropomorphic beings. What exactly is involved in "talking" to them? How will their answer be transmitted?


 
  
 Probes.


----------



## headdict

And how do you know they are insensitive to high-frequency music?


----------



## bfreedma

headdict said:


> And how do you know they are insensitive to high-frequency music?


 
  
 Watchnerd has been probed.  Apparently, they didn't respond to his high pitched screaming...


----------



## watchnerd

bfreedma said:


> Watchnerd has been probed.  Apparently, they didn't respond to his high pitched screaming...


----------



## bfreedma

watchnerd said:


>


 
  
 I was thinking of the movie Independence Day, but I can work with that


----------



## StanD

watchnerd said:


> I'll ask next time I talk to them.


 

 Next time they abduct you don't forget to have a nanny cam at the ready while you're sleeping, the youtube will go viral.  I hope you have an Android Smartphone, all you'll need is an ultrasonic USB microphone and the below app.
https://play.google.com/store/apps/details?id=com.digitalbiology.audio&hl=en
 This will silence the naysayers.


----------



## karmazyn

Generally 24 bits give you more quality sound more clearly and more dynamique, 16 bit this only CD quality .is not studio quality.


----------



## dukefx

karmazyn said:


> Generally 24 bits give you more quality sound more clearly and more dynamique, 16 bit this only CD quality .is not studio quality.




For playback 10 bits are sufficient. You get 6 extra bits as headroom in case of CD quality. Now repeat after me: "I have no idea what I'm talking about". I should really open a studio and sell 32 bit recordings that are sourced from 16 bit ones and people will buy them and swear how much better those are. I'd be filthy rich!


----------



## NoteEater

I have read enough of this thread that I think it is time to throw something out there for the refuters to 24bit audio benefits.
  
 How many of you watch movies on a surround sound system?  Do you listen to to Dolby True HD or DTS Master Audio HD?  
  
 If you have, I say you should just go back to 16bit Dolby Digital or 16bit DTS.  Why bother with hi res 24bit Dolby True HD or DTS Master Audio HD if 24bit makes no difference at all?
  
 If you can think logically you will remember when you first heard Dolby True HD or DTS MAster Audio.  Most people recall the "HUGE" difference in sound staging, a higher perceived amount of spatial information, more dynamic range in the signal which in my opinion is just as important or more important than in music, and lastly just a simply better overall listening experience.  
  
 I have never heard of anyone going back to 16bit Dolby Digital or 16bit DTS after being able to reproduce better sound with a 24bit 48Khz/96khz Movie sound track. 
  
 Have you?


----------



## spruce music

noteeater said:


> I have read enough of this thread that I think it is time to throw something out there for the refuters to 24bit audio benefits.
> 
> How many of you watch movies on a surround sound system?  Do you listen to to Dolby True HD or DTS Master Audio HD?
> 
> ...



Dolby digital is compressed like mp3. So there is more than just a bit depth difference.


----------



## NoteEater

Dolby Digital can have a bit rate as high as 3mbits/s and as low as 500kbps.  All depends on how compressed they make it.  
  
 Dolby Digital Plus is even higher rates and I feel people will still feel there is a huge difference between Dolby Digital Plus and the 24bit Movie soundtracks.  Mostly due to dynamic range expansion going to 24bit depth and not just bitrate.
  
 If you listened to 16bit uncompressed PCM movie soundtracks against the Dolby True HD or DTS Master Audio version I would imagine you would draw the same conclusion of an overall improvement to sound quality.


----------



## dukefx

I listen to nothing but FLACs (44.1k/16), worst case scenario 320kb MP3 and I can't stand _Dolby whatever. _They have an artificial stench and quality wise they remind me of 128kb MP3s mixed with a pinch of "_now your ears bleed_" and a table spoon of "_now you can't understand a word_".


----------



## spruce music

noteeater said:


> Dolby Digital can have a bit rate as high as 3mbits/s and as low as 500kbps.  All depends on how compressed they make it.
> 
> Dolby Digital Plus is even higher rates and I feel people will still feel there is a huge difference between Dolby Digital Plus and the 24bit Movie soundtracks.  Mostly due to dynamic range expansion going to 24bit depth and not just bitrate.
> 
> If you listened to 16bit uncompressed PCM movie soundtracks against the Dolby True HD or DTS Master Audio version I would imagine you would draw the same conclusion of an overall improvement to sound quality.



You're still comparing apples to oranges. Other differences in those formats than just bit depth.


----------



## RRod

noteeater said:


> Dolby Digital can have a bit rate as high as 3mbits/s and as low as 500kbps.  All depends on how compressed they make it.
> 
> Dolby Digital Plus is even higher rates and I feel people will still feel there is a huge difference between Dolby Digital Plus and the 24bit Movie soundtracks.  Mostly due to dynamic range expansion going to 24bit depth and not just bitrate.
> 
> If you listened to 16bit uncompressed PCM movie soundtracks against the Dolby True HD or DTS Master Audio version I would imagine you would draw the same conclusion of an overall improvement to sound quality.


 
  
 The proper comparison would be to take the True HD version, truncate/dither/shape to 16 bits, then pad back to 24-bits and recode as True HD. In any case, the difference between purely truncated 16-bit vs. 24 are truncation errors that *peak* at -96dB, which means if you set your max peaks to 120dBSPL you are claiming to hear stuff that is at most 24dBSPL. Noise-shaped dither makes the perceived difference even quieter. So how quiet is your listening room?


----------



## NoteEater

Listen this thread has been going on since 2009 and it does not look like the argument will ever stop.  This will be the last time I am going to post here. It is just a waste of my energy.  Sorry nothing personal, just a topic that irks me every time I see another post or article about it.  I am going to keep doing what I am doing because I hear a difference on most recordings.  Now I understand we probably have a median age on this forum of probably 25 year sold and I get that most of the well recorded music is not your taste.  I am sorry that your era of music has compressed the living hell out of your favorite tunes sometimes giving you as little as 3db of dynamic range.
  
 If you have any faith in your fellow Head-Fiers you would agree that there are some very smart people on here that would not waste their hard drive space, their time, and their money if they saw no sonic benefits from 24bit.  Most who have properly experienced 24bit audio will try to listen to 24bit whenever they can because they know what they are missing if they don't.  These very smart people hear it.  I don't believe for a second that everyone that purchases SACD, DSD, or 24bit PCM are tricking themselves into buying something without sonic benefits.  This is not a small group of people in the audiophile community. It is a large portion of it which I am sorry to say but extends beyond listening to music on headphones.  
  
 I take this 16bit vs 24bit thing very personally.  I felt that over the last 18 years since 24bit audio was released to us that I was cheated out of a real HiFi revolution that was squashed by compressed dynamic mp3 files. America decided what it wanted and as usual chose to eat junk food again.  I really hope the naysayers don't ruin it for all of us again as they did by downloading free junk food in the early 2000s.  Our only hope is Hi Res audio is selling well right now and that might just be profitable enough to get new releases and re releases on a regular basis.  
  
 I have been listening to 24bit audio for 18 years now and have enlightened so many people.  This is commonly recited back to me, "I have never heard music sound like that before, and I have heard some really good sound systems." I will just keep working on people with open ears.  Tired of trying to convert those that listen to a one minute comparison track on mediocre hi res equipment that is probably not even set to play the 24bit file in anything but 16bit and then they say there is no difference.  
  
 If you came on a drive with me for an hour, I know you would be a convert. I have done it a hundred times. You just need to know how to set it up right and don't trust the recording you are listening to if you don't hear a difference.  Try a bunch of different 24bit recordings.  You will find bliss eventually. Hopefully


----------



## RRod

Seeing as how I'm looking at a box of about 200 SACDs, which adds to the over 1000 CDs of highly-dynamic, classical recordings extending back to the late 70s, I find your lack of faith… disturbing. Enjoy whatever you think you're hearing from 24-bit delivery.


----------



## watchnerd

noteeater said:


> Now I understand we probably have a median age on this forum of probably 25 year sold and I get that most of the well recorded music is not your taste.  I am sorry that your era of music has compressed the living hell out of your favorite tunes sometimes giving you as little as 3db of dynamic range.


 
  
 Not me (nor I suspect much of this sub-forum).  I'm in my 40s.  Most of my listening is jazz and classical, with ample dynamic range.
  
 I also work as a volunteer recording engineer for the local symphony and jazz venue.  Recent recordings I've done have about a 40dB dynamic range. Originally recorded in 24bit, distributed in 16bit.
  


noteeater said:


> If you have any faith in your fellow Head-Fiers you would agree that there are some very smart people on here that would not waste their hard drive space, their time, and their money if they saw no sonic benefits from 24bit.  Most who have properly experienced 24bit audio will try to listen to 24bit whenever they can because they know what they are missing if they don't.  These very smart people hear it.  I don't believe for a second that everyone that purchases SACD, DSD, or 24bit PCM are tricking themselves into buying something without sonic benefits.


 
  
 I believe exactly what you said: people are tricking themselves into buying 24bit PCM without sonic benefits.
  
 I do not waste hard drive space or money on 24bit recordings.  I have performed multiple ABX tests on Redbook vs high resolution.  
  
 It's simple logic:
  
 1. The extra dynamic range of 24bit is useful in production.  It's not useful in playback.  You will not find a commercial available recording of music that exceeds the 96 dB available with 16bit.  And, even more to the point, the 144 dB dynamic range is beyond the noise floors of playback equipment.
  
 2. You can't hear ultrasonics.  You don't need sample rates extending into dog hearing range.  And your playback system isn't designed for ultrasonics, either.
  
 As for the "very smart people"...well look at the AES papers where the subject matter experts (not just smart, but experts in the domain) have done multiple studies on the audibility of high resolution.
  


noteeater said:


> I take this 16bit vs 24bit thing very personally.  I felt that over the last 18 years since 24bit audio was released to us that I was cheated out of a real HiFi revolution that was squashed by compressed dynamic mp3 files. America decided what it wanted and as usual chose to eat junk food again.  I really hope the naysayers don't ruin it for all of us again as they did by downloading free junk food in the early 2000s.  Our only hope is Hi Res audio is selling well right now and that might just be profitable enough to get new releases and re releases on a regular basis.


 
   
 You're conflating high resolution with lack of compression.
  
 Apples vs oranges.  Two different issues.
  
 I can make a Redbook recording with high DR, or a highly compressed high-rez recording with low DR.  In fact, there are many on the market.
  
 Quote:


noteeater said:


> If you came on a drive with me for an hour, I know you would be a convert. I have done it a hundred times. You just need to know how to set it up right and don't trust the recording you are listening to if you don't hear a difference.  Try a bunch of different 24bit recordings.  You will find bliss eventually. Hopefully


 
  
 Anecdotal experience, with full human sighted biases, does not really meet the criteria for good empirical evidence.


----------



## spruce music

noteeater said:


> If you came on a drive with me for an hour, I know you would be a convert. I have done it a hundred times. You just need to know how to set it up right and don't trust the recording you are listening to if you don't hear a difference.  Try a bunch of different 24bit recordings.  You will find bliss eventually. Hopefully


 
 Is this really Neil Young?  Please don't tell me you can take me on a drive in your car for one hour and convert me.  The only thing that would convince me of firmly is you fooling yourself.  Of all places a car is not where you would hear a 24 bit vs 16 bit difference.  You lost credibility there friend. 
  
 Your ideas about those who don't agree don't match reality.  I too have done recording, and have heard 24 bit on excellent systems.  My experience doesn't match your description of the large difference whatsoever.


----------



## watchnerd

spruce music said:


> Is this really Neil Young?  Please don't tell me you can take me on a drive in your car for one hour and convert me.  The only thing that would convince me of firmly is you fooling yourself.  Of all places a car is not where you would hear a 24 bit vs 16 bit difference.  You lost credibility there friend.
> 
> Your ideas about those who don't agree don't match reality.  I too have done recording, and have heard 24 bit on excellent systems.  My experience doesn't match your description of the large difference whatsoever.


 
  
 I think you're right -- it is Neil Young!
  
 I should have caught on when piracy was mentioned...


----------



## NoteEater

You guys ever run Hertz Mille Speakers Active not passive off a 32 band capable Calibration with appropriate Time Delay personally by the Midwest Hertz Audison rep.  He used a very expensive Audison Bit Tune Auto Calibration System through an Audison 24 Audison Bit One processor also supporting a 10" Dual Voice Coil Sealed Hertz Audison subwoofer with 900 watts pumped through it? 
  
 Probably not huh? I have for 7 years now.  You have no idea what you are missing. One of the best sound spaces I have been in myself and hundreds of people who know good sound have sat in it for hours and agree. Don't speak of what you don't know, please. Talk of your own experience or call it your own theory.  
  
 I love Neil Young.  Show some respect boys.  The man has pushed the envelope of sound quality on more musicians than you can probably name.  Give it a rest already.  Who cares if he came out with an average DAP that blew up in his face because a company decided to sell uprezzed 16bit files. The guy has wrote more songs than ten post 2000 era musicians combined.  
  
 Just because new members on this forum have few posts, don't underestimate there experience and their anecdotal evidence.  After all we are listening actively right.  Not just writing about it on our computers?   I see live music all the time.  Is this not what we are trying to replicate.  At least trying to replicate the studio. Half of my collection is 16bit FLAC recording of live performances and just about as many 24bit FLAC live recordings. Modern recordings done right.  The 24bit recordings are the only recordings that put me back at the venue I was at often the night before.  Many agree with this anecdotal evidence.  Been doing it for years. Have you ever gone out and recorded a live music venue in DSD or at least 24bit 192khz and then listened to it later that evening?  Probably not huh?  Well I have many many times. I can replicate the show I just saw often better than most spots in the venue.  As if I have the sweet spot in the venue.
  
 If you take a DSD recording and convert it down to 24bit 96khz and play that back on a proper system, most people will say they have never heard any sound like that before from a sound system.  If you record the same performance in 24bit 192khz and convert that down to 16bit after compressing the hell out of the mix, there is no comparison in quality.  The way something is recorded and mastered makes all the difference in the world in whether you will hear a difference in 16bit vs 24bit.  I agree that I have heard a lot of bogus files in 24bit that should never been sold as 24bit.  They exist, in droves.  It is a shame and should be considered fraud in my opinion especially if you are profiting from it. Sorry.  Many naysayers here probably bought a bunch and are ticked off.  I would be too.  Sorry you wasted your money.
  
 I hate to break it to you guys, but with all this flac I am getting, I really I am starting to believe that headphones are not the proper venue for 24bit audio.  :-/ The waveforms cannot develop in a headphone the way it can in an open space.  Maybe a Smyth Realizer is what is needed.  Not sure. Have not heard one yet.  The point here is the way the sound fills a room is completely different when it is a proper 24bit recording.  There is a sense of not being able to place sound coming from the speaker itself and instead you just the room filled with music.  An extension of sound from the speaker cabinet is evident up to 3ft from he speaker in a proper room. Maybe even further in a large room with better equipment.  16bit Files have a tendency to sound veiled and ever so slightly extended from the speaker.  As if the sound is struggling to fill the air and I can tell I am in a room with speakers.   This is a sensory perception and not necessarily and auditory one, which I actually do have some study in and experience for 14 years.  Headphones cannot give you this experience.  At least I have not heard it yet and it is because the waveforms have specific distances they need to travel to completely form(just physics here really) and there is obviously nothing but SPL exhibited in an ear/headphone due to lack of space.  Sorry but its the truth.  Pretty darn sure. Right Tyll?
  
 I listen on my home system as much as I do on my headphones currently.  But mostly I do my serious listening in the car.  There is enough room in a vehicle to get a similar sense of music filling the entire vehicle as opposed to sensing sound from the left and right speaker.  When done properly the car can be one of the best sound spaces you will ever experience.  You can find quotes from Peter Gabriel talking about this very thing.  Another that deserves respect and certainly knows good sound experiencing it in person his whole life.  Created a whole Society around Sound if I am not mistaken.  
  
 By the way, Neil Youngs newly pressed vinyl releases over the past few years ripped to 24bit 96khz FLAC sound absolutely AMAZING!!!!


----------



## spruce music

noteeater said:


> You guys ever run Hertz Mille Speakers Active not passive off a 32 band capable Calibration with appropriate Time Delay personally by the Midwest Hertz Audison rep.  He used a very expensive Audison Bit Tune Auto Calibration System through an Audison 24 Audison Bit One processor also supporting a 10" Dual Voice Coil Sealed Hertz Audison subwoofer with 900 watts pumped through it?
> 
> Probably not huh? I have for 7 years now.  You have no idea what you are missing. One of the best sound spaces I have been in myself and hundreds of people who know good sound have sat in it for hours and agree. Don't speak of what you don't know, please. Talk of your own experience or call it your own theory.
> 
> ...


 

 When I play this back over my gear  in my boat at 32 bit (like 24 bit could compare...you must be joking), it sounds incredible.  Best sound you never heard.


----------



## dukefx

That terrible noise you linked has a dynamic range of about 40dB and I'm being generous here. That's about 7 bits. What do you need 32 bits for? Even 16 bits are twice as much as required.


----------



## spruce music

dukefx said:


> That terrible noise you linked has a dynamic range of about 40dB and I'm being generous here. That's about 7 bits. What do you need 32 bits for? Even 16 bits are twice as much as required.


 

 To make it sound good is what the bits are for mate.  Spend an hour in my boat and you'll get it clear as a bell or marker buoy. 
  
 Neil Young would get it too.  Here is a bit from a Fresh Aire interview with Graham Nash about Neil.
  
In the midst of the interview, Gross asked Nash to talk about his friendship with Neil Young, a man Nash has called “the strangest of my friends.” Just what makes him strange? Nash explains:
  


> The man is totally committed to the muse of music. And he’ll do anything for good music. And sometimes it’s very strange. I was at Neil’s ranch one day just south of San Francisco, and he has a beautiful lake with red-wing blackbirds. And he asked me if I wanted to hear his new album, “Harvest.” And I said sure, let’s go into the studio and listen.
> 
> Oh, no. That’s not what Neil had in mind. He said get into the rowboat.
> 
> ...


----------



## dukefx

Let me tell you the story of my very young and naive self who sometimes burned 128kb MP3s to audio CDs (back in the days there weren't any MP3 players) and realized how much better the audio CD sounded compared to the original source (yes, that crappy MP3). Back then I didn't understand the science behind it, but now I do and laugh at myself. Before you go into bits and sampling rates you need to understand human hearing.


----------



## spruce music

dukefx said:


> Let me tell you the story of my very young and naive self who sometimes burned 128kb MP3s to audio CDs (back in the days there weren't any MP3 players) and realized how much better the audio CD sounded compared to the original source (yes, that crappy MP3). Back then I didn't understand the science behind it, but now I do and laugh at myself. Before you go into bits and sampling rates you need to understand human hearing.


 

 Come on, try and keep up now.  Look at my posts in just this thread and it should be enough.  Look at the posts of NoteEater.  That should suffice.  TIC and all of that. I do believe you have taken me the wrong way round.  Parodies are complicated sometimes.


----------



## gregorio

noteeater said:


> You guys ever run Hertz Mille Speakers Active not passive off a 32 band capable Calibration with appropriate Time Delay personally by the Midwest Hertz Audison rep.  He used a very expensive Audison Bit Tune Auto Calibration System through an Audison 24 Audison Bit One processor also supporting a 10" Dual Voice Coil Sealed Hertz Audison subwoofer with 900 watts pumped through it?
> 
> Probably not huh? I have for 7 years now.  You have no idea what you are missing. One of the best sound spaces I have been in myself and hundreds of people who know good sound have sat in it for hours and agree. Don't speak of what you don't know, please. Talk of your own experience or call it your own theory.


 
  
 If you're going to play that card then OK, let's play:
 No, I've never heard the specific system/space you're talking about. On the other hand, have you ever worked in or even heard a commercial dubbing theatre? 900 watts is pathetic for a sub, the last system I worked with had over 20,000 watts of sub. The system was not calibrated by some sales rep but by Dolby techs and real experts and your idea of expensive is another pathetic joke, was it a $20,000,000+ purpose built film audio facility? Probably not huh? I have for about 20 years now, you have no idea what you're missing, please don't speak of what you don't know. Talk of your own lack of experience and call it your own theory!!!
  
 OK, now we got that nonsense out of the way, let's deal with some of your so called "facts": Dolby Digital is NOT 500kbps  - 3mbps! It's maximum is 768kbps but that is rarely used, for HDTV, DVD and BRD 448kbps is the DD standard, IE. It's highly compresse!. Dolby Digital is NOT 16bit and furthermore, it's a 5.1 format whereas TrueHD is 7.1, you are comparing apples and oranges, as stated by others. Your statements about localisation are also nonsense; yes, 7.1 is better than 5.1 for localisation but it still has the same basic issues, which is why it was replaced by formats such as Dolby Atmos. But none of this has anything to do with 16 vs 24bit. Is 24bit better than lossy compression? Of course but again, that's nothing to do with 16 vs 24bit!
  
 You having spoken about "converting" people is very troubling. Converted them to what? Converting them from ignorance to incorrect/false information is doing them a serious disservice and as a professional in the field, I'd appreciate if you'd STOP your phoney "conversions"! If you want to learn how it really works, then ask, we're happy to help but don't make-up factual statements you can't back up, which conflict with the science and with how it really works or about your experience of a high-end, so called expensive system which is actually a very cheap, low-end system!
  
 Quote:


rrod said:


> The proper comparison would be to take the True HD version, truncate/dither/shape to 16 bits, then pad back to 24-bits and recode as True HD. In any case, the difference between purely truncated 16-bit vs. 24 are truncation errors that *peak* at -96dB, which means if you set your max peaks to 120dBSPL you are claiming to hear stuff that is at most 24dBSPL. Noise-shaped dither makes the perceived difference even quieter. So how quiet is your listening room?


 
  
 We have to be careful here, film sound and music in effect are two very different things, they have very different workflows and distribution chains. We don't apply noise shaped dither in film/TV products, due to considerable amounts of additional processing being required downstream, after the print-master is completed. For this reason distribution is always 24bit or a proprietary lossy compressed format, to avoid any build up of dither, truncation or noise-shaping artefacts. We can't therefore use the same comparison logic as we can with music because there is no 16bit consumer content out there in the film world, let alone a dominant 16bit format in which the application of noise-shaped dither has been standard practise for commercial release for nigh on 20 years.
  
 G


----------



## Cerastes

noteeater said:


> I have been listening to 24bit audio for 18 years now and have enlightened so many people.  This is commonly recited back to me, "I have never heard music sound like that before, and I have heard some really good sound systems." I will just keep working on people with open ears.  Tired of trying to convert those that listen to a one minute comparison track on mediocre hi res equipment that is probably not even set to play the 24bit file in anything but 16bit and then they say there is no difference.



 


Maybe people recite back to you simply because they know better than you? Sorry to burst your bubble but just because we can select "24 bit" or "32 floating point" from a drop-down menu in our DAW during recording and mixing, doesn't mean the actual content uses 144 dB or comes even close to that. People who have been saying to you that "you have never heard music sound like that" are right, since such music doesn't exactly exist ... at least when it comes to commercial recordings and albums.

"24 bit" is a simple marketing trick since it has zero benefits when it comes to audio playback, but is commonly used in audio editing for headroom and convenience reasons.


----------



## RRod

gregorio said:


> We have to be careful here, film sound and music in effect are two very different things, they have very different workflows and distribution chains. We don't apply noise shaped dither in film/TV products, due to considerable amounts of additional processing being required downstream, after the print-master is completed. For this reason distribution is always 24bit or a proprietary lossy compressed format, to avoid any build up of dither, truncation or noise-shaping artefacts. We can't therefore use the same comparison logic as we can with music because there is no 16bit consumer content out there in the film world, let alone a dominant 16bit format in which the application of noise-shaped dither has been standard practise for commercial release for nigh on 20 years.
> 
> G


 
  
 Right. My comment was aimed at how you'd do a blind comparison of something like TrueHD versus a theoretical version of the same material delivered at 16-bits. It simply doesn't do to compare the multi-channel 24-bit track to the stereo 16-bit track and say "ah HA, 24 bits is better".


> Maybe people recite back to you simply because they know better than you? Sorry to burst your bubble but just because we can select "24 bit" or "32 floating point" from a drop-down menu in our DAW during recording and mixing, doesn't mean the actual content uses 144 dB or comes even close to that. People who have been saying to you that "you have never heard music sound like that" are right, since such music doesn't exactly exist ... at least when it comes to commercial recordings and albums.


 
  
 If someone actually made a 24-bit capable system I wouldn't get within a mile of the thing.


----------



## NoteEater

Wow.  It is evidently apparant that many people on this thread have lost quite a bit of their hearing.  Must be, otherwise they could actually hear 24bit.  I am really sorry to hear that you are not able to hear what I hear.  I must be the lucky one.  Maybe I should start being a missionary like my friend Jesus and start curing the deaf as opposed to the blind.


----------



## CraftyClown

noteeater said:


> Wow.  It is evidently apparant that many people on this thread have lost quite a bit of their hearing.  Must be, otherwise they could actually hear 24bit.  I am really sorry to hear that you are not able to hear what I hear.  I must be the lucky one.  Maybe I should start being a missionary like my friend Jesus and start curing the deaf as opposed to the blind.


 
  
 Wow indeed! Perhaps your friend Jesus blessed your ears?


----------



## Cerastes

noteeater said:


> Must be, otherwise they could actually hear 24bit.


 
  
 It has very little to do with our hearing actually.
  
 If the music being recorded has less than 120 dB of dynamic range _(the limit for 16 bit audio, EDIT: with Dither)_ then it makes no difference if the 24 bit is being used or if the bit depth is increased from 16 bit to 24 bit - the dynamic range of the recording will remain unchanged. If the music doesn't go over that 120 dB threshold, then it's not a 24 bit recording.
  
 Also ironically, if you would actually try to hear the difference between 16 bit and 24 bit music, you would have to blast the music so loud that it would permanently damage your hearing _(not even mentioning issues with the ambient noise floor)_.
  
 So it begs the question, are you sure you even understand what bit depth is and what it does?
  


noteeater said:


> I must be the lucky one.


 
  
 Based on your recent comments, no ... just ignorant.
  
 If you want a simple & quick explanation how the bit depth works, you could check out this video for example.


----------



## watchnerd

noteeater said:


> [deleted bunch of irrelevant stuff]


 
  
 Nothing you said provides credible refutation of the science.
  
 If you're willing to learn, we can point you to several sources to clarify things for you.
  
 If you're not willing to learn, why are you here?


----------



## dukefx

> Originally Posted by *Cerastes* /img/forum/go_quote.gif
> 
> If the music being recorded has less than 120 dB of dynamic range _(the limit for 16 bit audio)_ then it makes no difference if the 24 bit is being used or if the bit depth is increased from 16 bit to 24 bit - the dynamic range of the recording will remain unchanged. If the music doesn't go over that 120 dB threshold, then it's not a 24 bit recording.


 
 96dB actually (6dB/bit), but yes, it's like living in a room that has a much higher (16bit) than required (10-12bit) ceiling and someone is bragging that their ceiling is so much higher (24bit) while having the exact same square meters of space.


----------



## NoteEater

You guys are so wrapped up in any scientific article that refutes 24bit as much as you can as opposed to doing some subjective listening yourself. You should all know by now measurements and graphs do not necessarily equate to what we actually hear.  Have you guys not seen the research done by guys doing measurements that realize their measurements and science does not mean everything. We still have to listen. And not just on headphones.  
  
 Everyone's ears are different.  Apparently mine are much different than those here as well.  And I love listening to crystal clear transparent 24bit FLAC files through these wonderful ears of mine at 90-100db.  Sounds AMAZING!!!. Live on 24bit HiRes Audio!!!


----------



## U-3C

noteeater said:


> You guys are so wrapped up in any scientific article that refutes 24bit as much as you can as opposed to doing some subjective listening yourself. You should all know by now measurements and graphs do not necessarily equate to what we actually hear.  Have you guys not seen the research done by guys doing measurements that realize their measurements and science does not mean everything. We still have to listen. And not just on headphones.
> 
> Everyone's ears are different.  Apparently mine are much different than those here as well.  And I love listening to crystal clear transparent 24bit FLAC files through these wonderful ears of mine at 90-100db.  Sounds AMAZING!!!. Live on 24bit HiRes Audio!!!




I'm pretty sure everyone is here because they love music and they either listen to it every day or even work on it as a profession due to pain. 0.0;


----------



## NoteEater

24bit Audio fills the air differently.  If you don't feel that, sorry you don't.  Many do.  
  
 I just don't understand why people have to refute us ignorant people so much.  Why not just let us have our 24bit audio and leave it alone.  You don't have to buy it.  And in fact the cost of your 16bit files will just be less if there is higher bit depth/bit rate for more money. You need competition to bring your prices down.  
  
 I understand the statements regarding Dolby Digital/DTS.  The movie industry forever has been doing anything they can to save space on physical media.  If the movie studios can get away with not lossy Dolby Digital but a 16bit uncompressed soundtrack as opposed to a 24bit uncompressed soundtrack, why wouldn't they save the space.  Space is a premium on disc and for a two hour movie, that is a considerable amount of space to use on something that has no sonic benefit. 
  
 I use the analogy because many of you are listening to 24bit audio in movies but cannot acknowledge that this was a benefit to movie soundtracks. 
  
 I am just tired of this kind of talk ruining it for those that want HiRes Audio.  That is all I am trying to say guys.  Stop ruining it for everyone interested.  Let us wear our Emperors clothes and stop trying to change my mine and many other peasants minds. You have your opinion.  Let us have ours.  
  
 24bit audio is a sensory perceived experience, not necessarily an auditory one.  Again, it fills the space differently unachievable on headphones. Sorry but the truth.  Scientifically proven that waveforms at many frequencies cannot fully develop in an a headphone.  You won't get the sensation I get in my room or car.  
  
 I get I can't hear above 20khz guys.  I know what sampling frequencies and bit depth refer to.  I have been reading about this stuff for 30 years.  Started spinning my brothers vinyl records when I was 5 years old.  Regardless of all the science I have read, I still know how to do subjective listening and make my own conclusions regardless of the science. 
  
 Its so funny how these forums tend to not allow people to have their own opinion, but whenever someone tries to take "your" opinion from you, the world is going to end.


----------



## dukefx

noteeater said:


> You guys are so wrapped up in any scientific article that refutes 24bit as much as you can as opposed to doing some subjective listening yourself. You should all know by now measurements and graphs do not necessarily equate to what we actually hear.  Have you guys not seen the research done by guys doing measurements that realize their measurements and science does not mean everything. We still have to listen. And not just on headphones.
> 
> Everyone's ears are different.  Apparently mine are much different than those here as well.  And I love listening to crystal clear transparent 24bit FLAC files through these wonderful ears of mine at 90-100db.  Sounds AMAZING!!!. Live on 24bit HiRes Audio!!!


 
 It seems to us that you are delusional, which is quite common among audiophiles, no offense. Specs (and prices) tend to create a "this has to be better" feeling and most of the time they do. You are fooled by your own brain. This is why gain matched blind testing was invented. Btw. 80+ dB for an extended period can cause hearing loss. A symphonic orchestra is at about 60dB which is loud enough. Add another 10 if you want to party.


----------



## NoteEater

Here is a transcript for Scientific American.  Someone else was doing research too and actually found more than 50% of people that can DBT pick the higher than CD quality file.  Interesting huh? 
  
 Published just 6 months ago.  Hmmm...
  
 Hi Res is not an auditory experience in that we will not hear frequencies outside our range, however we do get a feeling or sensory perception from it.  So you guys are right to a point with regards to human hearing and I have always known that.  There is more going on here than not being able to hear like a dog.  
  
 You want science.  I give you science and a scientist that is changing his listening habits due to his own research. 
  
 Maybe we all need to come here for some learning.
  
Jay Z's "Tidal" platform promises listeners CD-quality streaming music, in all its 44.1 kilohertz, 16 bit glory—much better, they say, than compressed files, like mp3s. But why stop there? Neil Young's PonoMusic Store sells music that's even _better_ than CD quality. 
In a YouTube video for the service Young compares mp3 listeners to scuba divers, muddling around the seafloor. "You know you're walking around in the murk and there's big fish down there, that's kind of like listening to an mp3." 
CD listeners are underwater, too. The only way to rise to the top, he says, is to dial up sample rate to over _four times_ that of CD: to 192 kilohertz. "When you make it to 192, you actually break through the surface, and you're breathing air. And the feeling is different, it actually is a visceral relief. You feel good."
But… how good? What researchers, record producers, audiophiles, sound engineers, want to know is: "Is CD, compact disc, enough?" Joshua Reiss (RICE), who leads audio engineering research at Queen Mary University of London. "And the arguments seem to be never-ending."
Reiss took a stab at settling the argument with a meta-analysis—a study of studies—on whether people can really perceive better-than-CD quality sound. He analyzed data from 18 studies, including more than 400 participants and nearly 13,000 listening tests. Overall, listeners picked out the better-than-CD-quality track 52.3 percent of the time. Statistically significant, if not all that impressive.
But when Reiss isolated studies that _trained_ listeners first and gave them a chance to feast their ears on the difference, their odds of picking the higher-quality track climbed to 60 percent. Suggesting there may actually be _some_ perceptible difference... at least enough to convince Reiss to change his listening habits. "Yes I think I actually will, based on this." The analysis is in the _Journal of the Audio Engineering Society_. [Joshua D. Reiss, A Meta-Analysis of High Resolution Audio Perceptual Evaluation]
Not that it will settle _all _arguments. "No, no never. But what I think it might do is allow the researchers to move on a little bit from this question and to start looking deeper into the causes of the perception." And for the audiophiles out there? It's no doubt music to their ears.
—Christopher Intagliata


----------



## NoteEater

You guys going to refute this science too?  Or are you open to new research?
  
 http://www.aes.org/e-lib/browse.cfm?elib=18296
  
 All the research you guys have been discussing was older research.  Playback systems were not even up to par or controlled most likely.  
  
 I have read all the same papers you guys have.  I just don't need to recite science to make my points.  This research here is making me love science more and more.


----------



## NoteEater

Want to retract your delusional statement.  I have not started directly attacking any of you.  Not cool guys.  Sorry you must not be a trained listener yourself or you simply just do not know what to listen for otherwise you would feel what I feel.


----------



## CraftyClown




----------



## dukefx

noteeater said:


> You guys going to refute this science too?  Or are you open to new research?
> 
> http://www.aes.org/e-lib/browse.cfm?elib=18296
> 
> ...


 
 I think I can speak for all of us when I say we are open for any kind of research *done properly*. Most of the time the method used is flawed and doesn't account for many things. Having read most of that paper you linked I can sum it up easily: people can't differentiate and are just guessing which pretty much proves our point.


----------



## NoteEater

Proves your point how.  I think you need to read further and maybe others here can reeducate themselves since we are all so willing to learn.  This was not one test by the way.  This is data pulled from 18 studies.  The researcher himself decided he needed to start listening to 24bit audio due to this research.  If we want to trust scientists, maybe we should take one out of this researchers palybook and do some subjective 24bit listening and feel the AIR Man!!!


----------



## dukefx

I've read most of those. Do I need to start quoting? (I fear it may exceed the character limit) Your own quote also proves it. "_more than 50% of people that can DBT pick the higher than CD quality file_". Well, if you have to guess if A or B is better your chances are exactly 50% which is also included in that study. Take a low sample (participants) and it might deviate more, pick a large sample it gets a lot more accurate, simple statistics. Also... saying "equipment back then was crap" is pure ignorance. I can name you amps that can beat the hell out of a lot of today's high ends. A lot of people are also going back to R2R DACs (another one of those threads I like to read here). Technology hasn't advanced that much in this field except for Class D.


----------



## Mikko Peltonen

Didn't bother to read the whole discussion as it's thousands of posts long, but here's one excellent article.
  
 https://people.xiph.org/~xiphmont/demo/neil-young.html
  
 Sums it up pretty nicely.


----------



## U-3C

noteeater said:


> I am just tired of this kind of talk ruining it for those that want HiRes Audio.  That is all I am trying to say guys.  Stop ruining it for everyone interested.
> 
> 24bit audio is a sensory perceived experience, not necessarily an auditory one.  Again, it fills the space differently unachievable on headphones. Sorry but the truth.  Scientifically proven that waveforms at many frequencies cannot fully develop in an a headphone.  You won't get the sensation I get in my room or car.
> 
> ...




It is prohibited to talk about this stuff scientifically anywhere else on the forum, and it will be deleted or banned based on the TOS. 

Indeed, Head-Fi does not allow one to have ones opinion or facts mentioned outside of Sound Science. However, you are here and you are participating in this debate, so it's welcomed for you to explain why 24 bit audio can sound better and can continue this thread.

Anywhere else, you will be the only one not being insta-banned for voicing your opinions or facts, or everyone but you will be locked out of it.


----------



## NoteEater

You are missing the point of his research.  His point is how much people's perception improved after gaining some experience in how to listen better.  There is no such thing as golden ears.  But there are certainly trained ears.  He found trained ears did better than those that were untrained. If you don't know what you are supposed to perceive, your brain automatically focuses on the perception of something else.  Hence you are missing out on the feeling.


----------



## RRod

A cherry-picked meta-analysis of a bunch of non-replicated studies that confounds bit depth with sample rate. Nothing to see here beyond the trolling at this point. Weren't you peacing-out, btw?


----------



## castleofargh

noteeater said:


> 24bit Audio fills the air differently.  If you don't feel that, sorry you don't.  Many do.
> 
> I just don't understand why people have to refute us ignorant people so much.  Why not just let us have our 24bit audio and leave it alone.  You don't have to buy it.  And in fact the cost of your 16bit files will just be less if there is higher bit depth/bit rate for more money. You need competition to bring your prices down.
> 
> ...


 
 the bold part is you admitting that you can't even read the name of the section where you're posting.
 coming in this section and whine that we don't let you have your 24bit audio and leave it alone, is the pot calling the kettle black. why can't a clear subjectivist let us have this small space of objectivity and leave it alone?


----------



## U-3C

castleofargh said:


> the bold part is you admitting that you can't even read the name of the section where you're posting.
> coming in this section and whine that we don't let you have your 24bit audio and leave it alone, is the pot calling the kettle black. why can't a clear subjectivist let us have this small space of objectivity and leave it alone?




Lol.


----------



## uchihaitachi

noteeater said:


> You guys are so wrapped up in any scientific article that refutes 24bit as much as you can as opposed to doing some subjective listening yourself. You should all know by now measurements and graphs do not necessarily equate to what we actually hear.  Have you guys not seen the research done by guys doing measurements that realize their measurements and science does not mean everything. We still have to listen. And not just on headphones.
> 
> Everyone's ears are different.  Apparently mine are much different than those here as well.  And I love listening to crystal clear transparent 24bit FLAC files through these wonderful ears of mine at 90-100db.  Sounds AMAZING!!!. Live on 24bit HiRes Audio!!!


 

 Hey, can you just upload a foobar abx log?


----------



## gregorio

noteeater said:


> [1] Wow.  It is evidently apparant that many people on this thread have lost quite a bit of their hearing.  Must be, otherwise they could actually hear 24bit. [2]  I am really sorry to hear that you are not able to hear what I hear.


 
  
 1. Yep, my ears are so bad that I make a good living from them.
  
 2. I'm not, if I were able to hear what you hear then I wouldn't make any money at all, let alone a good living!
  
 You want to delude yourself, that's fine. You want to ignore the science without which you wouldn't have any recordings or anything to play them back on, OK that's rather silly but still fine. You want to come here to the science forum and talk about how you can hear magic, fairies and Jesus and quote science you don't understand and dismiss the rest of science, that's not fine. You know that's not fine because you're in the SCIENCE forum, and you call us not cool, sheesh! Do everyone a favour, including yourself, use this forum to learn how the stuff you're listening to actually works, or leave!!
  
 G


----------



## NoteEater

This makes no sense.  You feel the thread owner created this thread to only hear people agree with him/her that there is no benefit to 24bit over 16bit?  Come on.  The thread was clearly started to get a rise out of anyone that does hear/feel a difference with hi res audio only to coordinate a lot of minions here in order to attack anyone that does not agree it is a myth.


----------



## dukefx

That's not really the case here. It's a science forum and you are expected to reason with us using science. Saying you have magical ears is no argument. If you really want to convince people that there is actually a difference then you have to come and say something like: "hey folks, forget all the measurements you know, here's a new, unconventional method that shows the difference." Inconclusive studies and others that prove *our* point won't really help.


----------



## gregorio

noteeater said:


> You feel the thread owner created this thread to only hear people agree with him/her that there is no benefit to 24bit over 16bit?  Come on.  The thread was clearly started to get a rise out of anyone that does hear/feel a difference [blah, blah, blah]


 
  
 So, not only are you apparently incapable of understanding a very simple explanation of why 24bit as a music distribution format is nothing more than a marketing ploy but now you're completely making up nonsense as to why this thread was started. AGAIN, if you want to remain ignorant and deluded that's up to you but don't try and defend your ignorance/delusions in this forum by attacking everyone else who is not so ignorant/delusional!!!
  
 G


----------



## pinnahertz

noteeater said:


> 24bit Audio fills the air differently.  If you don't feel that, sorry you don't.  Many do.


 
 Yes, some feel that way. Not "many" with respect to the total population of those listening to digital audio. 24bit Audio doesn't "fill the air" at all. It's very analog before it gets even close to the air.


noteeater said:


> I understand the statements regarding Dolby Digital/DTS.  The movie industry forever has been doing anything they can to save space on physical media.  If the movie studios can get away with not lossy Dolby Digital but a 16bit uncompressed soundtrack as opposed to a 24bit uncompressed soundtrack, why wouldn't they save the space.  Space is a premium on disc and for a two hour movie, that is a considerable amount of space to use on something that has no sonic benefit.



 Space for audio on BD is no longer at a premium. The above logic has no basis in reality for the BD medium.


noteeater said:


> I use the analogy because many of you are listening to 24bit audio in movies but cannot acknowledge that this was a benefit to movie soundtracks.


 
 I'll defer to gregorio on this too, but in post 24bits does provide a lot more room for manipulation, but it's not base quality issue. It's well known that for release, 16 would be enough, but there's also no need to truncate if the original is already 24.
 Also keep in mind there are no ADCs...and I mean there's like one exotic exception...that actually produce a true 24 bit dynamic range. All are limited to substantially fewer bits of DR, like 20. The words dribbling out are 24 bits, the bottom 4-6 bits are all noise.


noteeater said:


> 24bit audio is a sensory perceived experience, not necessarily an auditory one.  Again, it fills the space differently unachievable on headphones. Sorry but the truth.  Scientifically proven that waveforms at many frequencies cannot fully develop in an a headphone.  You won't get the sensation I get in my room or car.


 
 Science has not proven that 24 bit audio is even clearly discernable, though.


noteeater said:


> Proves your point how.  I think you need to read further and maybe others here can reeducate themselves since we are all so willing to learn.  This was not one test by the way.  This is data pulled from 18 studies.


 
 The conclusions, specifically, were, "In summary, these results imply that, _*though the effect is perhaps small and difficult to detect*_, the *perceived fidelity* of an audio recording and playback chain *is affected* by operating beyond conventional consumer oriented levels."
  
 Note specifically, "small and difficult to detect", and that percieved fidelity is _*affected*_.  He didn't say "improved".  If you look at the studies cited in the paper, you'll also find that not a single person was able to detect hi-res every time.  Not one set of "golden ears" in the whole bunch.  
  
 Now, if you want to apply the results in the paper and face value, then go on to claim emphatically that hi-res (not 24bit alone, BTW) is somehow a night/day thing that everybody on the planet can hear, well the paper doesn't say that at all, quite the opposite.   The results in the paper are 3% better than flipping a coin.  
  


noteeater said:


> The researcher himself decided he needed to start listening to 24bit audio due to this research.


 
 No, Reiss already held these positions:
 • Co-Chair of the Audio Engineering Society (AES) Technical Committee on High-Resolution Audio
 • General Chair of the 31st AES Conference; New Directions in High Resolution Audio, 2007
 Pretty sure he'd been listening to high-res for a bit. If anything, those positions imply a bias.


noteeater said:


> If we want to trust scientists, maybe we should take one out of this researchers palybook and do some subjective 24bit listening and feel the AIR Man!!!


 
 If you want to really feel the air, go hang gliding. Trust me on that one.


----------



## Ruben123

noteeater said:


> 24bit Audio fills the air differently.  If you don't feel that, sorry you don't.  Many do.
> 
> I just don't understand why people have to refute us ignorant people so much.  Why not just let us have our 24bit audio and leave it alone.  You don't have to buy it.  And in fact the cost of your 16bit files will just be less if there is higher bit depth/bit rate for more money. You need competition to bring your prices down.
> 
> ...




You said it well. Wait, small correction: let you have your opinions, let us have our facts. I haven't seen one single fact from you. 24 bits filling the air differently? Are you sure my friend?


----------



## watchnerd

noteeater said:


> You guys are so wrapped up in any scientific article that refutes 24bit as much as you can as opposed to doing some subjective listening yourself. You should all know by now measurements and graphs do not necessarily equate to what we actually hear.


 
  
 I will repeat, since apparently it didn't register the first time.
  
 1. I record in 24bit.  I distribute in 16bit.
  
 2. I have done ABX tests of my own 24bit masters vs dithered 16bit SRC using SOX of the same master.  They are transparent to each other.


----------



## TheoS53

watchnerd said:


> I will repeat, since apparently it didn't register the first time.
> 
> 1. I record in 24bit.  I distribute in 16bit.
> 
> 2. I have done ABX tests of my own 24bit masters vs dithered 16bit SRC using SOX of the same master.  They are transparent to each other.


 
 Out of interest, have you tried doing a null test with 2 recordings (16 vs 24 bit)?


----------



## watchnerd

theos53 said:


> Out of interest, have you tried doing a null test with 2 recordings (16 vs 24 bit)?


 
  
 Passband limited or not?


----------



## TheoS53

watchnerd said:


> Passband limited or not?


 
 Errm. not sure...what's the difference?


----------



## NoteEater

You Win.  16 bit is the same as 24bit.  No difference.  Happy now. You can now move on to your next prey.  Have fun guys


----------



## pinnahertz

Darn.  Just when I was going to have some fun with "a cassette deck sounding better than any know reel to reel", "absolute phase", "mic response to 5Hz", "recording DC"  and my favorite, "24bit Audio fills the air differently."  
  
 Oh well, really didn't have enough time...in my life...


----------



## TheTrace

noteeater said:


> You Win.  16 bit is the same as 24bit.  No difference.  Happy now. You can now move on to your next prey.  Have fun guys


Damn the war has ceased after years of heated debate. 

Now for American politics.


----------



## spruce music

NoteEater,
  
 What proof do you have other than your ears?
  
 Under what conditions would you accept that your ears/brain have fooled you into thinking 24 bit sounds vastly better than 16 bit when it isn't actually true?


----------



## U-3C

thetrace said:


> Damn the war has ceased after years of heated debate.
> 
> Now for American politics.


 
 I just got reported and received a warning for posting a pic of yugioh cards and a certain political figure.
  
 Don't go there! D:


----------



## pinnahertz

spruce music said:


> Under what conditions would you accept that your ears/brain have fooled you into thinking 24 bit sounds vastly better than 16 bit when it isn't actually true?


 
 Three sheets to the wind for starters....


----------



## watchnerd

theos53 said:


> Errm. not sure...what's the difference?


 
  
 Well if it's not passband limited it certainly can't null.


----------



## TheoS53

watchnerd said:


> Well if it's not passband limited it certainly can't null.


 
 what I mean is, taking 2 recordings of the exact same song (for example), one at 24 and the other at 16...overlaying them and inverting one, then seeing what the result is. If they're identical, there should be zero output


----------



## spruce music

theos53 said:


> what I mean is, taking 2 recordings of the exact same song (for example), one at 24 and the other at 16...overlaying them and inverting one, then seeing what the result is. If they're identical, there should be zero output


 

 Or the result could be a non-zero output.  Yet one so low that the difference could never alter the sound in a way audible to humans.  Playing the difference signal at normal listening volume the human listener will hear complete silence even though the actual signal level is not zero.


----------



## castleofargh

theos53 said:


> watchnerd said:
> 
> 
> > Well if it's not passband limited it certainly can't null.
> ...


 

 why would they be identical? and what would that demonstrate? we're discussing audibility while playing music at normal listening levels. not pretending that 24bit files don't have extra data compared to 16bit. so aside from demonstrating what is already a consensus, i.e. that bigger bit depth can store quieter data, I don't get the point.


----------



## TheoS53

spruce music said:


> Or the result could be a non-zero output.  Yet one so low that the difference could never alter the sound in a way audible to humans.  Playing the difference signal at normal listening volume the human listener will hear complete silence even though the actual signal level is not zero.


 
  
  


castleofargh said:


> why would they be identical? and what would that demonstrate? we're discussing audibility while playing music at normal listening levels. not pretending that 24bit files don't have extra data compared to 16bit. so aside from demonstrating what is already a consensus, i.e. that bigger bit depth can store quieter data, I don't get the point.


 


 The point would be to demonstrate to those who say that they can hear a difference, that in fact it would be practically impossible for them to do so. Whilst a difference between 16 and 24 bit can be measured and represented graphically, that same graphic could be used to illustrate how it would technically be possible for humans to actually hear that difference (because it falls within the audible spectrum), but probably not loud enough. 

 Anyways, I've done exactly that:

 192khz vs 96khz:



 192khz vs 48khz:



 192 vs 44.1-16bit:
 

 We can see that for the 16 bit file, there is some noise present within the audible spectrum. But is it loud enough for us to hear? Probably not. But the difference between 48khz and 192khz is absolutely inaudible


----------



## watchnerd

theos53 said:


> what I mean is, taking 2 recordings of the exact same song (for example), one at 24 and the other at 16...overlaying them and inverting one, then seeing what the result is. If they're identical, there should be zero output


 
  
 They're not identical, nor would they be expected to be -- they have different data in them.


----------



## TheoS53

watchnerd said:


> They're not identical, nor would they be expected to be -- they have different data in them.


 
 check my previous post


----------



## watchnerd

theos53 said:


> check my previous post


 
  
 Right.


----------



## Brooko

I guess you guys all missed this bit - which struck me as hilariously funny.


noteeater said:


> I have been listening to 24bit audio for 18 years now and have enlightened so many people.


 
  
 along with
  


noteeater said:


> Everyone's ears are different.  Apparently mine are much different than those here as well.  *And I love listening to crystal clear transparent 24bit FLAC files through these wonderful ears of mine at 90-100db*.  Sounds AMAZING!!!. Live on 24bit HiRes Audio!!!


 
  
 So 18 years at 90-100 dB, and claiming you guys are deaf.
  
 I'm still giggling at it.
  
 Anyway - hope you didn't mind that I didn't step in.  Realistically, some of the posts were getting the tiniest bit antagonistic at time.  Just remember that you don't need to get personal with some of these guys.  Just point them to the actual science, and then if they are obviously trying to troll the thread - flag it, and we'll get them back on track.


----------



## gregorio

pinnahertz said:


> I'll defer to gregorio on this too, but in post 24bits does provide a lot more room for manipulation, but it's not base quality issue. It's well known that for release, 16 would be enough, but there's also no need to truncate if the original is already 24.


 
  
 Not sure if you or anyone else is interested but that's not really how it works, in fact for mixing, editing, production or post 16bit is still perfectly sufficient, IE. 24bit does not provide a any more room for manipulation. What happens in practice is that the DAW creates an internal (virtual) mixer, typically today 64bit float or 48bit fixed. Our 16bit recorded tracks are streamed to this mixer where we apply any amount of processing, either with plugin processors or with built-in processing. Let's say all we do is reduce the output fader level of a recorded track by 6 dB, this effectively shifts all the data down by one bit. So what was in the 16th bit (LSB) would now be in the 17th bit but as our recording only has 16bits we're going to loose whatever was in that 16th bit (17th bit after the 6dB fader reduction). This isn't a serious problem because that 17th bit was probably just noise anyway but if we apply several processing steps, each time truncating that 17th bit before that result is passed on to the next processing step we're quickly going to accumulate truncation errors to the point of audibility. Fortunately, this is not what actually happens! What happens is that all the 17bits of our result is maintained within our 64f/56bit virtual mixer, truncation of processing results still occurs but at the point of the 48th bit (in the case of a 48bit fixed virtual mixer). We could apply several hundred different processors to our 16bit recorded track before the build-up of error/s gets even close to audibility. Of course we can't actually output that 48bit virtual mix, it's truncated to 24bit for output to our studio DAC for monitoring or when the mix is completed we can record it down also as a truncated 24bit file or we can choose to dither it to either a 24bit file (although there's no point) or noise-shape dither it to a 16bit file. ... So, it's irrelevant whether we feed our virtual mixer with 16bit or 24bit recordings, the "room for manipulation" is 48bit fixed or 64bit float either way. The only time 24bit has any use at all is when recording. Until we've actually completed the recording (by which time it's too late) we don't know what the peak level is going to be. 24bit provides very useful additional headroom, a raw 24bit recording can easily stand having the first few MSBs empty without loosing the quietest parts to the digital noise floor as could be the case if recording at 16bit.
  
 However, the above does not apply with film sound. With music we're typically dealing with <24 recorded files per song, even in the most extreme cases it's rarely more than about 100. In film, say an average blockbuster, there are usually tens of thousands of recorded files to deal with! There is simply no way of mixing/processing all those recordings in real time in a virtual mixer, either in terms of available processing power or logistics, it has to be sub-mixed (mixed into different groups of related audio material), recorded down and then those recorded sub-mixes mixed together to create the full, final mix. Obviously, when we record these sub-mixes we have to leave our 48/64bit mix environment and as there are usually several sub-mixing steps we'd run into the problems mentioned above (what would happen if a virtual mixer were only 16bit), so it's always been required procedure that these submixes are recorded at 24bit. ... In practise I've over-simplified the process somewhat compared to what really occurs today, where far more of a film mix is kept within the virtual mix environment than was once necessary. I was involved (uncredited additional technical support) in one of the early implementations of this new film mix paradigm in 2004 on the Harry Potter Prisoner of Azkaban film, were we sync-locked 13 ProTools HD systems to effectively provide a virtual mix environment capable of 2,496 simultaneous channels of audio. Even so, some submixing was still necessary, for example a single explosion (including flying debris) can use 192 channels for just a second or two of sound, so that needs to be mixed down (to 7.1) and that 2 second, 8 channel submix then brought into the full mix.
  
 Quote:


theos53 said:


> We can see that for the 16 bit file, there is some noise present within the audible spectrum. But is it loud enough for us to hear?


 
  
 I can't know for sure but that noise appears to be what one would expect to see from the application of noise-shaped dither, although in that case I'd expect it to be rather higher in level, maybe up around -80dBFS or so. Either way, being in the least sensitive area of the audible spectrum it's certainly inaudible, even noise-shaped dither at -80dBFS is inaudible and your graph indicates -105dBFS in the same frequency band (>17kHz)!
  


noteeater said:


> You Win.  16 bit is the same as 24bit.  No difference.  Happy now.


 
  
 No we're not, because you're saying that just to avoid defending your views with more illogical nonsense and getting yourself banned but it's not going to stop you trying to "convert" others who don't know enough to realise it's all illogical nonsense. To take a popular quote of irrefutable logic, "_when you have eliminated the impossible, whatever remains, however improbable, must be the truth!_". In this case, the impossible is effectively that there is no audible difference between 16 and 24bit, which you'd realise if you'd read the original post (and understood it). What remains is the effect of human biases/perception and the differences you hear are ONLY in your (and others') mind. This MUST be the truth, no matter how improbable it seems to you!
  
 The problem with many of the more extreme audiophiles is that they get confused between what is "impossible" and what "remains". They believe it's impossible that what they perceive is only in their mind rather than an actual reality, so they eliminate perception as a cause of differences and what remains is the science, which however improbable, must therefore be wrong about there not being any audible differences. Irrefutable logic appears to be satisfied but of course it's not because this conclusion is a fallacy. You can't have an opinion on whether the basic science explained in the OP is wrong because it's been mathematically proven (as you'd know if you'd read it). Furthermore, not only is it proven mathematically but is demonstrated in practice in billions of devices we use every day. All digital equipment, computers, cell phones, sat navs, etc. etc. are based on that same proven maths. And even if you could un-prove that math, how can you prove that all digital devices don't exist (or do exist but never work)? That would obviously be completely irrational, illogical nonsense but that is in effect exactly what you are arguing if you're saying the science is wrong!! The "improbable" which must be the truth is not that science is wrong it's that you're being fooled by your senses: "Seeing is believing" is patently not true (unless you believe Star Wars, Star Trek, Harry Potter and Marvel films are all factual documentaries), "hearing is believing" is just as easily fooled and incidentally, just as consistently deliberately fooled as sight is in films. Wise up, if not for your own sake, then for the sake of those you're trying to convert!!
  
 G


----------



## pinnahertz

gregorio said:


> Not sure if you or anyone else is interested but that's not really how it works, in fact for mixing, editing, production or post 16bit is still perfectly sufficient, IE. 24bit does not provide a any more room for manipulation. What happens in practice is that the DAW creates an internal (virtual) mixer, typically today 64bit float or 48bit fixed.


 
 I wasn't referring to mixing and internal bit depth structure, I know that's how it works, but thanks for providing the explanation, It's probably not commonly known.


gregorio said:


> The only time 24bit has any use at all is when recording. Until we've actually completed the recording (by which time it's too late) we don't know what the peak level is going to be. 24bit provides very useful additional headroom, a raw 24bit recording can easily stand having the first few MSBs empty without loosing the quietest parts to the digital noise floor as could be the case if recording at 16bit.


 
 I was thinking of that and...


gregorio said:


> Obviously, when we record these sub-mixes we have to leave our 48/64bit mix environment and as there are usually several sub-mixing steps we'd run into the problems mentioned above (what would happen if a virtual mixer were only 16bit), so it's always been required procedure that these submixes are recorded at 24bit.


 
 ...that. It's the "law of uncorrelated summing", you build level 3dB every time you double the channel count (assuming uncorrelated signals at the same level). It's why even analog mixers provide for lots of extra headroom at the mix bus, and a bus master.


gregorio said:


> The problem with many of the more extreme audiophiles is that they get confused between what is "impossible" and what "remains". They believe it's impossible that what they perceive is only in their mind rather than an actual reality, so they eliminate perception as a cause of differences and what remains is the science, which however improbable, must therefore be wrong about there not being any audible differences. Irrefutable logic appears to be satisfied but of course it's not because this conclusion is a fallacy. You can't have an opinion on whether the basic science explained in the OP is wrong because it's been mathematically proven (as you'd know if you'd read it). Furthermore, not only is it proven mathematically but is demonstrated in practice in billions of devices we use every day. All digital equipment, computers, cell phones, sat navs, etc. etc. are based on that same proven maths. And even if you could un-prove that math, how can you prove that all digital devices don't exist (or do exist but never work)? That would obviously be completely irrational, illogical nonsense but that is in effect exactly what you are arguing if you're saying the science is wrong!! The "improbable" which must be the truth is not that science is wrong it's that you're being fooled by your senses: "Seeing is believing" is patently not true (unless you believe Star Wars, Star Trek, Harry Potter and Marvel films are all factual documentaries), "hearing is believing" is just as easily fooled and incidentally, just as consistently deliberately fooled as sight is in films. Wise up, if not for your own sake, then for the sake of those you're trying to convert!!


 
 I think the problem is that audiophiles depend a lot on blind faith to accept mythology, and early on become biased against the evil scientists because they take the fun out of their beloved myth.   And belief is reinforced with conviction.  Others favor real, provable explanations and abhor mythology.  They study science and their belief is reinforced with conviction. 
  
 The first time a neophyte audiophile is exposed to an audiophile master-golden-ear "hearing" some fine detail, then attributing it to a magic tweak, he adopts the "master" as a model, an example to emulate.  The catch-phrases and lingo are adopted ("the veil was lifted' and the like) in order to self-identify with the audiophile-golden-ear master. Then they "evangelize" others into that belief system.
  
 The break-out mechanism, as it is in an belief system, is to ask "why?", and then look for verifiable, repeatable reasons for something to be, without regard to a pre-bias against science.  Like any belief system, it cannot be broken unless the believer recognizes a deficiency, a flaw, or some other form of disillusionment. Then they may search for truth.  If the answer to "why" is clouded in pseudoscience or myth, but sounds satisfying, they become more deeply rooted in their faith, and the search for truth remains parked at the side of the road.  Since the audiophile is biased against science, their source for answers may not often come from that direction, and instead will draw them back to their belief system. 
  
 The biases are extreme, particularly favoring the unscientific over the scientific, and are based in the belief that science is incapable of explanation.  If you disable the alternative belief system, you reinforce the one you have.  Scientific method and proof, then, are easily dismissed in favor of "I know what I hear with my own ears".  
  
 In all the forum battles I've read and been involved with here and elsewhere I've never actually seen a conversion take place in either direction.  The debate may be ultimately "won" by one or the other view in print, but both sides remain unconvinced of the other's viewpoint.  One or the other party leaves the discussion in some way, or the thread is locked.  Eventually both sides settle into their respective strata, often not in the same forums.  And all remains peaceful until somebody comes in and mixes up the water again.  It ends with "supply proof" vs "I know what I hear" (to the exclusion of any recognition of bias).


----------



## TheoS53

pinnahertz said:


> I wasn't referring to mixing and internal bit depth structure, I know that's how it works, but thanks for providing the explanation, It's probably not commonly known.
> I was thinking of that and...
> ...that. It's the "law of uncorrelated summing", you build level 3dB every time you double the channel count (assuming uncorrelated signals at the same level). It's why even analog mixers provide for lots of extra headroom at the mix bus, and a bus master.
> I think the problem is that audiophiles depend a lot on blind faith to accept mythology, and early on become biased against the evil scientists because they take the fun out of their beloved myth.   And belief is reinforced with conviction.  Others favor real, provable explanations and abhor mythology.  They study science and their belief is reinforced with conviction.
> ...


 
 I think in many cases specifically related to all these audio myths or whatever is to ask "is it better?", followed by "does it actually matter?"

 I think many people too (perhaps the education systems are to blame) don't quite understand how to properly test something either. I don't mean in the sense that they don't have access to equipment, but rather the basic understanding that in order to properly compare you need to change as few variables (or ideally only 1) at a time, and to accept the fact that unless you have control over those variables, you cannot assume that the variables will remain the same. 

 Heck, the whole idea that some people have of "I don't care about your science, I know what I'm hearing", is a very human reaction to have. Our brains don't like to be wrong, it needs things to make sense to it, and if things don't make sense, it'll fill in its own BS information until the situation does make sense. It's for this very reason that optical illusions exists. It's not an illusion, it's the brain being caught red handed having made up BS info, lol.


----------



## spruce music

The below is an excerpt from the jriver wiki about volume control in their media center.  It is related to gregario's comments about working at high floating point bit depths.  The errors that build up from multiple changes in level are so small as to end up being non-existent or so close to it as to be a non-issue. 

The precision offered by Media Center's 64bit audio engine is billions of times greater than the best hardware can utilize. In other words, it is bit-perfect on all known hardware.

 To demonstrate the incredible precision of 64bit audio, imagine applying 100 million random volume changes (huge changes from -100 to 100 dB), and then applying those same 100 million volume changes again in the opposite direction.

 Amazingly, you will have the exact same signal at 32bit after 200 million huge volume changes as when you started.

 In other words, this incredible number of changes results in a bit-perfect output at 32bit, which is the highest hardware output bitdepth (most high-end hardware is 24bit).

 This also means one volume change or a series of 100 million volume changes that add up to the same net result is bit-identical.


----------



## lantian

Not sure about the rules of disclosing such things here, but here goes. In my case I can hear the difference between 24bit and 16bit files, here is the big "but". I do not hear much difference when I am sober, then almost all 16bit and 24bit files sound almost the same.


----------



## dukefx

We should include alcohol consumption in the next study


----------



## U-3C

lantian said:


> Not sure about the rules of disclosing such things here, but here goes. In my case I can hear the difference between 24bit and 16bit files, here is the big "but". I do not hear much difference when I am sober, then almost all 16bit and 24bit files sound almost the same.




Yeah, for me, music always sounded way better when I'm happy.

When I try to remove all the biases I can though, I can't hear any difference. The more I think I hear something, the more I hear it. Then I swear I hear the opposite once I confirm that it actually wasn't what I was playing.

D:

Key to enjoying music: be happy!


----------



## lantian

I do not consume alcohol though. Though there is one thing that should be included in the studies, but that will never ever ever happen.
 Agree happiness is key


----------



## spruce music

lantian said:


> I do not consume alcohol though. Though there is one thing that should be included in the studies, but that will never ever ever happen.
> Agree happiness is key


 

 So don't tease us.  Spill the beans.  What one thing should be included in the studies?


----------



## lantian

Is it allowed to say it here?


Spoiler: Warning: Spoiler!



Lucy in the sky with the diamonds


----------



## CraftyClown

lantian said:


> Is it allowed to say it here?
> 
> 
> Spoiler: Warning: Spoiler!
> ...


 
  
 Judging by some of the comments you read, I think a lot of the audiophiles are already experimenting with this


----------



## lantian

craftyclown said:


> Judging by some of the comments you read, I think a lot of the audiophiles are already experimenting with this


 

 In my opinion that is how hifi kicked of, but hey that's just me


----------



## watchnerd

dukefx said:


> We should include alcohol consumption in the next study


 
  
 why limit it to alcohol?


----------



## StanD

You guys have had a reality forum show going on, I missed the entertainment. Must be the audiophile grade of LSD that's been going around, makes one hear things.
 So what kind of precision does one need when multiplying 2 different 16 bit numbers for sound effects like ring modulation? Certainly 16 bit precision in not adequate, unless one is willing to compromise by scaling down the two inputs prior to multiplication.


----------



## castleofargh

please, no more drug stuff for most obvious reasons on a public forum where people are welcome at 13.
  
 Quote:


u-3c said:


> Yeah, for me, music always sounded way better when I'm happy.
> 
> When I try to remove all the biases I can though, I can't hear any difference. The more I think I hear something, the more I hear it. Then I swear I hear the opposite once I confirm that it actually wasn't what I was playing.
> 
> ...


 
 I guess most will agree about this,  but what if
 "I'm only happy when it rains
 I'm only happy when it's complicated"?


----------



## U-3C

castleofargh said:


> I guess most will agree about this,  but what if
> "I'm only happy when it rains
> I'm only happy when it's complicated"?


 
 Hey, my best music listening experiences are from when it is raining and I listen/dance/swing my hands wildly like a madman to music on the balcony, with my Q701 plugged into my computer indoors~
  
 I'm still perfectly fine and alive.
  
 *On a side note*
  
 Th...thanks AKG for your 20 feet cable...


----------



## pinnahertz

stand said:


> You guys have had a reality forum show going on, I missed the entertainment. Must be the audiophile grade of LSD that's been going around, makes one hear things.
> So what kind of precision does one need when multiplying 2 different 16 bit numbers for sound effects like ring modulation? Certainly 16 bit precision in not adequate, unless one is willing to compromise by scaling down the two inputs prior to multiplication.


 
 Ring modulation would be a kind of analog synthesis effect, and as such, I doubt high precision is even desired.  Remember, early FM  synths were 12 bit, then 15 bit.  With today's interest in vintage instruments you probably want that 12 bit grit.


----------



## gzubeck

so what makes some recordings sound insanely smooth vs. recordings that make you want leave the room? Whats the higher sampling rates for? Doesn't a higher sampling rate improve anything? Is it strictly poor equipment and increased loudness?


----------



## pinnahertz

gzubeck said:


> so what makes some recordings sound insanely smooth vs. recordings that make you want leave the room?


 
 Since your observation relates to a difference in the recordings themselves, that would be where to look. There have been good, bad, and inbetween recordings since the beginning. Generally a bad sounding recording, these days, is evidence of lack of skill or misplaced artistic priorities.


gzubeck said:


> Whats the higher sampling rates for?


 
 There is a strong psychological coupling between the concepts of "high" and "better", regardless of the actual result. Proponents would say they are trying to capture all of the audio, regardless if its audible or not. Opponents would say there's no need to capture it if it's not audible, since no difference can be heard. A realist would recognize that simply running a higher sampling rate doesn't ensure the capture and reproduction of ultrasonic frequencies and their delivery to the hear. And it turns out, delivery to the ear is one of the harder things to do.


gzubeck said:


> Doesn't a higher sampling rate improve anything?


 
 It increases the maximum frequency that can be captured. That's not necessarily an improvement.


gzubeck said:


> Is it strictly poor equipment and increased loudness?


 
 Back to those bad sounding recordings are we? It's actually kind of hard to buy really poor equipment these days. The loudness war has resulted in some really horrible sounding recordings. Mastering a recording to be loud is highly destructive, and is one of those misplaced artistic priorities. Ostensibly it's supposed to sell more music. It doesn't. It's much more of a ego thing.


----------



## gregorio

> Originally Posted by *pinnahertz* /img/forum/go_quote.gif
> 
> [1] It's the "law of uncorrelated summing", you build level 3dB every time you double the channel count (*assuming uncorrelated signals at the same level*). It's why even analog mixers provide for lots of extra headroom at the mix bus, and a bus master.
> [2] In all the forum battles I've read and been involved with here and elsewhere I've never actually seen a conversion take place in either direction.


 
  
 1. We can't assume uncorrelated signals though, particularly when mixing music and, they're not at the same level. Nevertheless, even though the "law of uncorrelated summing" is extremely imprecise in practice (the real world of mixing real acoustic sound signal), as we increase the channel count we would build level. However, none of this is directly relevant to the point I was trying to make because 0dBFS (max peak) has exactly the same reference level at both 16 and 24bit. 24bit has no additional headroom relative to 16bit, what it has is more "foot-room". So when recording, 24bit doesn't provide more headroom in terms of allowing us to record a louder signal, it provides more headroom in terms of allowing us to record the signal quieter (by reducing the mic-preamp), thereby increasing the gap between the peak signal level and the max system level (0dBFS). The reason in film we often submix down to a 24bit file rather than 16bit is because there's likely to be numerous further processing steps applied to that recorded file and that result will form part of another submix, which may in terrn form part of anther submix. This isn't the case with music because even though we often employ a (relatively, very simple) submix topology, those submixes don't need to be recorded (to a file) and therefore truncated (to 16 or 24bit), they never have to leave the original (48bit/64bit) virtual mix environment. I'm sure you're probably aware of all this but I thought I'd (attempt to) clarify for the wider benefit.
  
 2. I can't remember seeing such a conversion here (on head-fi) but I've certainly seen it elsewhere. In fact it's very common on pro music/sound engineering forums, where newbies are usually infected by at least some audiophile myths (if not most of them). The difference with pro sound engineering forums is that there is a deeper knowledge, both at the technical and personal level. For example if a newbie quotes some famous producer/engineer/studio/musician extolling the virtues of some dubious product, there will be those present who know first hand why that famous producer/... made that quote in the first place. Although unrelated, I find it interesting that there appears to be a relationship between the pro audio world and the audiophile world, a relationship commonly separated by a decade or more. Some issue will arise in the pro audio community and become a bit of a widely discussed "thing"; the how, why and what of particular observations and solutions sought. Occasionally the issue might be based on a fallacy (which is ultimately dispelled) but more commonly it's a real issue related to the technology at that point in time. 10 or 20 years later that exact same issue with the exact same initial arguments/observations appear in the audiophile world. Although typically it's completely fallacious because either: 1. It was fallacious to start with in the pro audio world. 2. The technology has moved on and the issue no longer exists or 3. The issue was never applicable to consumers anyway. Anti-alias and reconstruction filters (types, ringing, audibility, etc.), quantising error, resolution, noise/dither, bit depth, sampling rate, audio compression, data compression, jitter (types, audibility, etc.) and jitter clocking/re-clocking/rejection, transmission of data down cables (signal types, loss/distortion), to name but a few. It's almost as if audiophile manufacturers are trawling pro audio forum archives looking for some 20 or so year old issue (which never existed or no longer exists), so they can create a product which solves that (non-existent) issue or use it to support an existing product.
  
 Quote:


stand said:


> [1] So what kind of precision does one need when multiplying 2 different 16 bit numbers for sound effects like ring modulation? Certainly 16 bit precision in not adequate, [2] unless one is willing to compromise by scaling down the two inputs prior to multiplication.


 
  
 1. I'm not sure I understand what you're asking. In general, 16bit is enough when multiplying two different 16bit numbers (for whatever sound effect). The problem starts to arise when taking that result and multiplying it with something else and gets worse if you then take that result and multiply it with something else again, and then keep repeating that procedure.  Certainly, 48bit or 64bit float is more than enough for any conceivable, realistic number of such processes/procedures and btw, a 48bit fixed virtual mixing environment is not a new thing, it was introduced about 20 years ago and was pretty much ubiquitous by about 15 years ago.
  
 2. In practice, the 48bit systems did provide considerable headroom, in fact the mix busses were actually 56bit accumulators, with the additional 8 bits used as an additional 48dB of headroom (above 0dBFS). And of course with floating point (32 or 64), headroom is effectively astronomical, as all you're doing in effect is changing the position of the decimal point. Either way though, you do ultimately have to scale it down, either on the input to the various processing (multiplication for example) or the output of all the processing (from the mix buss) because this output has to ultimately go to a file and/or DAC with an integer output which cannot exceed 0dBFS. For example, with a 32bit float mixer you can in theory go up to about +500dBFS (if memory serves) without distortion but you obviously can't actually output that signal, you'd have to reduce the output (fader) level so the peak is less than 0dBFS, so as not to clip the DAC.
  
 G


----------



## TheoS53

gzubeck said:


> so what makes some recordings sound insanely smooth vs. recordings that make you want leave the room? Whats the higher sampling rates for? Doesn't a higher sampling rate improve anything? Is it strictly poor equipment and increased loudness?


 
 I suspect that it is this assumption that has lead to the belief that a higher sampling rate is better. Someone might listen to 2 recordings of different sampling rates, and it just so happens that the higher sampling rate one sounds better, so they assume that it is because of the sampling rate that the song sounds better. But, we cannot assume that, even though it's the same song, that they actually came from the exact same recording


----------



## lantian

Higher sampling rates are simply producing more life like resolution of your music.  Human ear is perfectly capable of hearing the difference between higher sample rates. The so called auditory resolution of human hearing is about 5-7us, a 44.1khz audio has each slice of 20us, 96khz has something like 10us, and only 192khz can offer time resolution of 5us. That would seem perfectly smooth to our auditory systems/brain. Go ahead and correct me If I am wrong on this.


----------



## TheoS53

lantian said:


> Higher sampling rates are simply producing more life like resolution of your music.  Human ear is perfectly capable of hearing the difference between higher sample rates. The so called auditory resolution of human hearing is about 5-7us, a 44.1khz audio has each slice of 20us, 96khz has something like 10us, and only 192khz can offer time resolution of 5us. That would seem perfectly smooth to our auditory systems/brain. Go ahead and correct me If I am wrong on this.


 
 have a look 2 pages back of the images I posted.... in 24 bit resolution it's impossible for any human to hear the difference between 192khz, 96khz, and 48khz, as the files are IDENTICAL below 23khz


----------



## lantian

theos53 said:


> have a look 2 pages back of the images I posted.... in 24 bit resolution it's impossible for any human to hear the difference between 192khz, 96khz, and 48khz, as the files are IDENTICAL below 23khz


 

I am not so sure if that is correct after all audio works on a logarithmic scale, so while they seem identical they are not, I do not posses enough knowledge about this, but  fact is the time resolution changes and human auditory system can pick that up. 
 Never mind should not have said anything


----------



## gregorio

gzubeck said:


> [1] so what makes some recordings sound insanely smooth vs. recordings that make you want leave the room? [2] Whats the higher sampling rates for? [3] Doesn't a higher sampling rate improve anything? [4] Is it strictly poor equipment and increased loudness?


 
  
 Essentially as pinnahertz said but put slightly differently:
  
 1. There's the initial problem of how you define "insanely smooth". For example, a fan of thrash metal would probably have quite a different definition of "insanely smooth" compared to say a fan of Debussy, if a thrash metal fan would even think "insanely smooth" desirable in the first place! Let's say, for argument sake, we all want "insanely smooth" and all define it the same, still there is no simple answer to your question as there can be many causes of a recording not being smooth; most typically a musician not performing smoothly, the instrument they're playing or the acoustics not responding smoothly but just as possibly; not optimally chosen mics, mic positioning, mic-preamp and pre-amp setting or downstream, some processing which reduces smoothness and further downstream still, some consumer equipment or environment not responding as smoothly to the frequency content of that particular piece of music/recording as to another.
 2. Essentially as far as the consumer is concerned, they're for marketing purposes.
 3. For the consumer, no. In fact it's likely to actually make it worse rather than improve anything, but we're only talking about slightly worse.
 4. No, as mentioned in #1 there are all sorts of potential causes and not all of them about poor equipment. I'm going to disagree with pinnahertz somewhat here; mastering a recording to be loud is not necessarily "highly destructive" it can just as easily be highly beneficial and likewise, not mastering a recording to be loud can be in effect even more "highly destructive", it all depends on our listening equipment, environment and what we're actually listening for at any particular instant in time.
  
 G


----------



## castleofargh

lantian said:


> Higher sampling rates are simply producing more life like resolution of your music.  Human ear is perfectly capable of hearing the difference between higher sample rates. The so called auditory resolution of human hearing is about 5-7us, a 44.1khz audio has each slice of 20us, 96khz has something like 10us, and only 192khz can offer time resolution of 5us. That would seem perfectly smooth to our auditory systems/brain. Go ahead and correct me If I am wrong on this.


 
 I'll let those who didn't stop using math the day school ended
	

	
	
		
		

		
		
	


	




, explain in detail if they feel like it .but basically it's the same mistake as the great classic "humans have 140db of dynamic range so of course 16bit isn't enough!". the values come from something real but specific, and are then abused out of context to try and make a point.
 get me loud iterated ripple noises and square waves as the only sounds in your favorite song, and then we can discuss some 5 to 10µs delays audible to humans in "music".
  
 for the "sample rate to period" thing, it doesn't mean what you seem to think it means. the actual signals in music are sine waves, the actual resolution will depend on the bit depth and loudness of the signal, and most certainly does better than 20µs. to degrade to any not even significant values, we'd need to go look at very low levels, which of course negates the 5µs of the paper anyway as nobody would ever pass the test at very low volume levels.
 so it's pretty safe to say that you are wrong on this. the results of a specific test are valid for the specific conditions of the test.


----------



## pinnahertz

gregorio said:


> 1. We can't assume uncorrelated signals though, particularly when mixing music and, they're not at the same level. Nevertheless, even though the "law of uncorrelated summing" is extremely imprecise in practice (the real world of mixing real acoustic sound signal), as we increase the channel count we would build level.


 
 Actually, you can assume uncorrelated. Correlated summing, and the resulting 6dB voltage increase when the number of summed channels doubles, only happens when you sum perfectly identical waveforms, in phase, and at an identical level.  As you increase the total number of channels, the rare possibility of that happening gets buried in the huge number of uncorrelated channels being summed.  Designing for correlated summing would only become necessary if you planned to identical signals to the majority of channels. Not something that would happen in music or film.  Uncorrelated summing is also a worst-case scenario because it too assumes identical levels, but of completely different signals, to build at the rate of 3dB/channel count doubling. Reality is usually below that, often by quite a bit. 
  
 Headroom is simply the ratio of a nominal (and somewhat arbitrary) reference level to 0dBFS.  16 bit puts that somewhere around -15 to -20dBFS,  usually.  24 bit lets you fudge that around without taking a noise hit.  We're saying the same thing in different reference frames.  If you look at a gain-structure diagram of an analog mixer you'll see reference levels moving all over the place depending on the specific need for headroom at that point in the design.  That's how I look at 24 bit, you get more room to adjust your gain structure during the original recording because you're not fighting with the noise floor as much. Where I think a lot of location sound guys mess up is they record lower, but don't match the overload points of the entire system, so shouted dialog still gets crunched by a clipped preamp even though it's below 0dBFS.   Just a guess, but easy to confirm.


----------



## lantian

castleofargh said:


> I'll let those who didn't stop using math the day school ended
> 
> 
> 
> ...


 

 Thank you


----------



## spruce music

lantian said:


> Higher sampling rates are simply producing more life like resolution of your music.  Human ear is perfectly capable of hearing the difference between higher sample rates. The so called auditory resolution of human hearing is about 5-7us, a 44.1khz audio has each slice of 20us, 96khz has something like 10us, and only 192khz can offer time resolution of 5us. That would seem perfectly smooth to our auditory systems/brain. Go ahead and correct me If I am wrong on this.




Old myth hard to kill. Time resolution of digitally sampled audio is not the time between samples. It is 1 divided by time between samples x 2pi x number of levels. For 44/16 that's about 56 picoseconds. Dither actually lowers that number further. So this is about 100,000 times smaller than your 5 microseconds.


----------



## gzubeck

This video will answer all your questions concerning audio from 16bit to 24bit audio...Lot of laughs starting at 20 minute mark.


----------



## TheoS53

gzubeck said:


> This video will answer all your questions concerning audio from 16bit to 24bit audio...Lot of laughs starting at 20 minute mark.




 Very good vid, well worth the watch. Thanks for sharing


----------



## StanD

pinnahertz said:


> Ring modulation would be a kind of analog synthesis effect, and as such, I doubt high precision is even desired.  Remember, early FM  synths were 12 bit, then 15 bit.  With today's interest in vintage instruments you probably want that 12 bit grit.


 
 Early synthesizers were analog computers and ring modulators were four quadrant multipliers. I have an ARP 2600 somewhere in my attic.


----------



## StanD

gregorio said:


> 1. I'm not sure I understand what you're asking. In general, 16bit is enough when multiplying two different 16bit numbers (for whatever sound effect). The problem starts to arise when taking that result and multiplying it with something else and gets worse if you then take that result and multiply it with something else again, and then keep repeating that procedure.  Certainly, 48bit or 64bit float is more than enough for any conceivable, realistic number of such processes/procedures and btw, a 48bit fixed virtual mixing environment is not a new thing, it was introduced about 20 years ago and was pretty much ubiquitous by about 15 years ago.


 

 In a strict 16 bit integer system, multiplying two 16 bit numbers would cause an overflow of more than what just a flag could contend with, hence clipping.
 Floating point is not perfect enough for audiophile applications.


----------



## gregorio

pinnahertz said:


> [1] Actually, you can assume uncorrelated. Correlated summing, and the resulting 6dB voltage increase when the number of summed channels doubles, only happens when you sum perfectly identical waveforms, in phase, and at an identical level.   .... Uncorrelated summing is also a worst-case scenario because it too assumes identical levels, but of completely different signals, to build at the rate of 3dB/channel count doubling. Reality is usually below that, often by quite a bit.
> 
> [2] Headroom is simply the ratio of a nominal (and somewhat arbitrary) reference level to 0dBFS.  16 bit puts that somewhere around -15 to -20dBFS,  usually.  24 bit lets you fudge that around without taking a noise hit.  We're saying the same thing in different reference frames.
> 
> [3] Where I think a lot of location sound guys mess up is they record lower, but don't match the overload points of the entire system, so shouted dialog still gets crunched by a clipped preamp even though it's below 0dBFS.   Just a guess, but easy to confirm.


 
  
 1. I did not say that we can't assume uncorrelated summing and therefore we must assume correlated summing. In reality when doubling the track count we don't generally get either exactly +3dB or +6dB but some variable amount.
  
 2. I think we're saying the same thing. Although, the point I was trying to get across is that no matter what the nominal reference level to 0dBFS is set to (on a particular ADC or DAC), it's the same nominal level for both 16 and 24bit. In lay speak, 24bit does provide more loudness, it provides more quietness.
  
 3. OK, this is way off topic but generally they tend not to "record lower". Location sound guys are usually taught to try and get a peak level of -6dBFS, they could/should easily go much lower still and not affect the noise floor. The mic-pres on industry standard location kit (Sound Devices) are very good but at the standard film reference level (-20dBFS = 0VU) there is no benefit in trying to peak at -6dBFS, -12dB would be better.
  
 Quote:


stand said:


> [1] In a strict 16 bit integer system, multiplying two 16 bit numbers would cause an overflow of more than what just a flag could contend with, hence clipping.
> [2] Floating point is not perfect enough for audiophile applications.


 
  
 1. I'm still not sure I understand, I'm not deliberately trying to be obtuse btw. Assuming two full-scale or near full-scale 16bit numbers/signals, then yes multiplying them together would result in clipping but if we take exactly the same two signals (same amplitude/level) in 24bit instead of 16bit, we still get exactly the same clipping. IE. There's no difference/advantage to 24bit vs 16bit. We would also get exactly the same amount of clipping with floating point. The difference with floating point is that instead of simply loosing any data which exceeds full-scale (0dBFS), we can recover it by lowering the output.
  
 2. Not sure if this is sarcasm (aimed at audiophiles)? If so, I wouldn't be surprised if that's what they think, regardless of the fact that there's in effect no difference (between 32bit float and 24bit fixed) and that virtually all PCM mixes are floating point at least in places if not in their entirety (until they get printed to a distribution format).
  
 G


----------



## watchnerd

stand said:


> Floating point is not perfect enough for audiophile applications.


 
  
 Well in that case, you're screwed, because all the DAW software uses floating point.


----------



## castleofargh

floating points make the music flow and sound smooth, but it works only with wav files.


----------



## limpidglitch

stand said:


> I have an ARP 2600 somewhere in my attic.


 
  
 One of those pieces of junk, eh? I'll be kind and pay for shipping if you want to get rid of it


----------



## sterling1

I remember when CDs were introduced the slogan for the medium was perfect sound, forever perfect. Now, if that statement was true, then how can todays hi-res be more perfect? I know that I can't distinguish 16/44 from 24/96 so all of these bigger files seem to do nothing more than fill up library space. I remember when digital was just an idea and the idea was that 16/44 was all that was necessary to assure the best possible outcome.


----------



## StanD

stand said:


> Early synthesizers were analog computers and ring modulators were four quadrant multipliers. I have an ARP 2600 somewhere in my attic.


 
  
  


limpidglitch said:


> One of those pieces of junk, eh? I'll be kind and pay for shipping if you want to get rid of it


 

 I probably have some spare parts like matched pairs of complementary transistors for the exponential voltage to current converter for the VCOs. When I was in college (EE) I was the Warranty Service guy for Arp and Moog in NYC. The first to do this outside of either company. I used to modify synths for all sorts of well known musicians. I met Robert Moog and spoke with him many times. His transistor ladder VCF was a stroke of genius.


----------



## StanD

castleofargh said:


> floating points make the music flow and sound smooth, but it works only with wav files.


 

 You must be smooth operator.


----------



## pinnahertz

stand said:


> Early synthesizers were analog computers and ring modulators were four quadrant multipliers. I have an ARP 2600 somewhere in my attic.


 
 Of course they were.  But we're talking digits here aren't we?  So, the digital realization of a ring modulator.


----------



## pinnahertz

stand said:


> Early synthesizers were analog computers and ring modulators were four quadrant multipliers. I have an ARP 2600 somewhere in my attic.


 
 And I'll pay for shipping on your 2600 plus $200.


----------



## pinnahertz

gregorio said:


> 3. OK, this is way off topic but generally they tend not to "record lower". Location sound guys are usually taught to try and get a peak level of -6dBFS, they could/should easily go much lower still and not affect the noise floor. The mic-pres on industry standard location kit (Sound Devices) are very good but at the standard film reference level (-20dBFS = 0VU) there is no benefit in trying to peak at -6dBFS, -12dB would be better.


 
 I checked with my friend who does location sound, and confirmed the -20dBFS reference, and he meters with PPMs (and true peak).
  
 I know and have owned Sound Devices stuff.  Great stuff.  If they use one of the field recorders it's probably fine, but the stand alone mic pres can easily be miscalibrated relative to the system.  Only when all clipping points are aligned in the system do you get maximum DR. 
  
 Why would you want max peaks at -12dBFS?  Talking Maximum now, not typical.


----------



## pinnahertz

sterling1 said:


> I remember when CDs were introduced the slogan for the medium was perfect sound, forever perfect. Now, if that statement was true, then how can todays hi-res be more perfect? I know that I can't distinguish 16/44 from 24/96 so all of these bigger files seem to do nothing more than fill up library space. I remember when digital was just an idea and the idea was that 16/44 was all that was necessary to assure the best possible outcome.


 
 You know what marketing is, right?  The terms "perfect" and "forever" are marketing terms.
  
 16/44.1 was chosen because it formatted well within a frame of NTSC video, and video recorders were the affordable solution to recording all that digital audio data.  Soundstream was already doing 16/50.  Limits were in data recording an reproducing, and sampling frequencies were considered to be high enough to capture all the necessary audio, but also it wasn't really possible to go higher initially.  44.1, in particular, caused a problem: the anti-aliasing and reconstruction filters were complex (analog), and pretty radical beasts, hard to make well and cheap.  A lot of fingers were pointed at those filters, even spawning an "upgrade" industry.


----------



## StanD

pinnahertz said:


> Of course they were.  But we're talking digits here aren't we?  So, the digital realization of a ring modulator.


 
 I was stating that for the historical record. Kind of like making a reference to vinyl.


----------



## StanD

pinnahertz said:


> And I'll pay for shipping on your 2600 plus $200.


 

 Sorry, I'm keeping that for the historical record, along with my Moog Sonic 6. I should've kept a Mini-Moog .I have a Roland SH-32 Midi/Digital simulation of an analog synth, think it lacks a ring modulator.


----------



## watchnerd

gzubeck said:


> This video will answer all your questions concerning audio from 16bit to 24bit audio...Lot of laughs starting at 20 minute mark.




  
 This video is so comprehensive, and addresses so many of the points that get continually restated here, that we should make a sticky post at the top of the forum for it.


----------



## gzubeck

watchnerd said:


> This video is so comprehensive, and addresses so many of the points that get continually restated here, that we should make a sticky post at the top of the forum for it.


 
 Sounds good to me and here"s the recording engineers website...
  
 http://www.realhd-audio.com/?cat=45
  
 http://www.aixrecords.com/


----------



## Traveller

watchnerd said:


> This video is so comprehensive, and addresses so many of the points that get continually restated here, that we should make a sticky post at the top of the forum for it.


 
 I watched it from beginning to end - excellent presentation.
  
 So my take, as I tried to present in the AptX thread (with little success) is that while I have no problem whatsoever understanding:
 1. "garbage in, garbage out"
 2. We can't hear above ~20K _(and in my case, probably much less than that)_
  
 I was nevertheless very pleased to hear him state hi-res in, hi-res out:
  
 (link goes directly to 29:10)

  
 Which is why I am (still) aiming for gear capable of  ~24b96K playback.
  
 And no, I doubt I could prove it makes a difference via A/B but when I see the "data" above 20KHz then I feel okay having spent a little more for so-called hi-res playback HW...


----------



## watchnerd

traveller said:


> And no, I doubt I could prove it makes a difference via A/B but when I see the "data" above 20KHz then I feel okay having spent a little more for so-called hi-res playback HW...


 
  
 What hi-res playback hardware would that be?


----------



## StanD

watchnerd said:


> What hi-res playback hardware would that be?


 
 This is my rig.


----------



## U-3C

stand said:


> This is my rig.


XD


----------



## Traveller

watchnerd said:


> traveller said:
> 
> 
> > And no, I doubt I could prove it makes a difference via A/B but when I see the "data" above 20KHz then I feel okay having spent a little more for so-called hi-res playback HW...
> ...


 
  
 Rats. It's late here... . So Reset.
 The answer to your question is listed in my signature.
  
 But given that 24b96K DACs are a dime a dozen and even my T5p.2's 5-50KHz specs will not pass muster I'm sure you've managed to make your point with the one question


----------



## sonitus mirus

stand said:


> This is my rig.


 
  
 Sweet!  You should be using an Entreq Olympus Tellus ground to really get the most out of your system.


----------



## Brooko

traveller said:


> Rats. It's late here... . So Reset.
> The answer to your question is listed in my signature.
> 
> But given that 24b96K DACs are a dime a dozen and even my T5p.2's 5-50KHz specs will not pass muster I'm sure you've managed to make your point with the one question


 
  
 Personally I find it refreshing to see someone acknowledges that there is no real audible difference - but acknowledges that they can feel better listening to 24/96.  That's simply humanity at work. Everyone knows I try to be as objective as possible.  I know that aac256 is pretty much transparent - so I use it pretty much exclusively for portable use.  But if I'm home, I use FLAC, and I even sometimes upsample to DSD (even though I know from an audible point of view it isn't making things any better - actually could be worse).  But it lights the little blue light on my iDSD - and sometimes it just sounds better to me with the little light on.  Pure placebo - but if it makes me happy, and doesn't hurt anyone - then why not?
  
 If only more people would do the proper research, and be open to the science, we could demystify it a lot.  Then the claims that one res is audibly better than the other would stop.  And we could be left with personal preference - and a lot less debates 
	

	
	
		
		

		
		
	


	



  
 As per usual though - the quality is in the mastering/recording.  I had a listen to some of Mark's catalogue.  They are indeed excellent quality.  My thanks to gzubeck for posting the video - more music for me to explore.  Fun!


----------



## StanD

stand said:


> This is my rig.


 
  
  


sonitus mirus said:


> Sweet!  You should be using an Entreq Olympus Tellus ground to really get the most out of your system.


 
 I'm trying to work in a Tesla Coil and a Jacob's Ladder but I think that might result in too much RFI.


----------



## Peti

Mark's websites are becoming staples in my audio self-education. Have had the chance to obtain some of his records for my headphone system, and it's really a jaw-meets-floor moment each time I listen to them. When he talks about the "Hi-Rez Mafia", DSD, etc. it's just make sense. I'm thoroughly enjoying these records:
  
 Stravinsky - Firebird Suite & Ravel - Boléro (Enescu Phil. Orchestra, 2001) AIX Recrods
  

  
 A. Dvorak - Symphony No. 6 & 9 (Aslop, Baltimore Symp. Orc.) (Naxos Records, available through Mark's online shop)


----------



## pinnahertz

Mark will be at Axpona this year again.  His demo room last year was, IMO, the best of the show.  Very low on Blue Smoke, very high no just plain great audio.  He's quite passionate about it.  His recordings would be excellent, high-res or not.  Visiting his room cost me money in disc purchases.  And I'm sure it will again this year. 
  
 As to the rest of the show...the Blue Smoke got pretty thick on occasion (I had to walk out of the analog tape session), but it was a good day outside the box anyway.


----------



## Ruben123

peti said:


> Mark's websites are becoming staples in my audio self-education. Have had the chance to obtain some of his records for my headphone system, and it's really a jaw-meets-floor moment each time I listen to them. When he talks about the "Hi-Rez Mafia", DSD, etc. it's just make sense. I'm thoroughly enjoying these records:
> 
> Stravinsky - Firebird Suite & Ravel - Boléro (Enescu Phil. Orchestra, 2001) AIX Recrods
> 
> ...




Perhaps they sound that good because they're mastered in surround instead of stereo? They definitely don't sound that good because they're high resolution.


----------



## sterling1

​Yeah, I know what marketing is. It's what 24/192 and DSD are today. And, no, the word perfect is not exclusive to marketing. I also know what the word fact means. And, for most the fact is 16/44 is all they can appreciate.


----------



## sterling1

Quote: 





pinnahertz said:


> You know what marketing is, right?  The terms "perfect" and "forever" are marketing terms.
> 
> 16/44.1 was chosen because it formatted well within a frame of NTSC video, and video recorders were the affordable solution to recording all that digital audio data.  Soundstream was already doing 16/50.  Limits were in data recording an reproducing, and sampling frequencies were considered to be high enough to capture all the necessary audio, but also it wasn't really possible to go higher initially.  44.1, in particular, caused a problem: the anti-aliasing and reconstruction filters were complex (analog), and pretty radical beasts, hard to make well and cheap.  A lot of fingers were pointed at those filters, even spawning an "upgrade" industry.


 
 Yeah, I know what marketing is. It's what 24/192 and DSD are today. And, I know about the word perfect too, it's not  exclusive to marketing. I also know what the word "fact'" means; and, for most, the fact is 16/44 is all they can digest. BTW, I believe that about 10 years after 16/44's appearance, DAC, as well as ADC were pretty well perfected. In my experience as a producer, it appeared to me that in the early 90's equipment and techniques had progressed enough that a live studio performances as listen to from studio monitors could not be discerned as being different sounding from a DAT recording of same listened to from studio monitors. I still use a pair of Sony PCM-7010F DAT recorder's from about 1992 for their excellent analog to digital and digital to analog conversion. Nothing that I've heard since sounds better, that's to say, nothing seems to retrieve more content. Also, I think Mark is right about all he professes. I believe multi-channel audio is the way audio recording should go. This is easily possible today since so many folks have AVR's or audio/video preamps and processors, as well as universal players to play multi-channel.


----------



## Peti

ruben123 said:


> Perhaps they sound that good because they're mastered in surround instead of stereo? They definitely don't sound that good because they're high resolution.




Yes, yes! these are exquisitely recorded materials. I will downconvert my ripped flac files to redbook and do some abx testing just for the hell of it and I'm not expecting much of a difference. Mind you, I was referring to stereo, as I've never heard the 5.1 version of it.


----------



## pinnahertz

ruben123 said:


> Perhaps they sound that good because they're mastered in surround instead of stereo? They definitely don't sound that good because they're high resolution.



He includes 2 channel stereo mixes and usually two different surround perspectives on most discs. 

There's no definitive evidence to prove that higher bit depth or sampling frequency provides an audible difference, but everyone to a person can hear the difference between two channel stereo and 5.1. 

The question is, how do you want to budget your bits? Inaudible data or audible channels?


----------



## EasyEnemy

great post. thank you. Always been wondering. On many occasion. I tried very hard and I can't hear different between bit depth 16 bit and 24 bit.


----------



## TheoS53

Here are a few more pics of my sample-rate/bit-depth null tests. You can be the judge as to whether or not anything above 44.1/16 is significant:


 192/24 original


 96/24 vs 192/24


 48/24 vs 192/24


 44.1/16 vs 192/24


 Lossy LAME320 MP3 vs 192/24


----------



## sonitus mirus

theos53 said:


> Here are a few more pics of my sample-rate/bit-depth null tests. You can be the judge as to whether or not anything above 44.1/16 is significant:


 
  
 While interesting, I'm not sure how practical it would be to provide the null test results between a lossless and any lossy audio file.  By design, mp3 is going to remove audio data, but the goal is to only remove those sounds that would be masked or otherwise be far too quiet to be heard in a normal listening environment.  A null test alone will give no indication if any difference can actually be heard, despite the fact that something might be heard in the null result.  A bit like having a jackhammer at 10 feet away with a string quartet playing 100 feet away.  The music might be there, but you won't hear any of it when the hammer is operating; so most, if not all, of the music can be discarded.


----------



## TheoS53

sonitus mirus said:


> While interesting, I'm not sure how practical it would be to provide the null test results between a lossless and any lossy audio file.  By design, mp3 is going to remove audio data, but the goal is to only remove those sounds that would be masked or otherwise be far too quiet to be heard in a normal listening environment.  A null test alone will give no indication if any difference can actually be heard, despite the fact that something might be heard in the null result.  A bit like having a jackhammer at 10 feet away with a string quartet playing 100 feet away.  The music might be there, but you won't hear any of it when the hammer is operating; so most, if not all, of the music can be discarded.


 
 That's pretty much the point of including the lossy null test as well. The point of all of this was to showcase the difference between the various sampling-rates and bit-depths. Graphically we can see that there is a significant difference between the lossy and 192/24 file. Yet, the vast majority of people can't tell the difference between lossy and lossless. So, if such a large difference can audibly sound pretty much identical, then it's safe to say that there's practically zero point in opting for a 192/24 FLAC file vs a 44.1/16 FLAC file (given how relatively little difference there is between them vs how big of a difference we could illustrate that the lossy file has)


----------



## sonitus mirus

theos53 said:


> That's pretty much the point of including the lossy null test as well. The point of all of this was to showcase the difference between the various sampling-rates and bit-depths. Graphically we can see that there is a significant difference between the lossy and 192/24 file. Yet, the vast majority of people can't tell the difference between lossy and lossless. So, if such a large difference can audibly sound pretty much identical, then it's safe to say that there's practically zero point in opting for a 192/24 FLAC file vs a 44.1/16 FLAC file (given how relatively little difference there is between them vs how big of a difference we could illustrate that the lossy file has)


 
  
 Ah, I get it now.  Thanks for sharing.


----------



## danadam

theos53 said:


> Graphically we can see that there is a significant difference between the lossy and 192/24 file.


 
 Didn't you mean 44/16?


----------



## TheoS53

danadam said:


> Didn't you mean 44/16?


 
 Not quite sure what you mean there. But all of the spectrogram images I provided were of 96/24 vs 192/24, 48/24 vs 192/24, 44.1/16 vs 192/24, and finally lossy vs 192/24


----------



## danadam

theos53 said:


> Not quite sure what you mean there. But all of the spectrogram images I provided were of 96/24 vs 192/24, 48/24 vs 192/24, 44.1/16 vs 192/24, and finally lossy vs 192/24


 

 Ah, sorry, I somehow misread that as relative differences between the formats as we increase "density", so lossy vs 44/16, 44/16 vs 48/24, etc.


----------



## TheoS53

danadam said:


> Ah, sorry, I somehow misread that as relative differences between the formats as we increase "density", so lossy vs 44/16, 44/16 vs 48/24, etc.


 
 That is one way of doing it. But, I thought it would perhaps be more relevant to test each against the original instead


----------



## Traveller

traveller said:


> ... 1. "garbage in, garbage out"
> 2. We can't hear above ~20K _(and in my case, probably much less than that)_
> I was nevertheless very pleased to hear him state hi-res in, hi-res out:
> (link goes directly to 29:10)
> ...




  
  
_So now all I need is a volunteer bat... 
	

	
	
		
		

		
		
	


	


_


----------



## gregorio

traveller said:


> Which is why I am (still) aiming for gear capable of  ~24b96K playback.


 
  
 You're out of luck, there is no gear capable of 24/96, in fact there's very little gear capable of 16/96 in practice. That's not a problem though as there's hardly any commercial recordings which use more than about 10 bits anyway.
  
 G


----------



## pinnahertz

gregorio said:


> You're out of luck, there is no gear capable of 24/96, in fact there's very little gear capable of 16/96 in practice. That's not a problem though as there's hardly any commercial recordings which use more than about 10 bits anyway.
> 
> G


 
 Precisely. 
  
 There are also no studios capable of 24 bits acoustically (not even close).


----------



## castleofargh

I tried to make a chart once with some range at each step of the audio chain. I gave up after 10mn. I don't know how to show something like that without taking too many shortcuts. if I use some idea of average dynamic range, it's misleading. and how do I decide that value without a lot of data crunching? if I try to get some mini/maxi, it's misleading because it goes all over the place.
 I don't know how to give a fair idea so people stop masturbating over absolutely impossible resolution numbers, without objectively being full of crap myself. some stuff are relatively easy, like the maximum dynamic range of an ADC or a DAC. but when looking at microphones, recording methods, different studio environments... it becomes hard to put a number on things.
  
 edit: much, many, I spik angrish good.


----------



## pinnahertz

Look at it in terms of the "weakest link".  The dynamic range of the entire system is dictated by the part of the system with the least dynamic range.  Playback environment would be it, most likely.


----------



## icebear

castleofargh said:


> I tried to make a chart once with some range at each step of the audio chain. I gave up after 10mn. I don't know how to show something like that without taking too much shortcuts. if I use some idea of average dynamic range, it's misleading and how do I decide that value without a lot of data crunching? if I try to get some mini/maxi, it's misleading because it goes all over the place.
> *I don't know how to give a fair idea so people stop masturbating over absolutely impossible resolution numbers, *without objectively being full of crap myself. some stuff are relatively easy, like the maximum dynamic range of an ADC or a DAC. but when looking at microphones, recording methods, different studio environments... it becomes hard to put a number on things.


 
*ROFL *




 Marketing is a powerful tool, they spew out stuff that has no solid relevance whatsoever, still people like to believe they get something better, something to make their listening experience great again. It's almost as bad as politics these days.
	

	
	
		
		

		
		
	


	




 Come along with scientific reasoning and you are painted as fake news bearer. You don't have a fair chance if the majority wants to believe the other message.


----------



## TheoS53

castleofargh said:


> I tried to make a chart once with some range at each step of the audio chain. I gave up after 10mn. I don't know how to show something like that without taking too much shortcuts. if I use some idea of average dynamic range, it's misleading and how do I decide that value without a lot of data crunching? if I try to get some mini/maxi, it's misleading because it goes all over the place.
> I don't know how to give a fair idea so people stop masturbating over absolutely impossible resolution numbers, without objectively being full of crap myself. some stuff are relatively easy, like the maximum dynamic range of an ADC or a DAC. but when looking at microphones, recording methods, different studio environments... it becomes hard to put a number on things.


 
 I think you've got the right idea going there, but perhaps thinking about it in too much depth. 

 Really, all we need to consider are the limitations of the devices on the consumer's end. Does the player/DAC have the capability to take advantage of the "extra" dynamic range offered by 24-bit? Can the player/amp/headphones reproduce the sounds present at >20kHz?


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## spruce music

Don't you guys realize this thread is passe now. We have 32 bit DAC chips widely available. Need to start a new thread. 24 bit vs 32 bit myths.


----------



## Peti

And 8X DSD!
  
 A brave new world indeed!


----------



## pinnahertz

peti said:


> And 8X DSD!
> 
> A brave new world indeed!


 
"There is a fine line between bravery and stupidity. If you get away with it, you are brave. If you don't, you are stupid." (Francisco Escario)


----------



## castleofargh

peti said:


> And 8X DSD!
> 
> A brave new world indeed!


 
 "take a Soma holiday!"


----------



## Peti

DSD 1024 will be the new trend soon, I predict. I'm just wondering where is the limit here? 32/192 is getting obsolete. This hobby (personal audio) has been getting out of control like the Hi-Fi audio world with 500k speakers, 3000$ cables/meter and 600$ cable lifters along with cable burn-in devices (!). Rant over.


----------



## watchnerd

peti said:


> And 8X DSD!
> 
> A brave new world indeed!


 
  
 Yes, a DAC I've been eyeing does 32bit/384khz and 11.x something MB DSD.
  
 Oh, and Roon can now up-sample boring old Redbook to those formats...
  
 ...and then I could run a DAC with no filter...
  
 ...whee!


----------



## old tech

spruce music said:


> Don't you guys realize this thread is passe now. We have 32 bit DAC chips widely available. Need to start a new thread. 24 bit vs 32 bit myths.


 
  
 Wow, even if 24bit output was possible it would mean a noise floor below the noise of the electronics, ie resistors etc.  What does 32bit offer outside the field of experimental science?


----------



## Traveller

icebear said:


> ...It's almost as bad as politics these days... Come along with scientific reasoning and you are painted as fake news bearer. You don't have a fair chance if the majority wants to believe the other message.


 
_"Alternative facts!!!" _


----------



## watchnerd

old tech said:


> Wow, even if 24bit output was possible it would mean a noise floor below the noise of the electronics, ie resistors etc.  What does 32bit offer outside the field of experimental science?


 
  
 ESP


----------



## StanD

Oh Lord, you guys are either going to trigger a cataclysmic geological event or create some fake news.


----------



## csglinux

old tech said:


> Wow, even if 24bit output was possible it would mean a noise floor below the noise of the electronics, ie resistors etc.  What does 32bit offer outside the field of experimental science?


 
  
 I'm usually the world's biggest cynic regarding hi-res audio. There are way too many charlatans out there getting rich by exploiting the ignorant - and some of these fakers are even selling fake hi-res files!: http://www.head-fi.org/t/648932/bandcamp-promotion-thread/15#post_13283024
  
 However, just for fun, I'm going to do my usual thing of playing Devil's advocate, for the very specific case of 24-bit. I made a comment on an earlier thread of Eke's, which I'll repeat here.
 Imagine you have a great recording with a huge dynamic range, good headphones (obviously, or why would you be on headfi?!) and a good amp with a low noise floor. Now imagine raising the volume (dramatically, if you like) just during the quietest moment of the track. This is the audio dynamic-range equivalent of putting a hi-def picture under a microscope. Your ears still only need to cope with a limited dynamic range - you're simply using your amplifier to lower the gain during the Saturn V rocket launch and then raise it during the string quartet intro. Theoretically, you should then be able to 1) not destroy your hearing and 2) hear the difference between a 24-bit and 16-bit dithered noise floor. This is also relevant to the (rather unhelpful) single THD+N measurements given by amp/DAP manufacturers. I would not assume you won't ever be able to hear those apparently tiny differences, unless you only ever listen to 1 kHz sine waves at 80 dB.
  
 I recently found the following article, which describes this more eloquently than I ever could:
  
 http://www.tonmeister.ca/wordpress/2014/09/15/audio-mythinformation-16-vs-24-bit-recordings/


----------



## watchnerd

stand said:


> Oh Lord, you guys are either going to trigger a cataclysmic geological event or create some fake news.


 
  
 In all seriousness:
  
 I would love to have a DAC that could handle 32 bit or even 64 bit input so that after my DSP process up-converts to 64bit float in order to perform operations I wouldn't have to down-convert to to something S/PDIF can handle before going to the DAC.
  
 Or maybe not....those would be some phat philes.


----------



## StanD

watchnerd said:


> In all seriousness:
> 
> I would love to have a DAC that could handle 32 bit or even 64 bit input so that after my DSP process up-converts to 64bit float in order to perform operations I wouldn't have to down-convert to to something S/PDIF can handle before going to the DAC.
> 
> Or maybe not....those would be some phat philes.


 
 So you want a DAC that does floating point conversions. The marketing department will take delight in that idea.


----------



## spruce music

csglinux said:


> I'm usually the world's biggest cynic regarding hi-res audio. There are way too many charlatans out there getting rich by exploiting the ignorant - and some of these fakers are even selling fake hi-res files!: http://www.head-fi.org/t/648932/bandcamp-promotion-thread/15#post_13283024
> 
> However, just for fun, I'm going to do my usual thing of playing Devil's advocate, for the very specific case of 24-bit. I made a comment on an earlier thread of Eke's, which I'll repeat here.
> Imagine you have a great recording with a huge dynamic range, good headphones (obviously, or why would you be on headfi?!) and a good amp with a low noise floor. Now imagine raising the volume (dramatically, if you like) just during the quietest moment of the track. This is the audio dynamic-range equivalent of putting a hi-def picture under a microscope. Your ears still only need to cope with a limited dynamic range - you're simply using your amplifier to lower the gain during the Saturn V rocket launch and then raise it during the string quartet intro. Theoretically, you should then be able to 1) not destroy your hearing and 2) hear the difference between a 24-bit and 16-bit dithered noise floor. This is also relevant to the (rather unhelpful) single THD+N measurements given by amp/DAP manufacturers. I would not assume you won't ever be able to hear those apparently tiny differences, unless you only ever listen to 1 kHz sine waves at 80 dB.
> ...


 

 What you are describing is gain riding.  Or in other words like compression.  Why not just compress the original 24 bit file judiciously and distribute in 16 bit?
  
 Now sure you could do what you describe and perhaps detect a difference in 24 bit vs 16 bit recordings if they were done impeccably well in some of the quietest venues possible.  Yet, who would listen to music that way?
  
 This is like high sample rates.  There might be some very minor benefit to some on the finest material and finest playback maybe.  Minor, now very minor very subtle very rarely.  Were it a real obvious improvement that some tout the argument against it would have died long ago.  It would be self evident to many people who also would be easily able to demonstrate under blind conditions.  If even 10% could do this it would be in no way contentious.  16 vs 24 bit is at best a supremely subtle improvement in audible terms if at all.  Not to mention that actual noise levels in gear, not to mention listening environments generally does not exceed 20 bit. 
  
 The far more interesting part about greater bit depth and higher sample rates is the large percentage who say the difference is obviously self evident, yet fail over and over and over and over again to show they can hear the difference when they don't know which is which.  Such should be highly cautionary and instead the reverse is true.


----------



## csglinux

spruce music said:


> What you are describing is gain riding.  Or in other words like compression.  Why not just compress the original 24 bit file judiciously and distribute in 16 bit?
> 
> Now sure you could do what you describe and perhaps detect a difference in 24 bit vs 16 bit recordings if they were done impeccably well in some of the quietest venues possible.  Yet, who would listen to music that way?
> 
> ...




I agree with most of what you said. Let me take your argument one step further  Imagine we had even just one person on this planet that could reliably A/B Redbook CD and hi-res audio. Pono or HDTracks, etc., would be all over them. That would be exactly the publicity they'd need to convince the masses. But we haven't seen this. Not even one person.

One little point we disagree on though. No - bit depth is not like high sample rates. You're on the wrong thread  Bit depth relates only to the noise floor.


----------



## Arnav Agharwal

csglinux said:


> One little point we disagree on though. No - bit depth is not like high sample rates. You're on the wrong thread  Bit depth relates only to the noise floor.


 
 As a newbie, I was quite confused because often the phrases, "24-bit" and "resolution", are used together. I later learned that the bit depth is indicative of the dynamic range, and not finer-grained resolution of a static loudness range. Hopefully, we can have a thread on popular Hi-Fi misconceptions (or does one already exist?) where this nugget can be put out


----------



## old tech

csglinux said:


> I'm usually the world's biggest cynic regarding hi-res audio. There are way too many charlatans out there getting rich by exploiting the ignorant - and some of these fakers are even selling fake hi-res files!: http://www.head-fi.org/t/648932/bandcamp-promotion-thread/15#post_13283024
> 
> However, just for fun, I'm going to do my usual thing of playing Devil's advocate, for the very specific case of 24-bit. I made a comment on an earlier thread of Eke's, which I'll repeat here.
> Imagine you have a great recording with a huge dynamic range, good headphones (obviously, or why would you be on headfi?!) and a good amp with a low noise floor. Now imagine raising the volume (dramatically, if you like) just during the quietest moment of the track. This is the audio dynamic-range equivalent of putting a hi-def picture under a microscope. Your ears still only need to cope with a limited dynamic range - you're simply using your amplifier to lower the gain during the Saturn V rocket launch and then raise it during the string quartet intro. Theoretically, you should then be able to 1) not destroy your hearing and 2) hear the difference between a 24-bit and 16-bit dithered noise floor. This is also relevant to the (rather unhelpful) single THD+N measurements given by amp/DAP manufacturers. I would not assume you won't ever be able to hear those apparently tiny differences, unless you only ever listen to 1 kHz sine waves at 80 dB.
> ...


 
 Yes, as others have commented, it is possible to hear a [noise] difference between 16bits and 24bits (though it is unlikely it is 24bits, more like 20) under contrived conditions.
  
 I know I can detect a difference between a 16bit file and 24bits if I turn up the volume very loud on a silent piece of track, but even then I'm not sure whether I would hear any difference if dither was not applied to the recording.  The point is that this has zero effect on listening to the actual music content.  The slightly audible hiss of 16bits is so low it is unlikely to mask anything that is musical.


----------



## old tech

arnav agharwal said:


> As a newbie, I was quite confused because often the phrases, "24-bit" and "resolution", are used together. I later learned that the bit depth is indicative of the dynamic range, and not finer-grained resolution of a static loudness range. Hopefully, we can have a thread on popular Hi-Fi misconceptions (or does one already exist?) where this nugget can be put out


 
 This might help explain how the concept of resolution applies (or does not apply) to digital audio.
  
 http://productionadvice.co.uk/no-stair-steps-in-digital-audio/


----------



## csglinux

arnav agharwal said:


> As a newbie, I was quite confused because often the phrases, "24-bit" and "resolution", are used together. I later learned that the bit depth is indicative of the dynamic range, and not finer-grained resolution of a static loudness range. Hopefully, we can have a thread on popular Hi-Fi misconceptions (or does one already exist?) where this nugget can be put out




Check out Monty's blogs and videos on xiph.org. They're quite fun as well as educational. This is a good place to start:

http://xiph.org/~xiphmont/demo/neil-young.html


----------



## watchnerd

stand said:


> So you want a DAC that does floating point conversions. The marketing department will take delight in that idea.


 
  
 Floating point is the new multi-bit.


----------



## StanD

watchnerd said:


> Floating point is the new multi-bit.


 

 We should float that past some audiophiles and watch the excitement.


----------



## RRod

arnav agharwal said:


> As a newbie, I was quite confused because often the phrases, "24-bit" and "resolution", are used together. I later learned that the bit depth is indicative of the dynamic range, and not finer-grained resolution of a static loudness range. Hopefully, we can have a thread on popular Hi-Fi misconceptions (or does one already exist?) where this nugget can be put out


 

 Actually they are one in the same. Say we want to pad 16 bits with 0s to make a 24-bit file, and that B00...B15 are our 16 bits. Then you can take two approaches**:
 1) 00000000 B00...B15
 2) B00...B15 00000000
  
 In the first approach, the extra 0s indicate the extended dynamic range of 24 bits that the 16 bits aren't using. In the second approach, the extra 0s "fill-in" the gaps between the 16-bit values. By convention a conversion from 16 to 24 bits uses the second approach so that the max level is consistent between the two files.
  
 **I'm fudging this a bit because PCM integers are technically stored as two's complement.


----------



## gregorio

csglinux said:


> [1] Now imagine raising the volume (dramatically, if you like) just during the quietest moment of the track. ... Theoretically, you should then be able to 1) not destroy your hearing and 2) hear the difference between a 24-bit and 16-bit dithered noise floor.
> 
> [2] I recently found the following article, which describes this more eloquently than I ever could: http://www.tonmeister.ca/wordpress/2014/09/15/audio-mythinformation-16-vs-24-bit-recordings/


 
  
 1. This was discussed, albeit briefly, in the first couple of pages of this thread. Let's take an example, an extreme one, a recording with a 72dB dynamic range, this is extreme because hardly any commercial recordings have a dynamic range of more than 60dB. For a 72dB dynamic range we need about 12bits of data/resolution. Now let's say we whack the volume up during the quietest parts by 20dB so that we can hear the digital noise floor (and differentiate 24 from 16bit): As Spruce Music says, what you've effectively done is manual compression, you've raised the noise floor by 20dB while peak volume remains the same (because you lower the volume again during the loud parts). The 72dB dynamic range of our recording is now 52dB, for which we only need about 9bits!
  
 2. I see this kind of thing quite often, even in some published papers. Most of the information provided in that article is based on 16bit with TPDF dither, the result of this dither is white noise (as the article stated) and under certain circumstances the potential problems described might exist, although I would dispute both the magnitude of these potential problems and how often they would be encountered in practice. However, my main problem is with the starting premise: In the real world, how many commercial 16bit recordings actually have TPDF dither? If comparing 24bit vs 16bit, the answer is pretty much none at all! It has NEVER been standard practice to apply TPDF dither to a 24bit master/mix for 16bit distribution, always some form of noise shaped dither. This brings us on to the appendix, where the author admits his potential problem with TPDF dither is eradicated by noise shaped dither but introduces a new potential problem in the form of possible IMD, caused by the shaped dither noise energy up around the >16kHz range. However, this again makes no sense when comparing 24bit to 16bit. In practise, we don't really encounter 24/44.1 files, 24bit consumer music files are typically 96kHz or 192kHz, which provide a significantly higher frequency response. If a replay system can't handle the audible range and is generating IMD products from frequency content at say 17kHz, what IMD products is it going to generate from frequency content at say 30kHz?
  
 Quote:


> Originally Posted by *castleofargh* /img/forum/go_quote.gif
> 
> some stuff are relatively easy, like the maximum dynamic range of an ADC or a DAC. but when looking at microphones, recording methods, different studio environments... it becomes hard to put a number on things.


 
  
 As Pinnahertz stated, dynamic range is a "weakest link" scenario, it's defined by the point in the whole recording/playback chain which has the smallest dynamic range. The peak/high point of that dynamic range is probably defined by the consumer amp speakers/cans part of the chain but let's generously assume high quality consumer playback equipment and define the peak part of our dynamic range equation as 120dBSPL (dictated by the human hearing part of the chain). For most music it's going to be a substantially lower figure than this but again, let's take an extreme case, say a symphony orch which could commonly have sustained peaks up around 105dB but may contain the odd transient up to nearly 120dBSPL, which is very loud but just about bearable provided those transients are infrequent and very short duration. Let's also take another extreme, very well isolating IEMs/headphones in a very quiet listening environment, and therefore a listening environment of say 20dBSPL. Using these two extreme circumstances to define our boundaries, we have a potential dynamic range of 100dB, which could in theory mean that we would be able to differentiate between 16 and 24bit. However, this is in theory, not in practice because there are a few serious holes in this scenario:
  
 1. I chose a symphony orchestra as an example because logic suggests this produces the largest dynamic range. Typically from just one, two or a small handful of musicians playing quietly at one extreme, to 80-120 musicians simultaneously playing as loud as they can at the other. A large top class studio should have a noise floor around 30dBSPL, which btw already reduces our potential dynamic range from 100dB to 90dB but it doesn't stop there: Put say 90 living, breathing, moving musicians in that studio and it's noise floor is no longer even close to 30dBSPL. Our potential dynamic range is now down to about 60dB for average peak levels and 75dB for the occasional transient. And obviously, if we're talking about recording a live performance then we typically have around 1-4 thousand living, breathing, moving audience members to add to the noise floor and probably around another 10dB or so reduction in our potential dynamic range.
  
 2. As far as dynamic range is concerned, our hearing operates on a similar fundamental principle as our sight. We have a wide visual dynamic range for brightness, from a bright sunny day to a fairly dark room. However, we're all well aware that this is a bit of a clever trick, our eyes actually have a much smaller dynamic range than the limits suggest but get around this fact by effectively making that limited dynamic range window moveable. We can see well in a darkened room but if we leave that room and walk into bright sunlight, it's dazzling to the point of painful. The upper limit of our visual dynamic range is significantly less than our theoretical/usual limit, until our eyes have had time to adjust their dynamic range window and once they have, if we re-enter the darkened room we can no longer see well, it's pitch black, again until our eyes readjust it's dynamic range window to a lower/darker level. The same happens with our hearing; if we really have achieved a listening environment of just 20dBSPL, our upper limit is no longer 105dBSPL (with occasional 120dB transients), it's significantly/proportionately lower. The evidence I've seen suggests our ears' moveable dynamic range window is somewhere between 30dB and 60dB.
  
 3. Closely related to my response #2 to csglinux above, we have to be careful about the quoted response figures, how do these figures actually apply to the real world? Yes, a symphony orch can produce transient peaks up to 120dB, yes, a harmon-muted trumpet can produce measurable volume at >80kHz, etc. But, what can be produced and what is actually heard are two different things. Not withstanding the fact that I don't know of any symphonies which require a trumpet to use a harmon mute or that we can't hear 80kHz, just because we can record and measure 80kHz content with a mic placed a few inches from the trumpet's bell is meaningless in terms of an accurate or realistic recording, unless you're accustomed to sitting just a few inches in front of the trumpet during a symphony performance?! In practice we're going to be at least about 30ft away and probably double that in a "prime" seat, add to this a dozen or so living absorption panels in the way (say three or so desks of violas and a few rows of audience) and see how much 80kHz trumpet content you can record now! Even worse with say a french horn, where between what a french horn actually produces and what you (in the audience) hear is maybe: 100ft of air, two percussion sections, a wall (!), 4 or 5 desks of violins and a few rows of audience! Same with the volume figures for the orchestra, where are we measuring say 120dBSPL transient peaks? From just in front of the conductor or from a few rows back in the audience? If it's the latter, then our peaks (and therefore dynamic range) are probably around 10dB or more lower.
  
 Taking the above three practicalities into account: 1. The dynamic range limit of the 16bit format is pretty much the least of our dynamic range bottlenecks. Even in extreme circumstances it's no more than the second least, with still several other bottlenecks of more significance actually defining the practical dynamic range. 2. Dynamic range is effectively an artistic decision, which in the case of acoustic performance genres is defined not by the sound the instrument/s actually produce but by where we choose to position the listener and therefore where we place the mics relative to the orch/sound sources. 3. I don't believe it's entirely coincidental that the dynamic range of the most dynamic music recordings is generally no more than about 60dB.
  
 G


----------



## TheoS53

Could someone please clarify this for me. I've seen a few posts now that say that various recordings (most of them) have such limited dynamic range, etc. Now, this confuses me, as I thought dynamic range was the difference between the loudest and quietest sounds.

 If we look at this spectrogram image I posted a little while back, notice the legend for the colours on the right hand side. That scale has a dB range of 100. So if we look at the actual spectrogram image, can we not indeed see all those colours..suggesting that the track does cover the full 100dB dynamic range?


----------



## RRod

theos53 said:


> Could someone please clarify this for me. I've seen a few posts now that say that various recordings (most of them) have such limited dynamic range, etc. Now, this confuses me, as I thought dynamic range was the difference between the loudest and quietest sounds.
> 
> If we look at this spectrogram image I posted a little while back, notice the legend for the colours on the right hand side. That scale has a dB range of 100. So if we look at the actual spectrogram image, can we not indeed see all those colours..suggesting that the track does cover the full 100dB dynamic range?


 
  
 Do this: make a 2kHz square wave that peaks at, say, -80dBFS. Now note how high you have to set your volume before you can hear it, call this V0. If a song is loud enough that it forces you to set your volume *lower* than V0, then arguably that song isn't using 80dB of dynamic range *in your listening room*. I add that last part because how soft a signal you can hear depends both on your equipment and your listening room: I'd hope we can all agree you aren't going to appreciate signals at -80dBFS on a loud bus.


----------



## VNandor

theos53 said:


> Could someone please clarify this for me. I've seen a few posts now that say that various recordings (most of them) have such limited dynamic range, etc. Now, this confuses me, as I thought dynamic range was the difference between the loudest and quietest sounds.


 
 Technically you are right, however when people say dynamic range, they often don't actually mean dynamic range.
 See here: http://www.head-fi.org/t/834222/understanding-the-parameters-in-the-dynamic-range-database
  
  


theos53 said:


> If we look at this spectrogram image I posted a little while back, notice the legend for the colours on the right hand side. That scale has a dB range of 100. So if we look at the actual spectrogram image, can we not indeed see all those colours..suggesting that the track does cover the full 100dB dynamic range?


 
 I'm not sure about the graph but I think this is not how it works. If you played a 5kHz sine wave that peaks at -20dBFS together with a 10 kHz sine wave that peaks at -100dBFS you would have the red and blue colour represented; the red would be a flat line at 5kHz and the blue would be a flat line at 10kHz. However it wouldn't have any dynamic range because it's just two sines playing at a constant (not changing) amplitude.


----------



## TheoS53

vnandor said:


> Technically you are right, however when people say dynamic range, they often don't actually mean dynamic range.
> See here: http://www.head-fi.org/t/834222/understanding-the-parameters-in-the-dynamic-range-database
> 
> 
> I'm not sure about the graph but I think this is not how it works. If you played a 5kHz sine wave that peaks at -20dBFS together with a 10 kHz sine wave that peaks at -100dBFS you would have the red and blue colour represented; the red would be a flat line at 5kHz and the blue would be a flat line at 10kHz. However it wouldn't have any dynamic range because it's just two sines playing at a constant (not changing) amplitude.


 
 I generated those 2 sine waves and combined them, here's the image :


----------



## VNandor

It didn't work as I expected, I stand corrected. It still shows a bit of variation of colour though.


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## TheoS53

vnandor said:


> It didn't work as I expected, I stand corrected.


 
 I made a small mistake. When I generated the 10kHz tone, I entered -100dB, but for some reason the generator can only go as low as -90 (so it defaulted back to -20). So I generated the new 10kHz tone at -90dB and combined it with the -20dB 5kHz tone Pic updated. But yeah, still looks a bit different than anticipated


----------



## VNandor

theos53 said:


> I made a small mistake. When I generated the 10kHz tone, I entered -100dB, but for some reason the generator can only go as low as -90 (so it defaulted back to -20). So I generated the new 10kHz tone at -90dB and combined it with the -20dB 5kHz tone Pic updated. But yeah, still looks a bit different than anticipated


 
 I got home so I could take a closer look at the graph. Looking at the 5kHz sine wave, I can see that in the middle it's red, then it gets progressively colder and colder down to purple with the green colour being dominant. Point is, looking at a graph like this, reading the dynamic range is far from being trivial.
  
 As an aside I think the program should plot an "infinitely" thin red line. If you could increase the resolution of the plotting, it might look closer to what I expected.


----------



## TheoS53

vnandor said:


> I got home so I could take a closer look at the graph. Looking at the 5kHz sine wave, I can see that in the middle it's red, then it gets progressively colder and colder down to purple with the green colour being dominant. Point is, looking at a graph like this, reading the dynamic range is far from being trivial.
> 
> As an aside I think the program should plot an "infinitely" thin red line. If you could increase the resolution of the plotting, it might look closer to what I expected.


 
 I did another test. 5kHz at -3dB, and this is what I got. It seems the louder the signal, the more it bleeds into other frequencies. This might explain why the 10kHz signal was just a blue line, because it's so quiet that the sound decays much faster


----------



## TheoS53

I find these graphs pretty cool. This is a CEA-2010 Burst at 12kHz -3dB:


----------



## gregorio

theos53 said:


> That scale has a dB range of 100. So if we look at the actual spectrogram image, can we not indeed see all those colours..suggesting that the track does cover the full 100dB dynamic range?


 
  
 No. The spectogram is giving you a breakdown of the energy contained in the frequencies which comprise a sound, not the total energy of the sound and therefore not the dynamic range. For example, at exactly 2:00 we have a sound whose total energy we can only guess to be about -20dB. The spectogram is showing us the breakdown of this sound, that the vast majority of the energy in this sound is in the low frequency band, roughly -25dB in the 0-1kHz band, about -40dB in the 1-5kHz band, about -80dB in the 5-15kHz band and about -110dB in the 15-30kHz band. Add all these energy levels in the different frequencies together and we'd have the total energy of this sound, which we can then compare with the quietest sound in the track which is at the very end. Here we can see the highest level is green/light blue, also in the low freqs, a total energy value of probably somewhere around 70-75dB, 50-55dB lower than the max level of about -20dB. At a guess then, the dynamic range of this recording is (very roughly) about 50-55dB, which would require about 9bits!
  
 G


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## pinnahertz

theos53 said:


> I did another test. 5kHz at -3dB, and this is what I got. It seems the louder the signal, the more it bleeds into other frequencies_*. This might explain why the 10kHz signal was just a blue line, because it's so quiet that the sound decays much faster*_


 
  


vnandor said:


> I got home so I could take a closer look at the graph. Looking at the 5kHz sine wave, I can see that in the middle it's red, then it gets progressively colder and colder down to purple with the green colour being dominant. Point is, looking at a graph like this, _*reading the dynamic range is far from being trivial.*_
> 
> As an aside I think the program should plot an "infinitely" thin red line. If you could increase the resolution of the plotting, it might look closer to what I expected.


 
  


theos53 said:


> I find these graphs pretty cool.


 
 I've highlighted a few points to respond to.
  
 You guys are having both fascination and difficulties with the spectrogram as it applies to DR because it's actually the wrong tool.  That display is actually compressing a 3D data block into a 2D/flat display, and giving you too much information presented in a confusing way.  
  
 The observation that quiet sounds "decay much faster" is an anomaly of the analysis only, not true in reality.  
  
 Yes, the graphs are cool, and yes, and reading the dynamic range from them is non-trivial.  Relating a spectrogram to any actual audible characteristic is extremely difficult.


----------



## spruce music

theos53 said:


> I did another test. 5kHz at -3dB, and this is what I got. It seems the louder the signal, the more it bleeds into other frequencies. This might explain why the 10kHz signal was just a blue line, because it's so quiet that the sound decays much faster


 

 Spectrograms are done using FFTs.  Usually the FFT uses a small number of bins or filter banks.  This means a loud signal will bleed into adjacent bins and show up on the graph as a wide line instead of a thin line in one frequency bin.
  
 Using different software and colors here is an example of a 5 khz sine wave.  At -120 the background goes to gray.
 This one is with the spectrogram set to 32 k FFT bins.
  

  
 This one is with the spectrogram using 256 FFT bins.
  

  
 You can see with only 256 bins the 5 khz signal actually bleeds over slightly at all frequencies.  It is just an artifact of windowing in an FFT.  Has nothing to do with decay.
  
 The above is from Audacity which is free.  It has a spectrogram view as well as the default waveform view. In preferences you can adjust the size of the FFT bins, the dynamic range over which it functions and the type of windowing to use.


----------



## VNandor

spruce music said:


>


 

 Great, that explained a lot to me.


----------



## gregorio

A couple of points maybe worth expanding upon:
  
@VNandor Spectral analysis is a fantastically useful tool but there are some inherent weaknesses/limitations. By increasing the FFT bin size we increase frequency resolution/accuracy, as Spruce Music explained/demonstrated. However, as we increase the number of bins and frequency resolution (the y-axis) we decrease the timing accuracy (the x-axis). A partial solution to this is to choose a moderate FFT window size but overlap them (vertically), which provides better frequency resolution. At the same time, one can also overlap the windows horizontally, giving greater time resolution/accuracy. There is however still a price to pay, in terms of the required amount of processing. Occasionally I need to really dig down deep in order to treat a specific freq/harmonic in a fast, dense piece of music/audio. I might use  x32 frequency overlap and x32 time overlap, I don't actually know the number of bins as I've found the best results in my program is if I set the number of bins to "auto" and let the program decide. I get relatively accurate frequency and time resolution but it takes my 12 core Mac Pro about 20secs to calculate/render about 2 seconds worth of audio. Using the default settings I can render several minutes of audio in just a second or two.
  
@TheoS53 As Pinnahertz stated, this is the wrong tool for measuring dynamic range and as I stated, the best we can do with a spectogram is make a very rough guess at dynamic range but we still need to understand/interpret what we're looking at, as well as understand what "dynamic range" actually means (which is not easy as it's a rather loosely defined term). For example, look again at the very end of the recording and let's say it continues for another few seconds, to say 2:25, all the time dying away so at say 2:23 there's no more green/light blue, only dark blue/purple, which is say -110dB. Would this recording now have a dynamic range of 90dB (-20dB at it's highest to -110dB at it's lowest)? Possibly but it's very unlikely, far more likely is that the actual noise floor is at about -60 or -70dB and all we're seeing is the mix engineer fading this noise floor to digital silence (black), not at all an uncommon practice. This transition from the noise floor to digital silence is NOT part of our dynamic range (!), because I've used the term "dynamic range" to mean the ratio of peak value to noise floor (in this case, say -20dB to -70dB, IE. 50dB) and therefore what happens beneath the noise floor cannot be part of the dynamic range calculation. If you think about it, this raises some interesting questions.
  
 G


----------



## Arpiben

gregorio said:


> A couple of points maybe worth expanding upon:
> 
> @VNandor
> Occasionally I need to really dig down deep in order to treat a specific freq/harmonic in a fast, dense piece of music/audio. I might use  x32 frequency overlap and x32 time overlap, I don't actually know the number of bins as I've found the best results in my program is if I set the number of bins to "auto" and let the program decide. I get relatively accurate frequency and time resolution but it takes my 12 core Mac Pro abo
> ...



In the occasional cases mentioned above, just by curiosity,which frequency & time resolution were you seeking or estimating?

Edited/Added:

Recent Real timing Spectrum Analzers (RSA) are using Discret Fourrier Transform (FFT methods) taking advantage of:

wide band ADCs (16bits/ 500MSps (max sampling rate per second) 
paralel operations sampling & FFT processing with FPGA/ASICS 

 With such you can catch *live* events/transients as short as 1us with a 100% Probability Of Interception POI. Frequency resolution is within the mHz order.Note that 1us is the Spectogram time resolution, time domain resolution is divided by the window bin numbers ( typ. 1024).

Dealing with Audio domain, I have no good idea about what resolution is needed or improved by the use of external software computation hence my above question. 

RSA samples the analog signal with a high rate and then use decimation in order to increase time domain resolution. RSA overlaps up to a certain percentage ( 67%) for avoiding missing events (cf table)

My concern/curiosity is dealing with an audio file, let's say 44.1kHz/16bits (22.6us sample time resolution) :

 FFT (512) -> 86Hz/bin 11.7ms/FFT 
 FFT (32)-> 1.3kHz/bin 0.7ms/FFT
 FFT(16384) ->2.7Hz/bin 0.4s/FFT

 In a real time spectogram (computer) even with overlapping we should be able to detect only signals with duration greater than 1ms (FFT 32) with a 100% POI.
In non real time spectogram analysis, I am assuming that the software is first reconstructing the signal previous to further FFT processing hence improving above resolution.

Anyhow thanks for having aroused my interest with your post. BTW If you have good knowledge of your specific freq/harmonic's shape a frequency mask trigger may happen to be more efficient for detection.

Hereunder as example, real time parameters of a lab spectrum analyzer


----------



## gregorio

arpiben said:


> In the occasional cases mentioned above, just by curiosity,which frequency & time resolution were you seeking or estimating?


 
  
 A resolution which takes a reasonable amount of time to calculate/render and provides the greatest likelihood of me identifying, selecting and processing the actual frequency/ies and time position/duration.
  
 G


----------



## Arpiben

gregorio said:


> A resolution which takes a reasonable amount of time to calculate/render and provides the greatest likelihood of me identifying, selecting and processing the actual frequency/ies and time position/duration.
> 
> G


 

  I was editing my previous post for better understanding when you replied since my purpose was only to have a rough idea of resolution needed in Audio applications.
 Are the 1Hz / 100us with 100% Probability Of Interception enough in most cases ?
 Thanks.


----------



## Deltrus

gregorio said:


> A resolution which takes a reasonable amount of time to calculate/render and provides the greatest likelihood of me identifying, selecting and processing the actual frequency/ies and time position/duration.
> 
> G


 
 I just found this thread and read the first couple pages. I saw it had nearly 250 pages and I had to see the ending - looks like even after 8, WHAT, EIGHT YEARS?! My God, you are one dedicated fellow, may I commend you for sticking it out that long!

 I guess I am not surprised there still is debate, but I just replaced my dying Schiit MODI (right RCA jack causes a lot of static/electrical noise to be very present in the right side) with a Schiit MODI 2, as much as I wanted the Jotunheim, I really can't afford even this right now. Then I got the Schiit Wyrd because of electrical noise from my graphics card working its butt off in games, Adobe Lightroom CC, and other programs. Yet, I still hear stuff, but typically only through my speakers, as they are bi-amped Presonus Eris E5, being fed XLR by the KRK 10S subwoofer, which is getting balanced 1/4" from the RCA outs on the Asgard 2. Did the Wyrd do anything? Sort of? Maybe? That's for another thread. With my Sennheiser HD598s, though, I hear nothing. Thankfully!

 Anyways, onto the reason I was reading this thread and ultimately wanted to comment...

 I decided, why not, let's try some 24 bit audio files, I've heard good things, so hey. I came to the realization that my .wav files I had specifically ripped might as well have been 320mp3 files, because they are 16 bit 44.1khz files. Here I am, looking at 7digital, wondering if the files are, I guess you could say, true 24 bit, because they all, almost sound THE EXACT SAME as my 16 bit counterparts.

 Is there something I should do? I'd love to see the supposed increased dynamic range and all, but your write-up makes a pretty damned killer point against even considering a need for 24 bit depth.

 Looking forward to what you have to say, even if it's just: "you didn't fall prey to placebo-man, good for you!"
  
 Cheers for such an amazing post and sticking around so long!


----------



## NeoG

deltrus said:


> I decided, why not, let's try some 24 bit audio files, I've heard good things, so hey. I came to the realization that my .wav files I had specifically ripped might as well have been 320mp3 files, because they are 16 bit 44.1khz files. Here I am, looking at 7digital, wondering if the files are, I guess you could say, true 24 bit, because they all, almost sound THE EXACT SAME as my 16 bit counterparts.


 
  
 I suppose the takeaway is that, all else being equal - the 16-bit and 24-bit files should sound the same to you when consuming them as intended. It's actually a good sign that your 24-bit playback path is working correctly, unnecessary as it may be.
  
 The "all else being equal part" is tricky, because you can have different masters, problems with 24-bit/hi res output modes in DACs etc which can cause audible issues in one or the other.


----------



## gregorio

deltrus said:


> [1] Yet, I still hear stuff, but typically only through my speakers, as they are bi-amped Presonus Eris E5, being fed XLR by the KRK 10S subwoofer, which is getting balanced 1/4" from the RCA outs on the Asgard 2. Did the Wyrd do anything? Sort of? Maybe? That's for another thread. With my Sennheiser HD598s, though, I hear nothing. Thankfully!
> 
> [2] I decided, why not, let's try some 24 bit audio files, I've heard good things, so hey.


 
  
 1. Something does seem odd about your description which could be the cause of you "still hearing stuff". It might not be odd in practice but I'd have to research the various bits of kit you're describing, what exactly their inputs and outputs are doing. How are you getting a balanced 1/4" input (to your KRK) from RCA outs? RCA outputs are unbalanced.
  
 2. Yes, that's the problem and that's why I started this thread. There's been so much marketing around consumer 24bit and so many people who've heard a significant difference/improvement with 24bit that the audiophile community is by and large thoroughly convinced of the benefit of 24bit. This creates an almost inescapable vicious circle of marketing and testimonials, which drags even more people into the 24bit myth, which creates even more positive testimonials, etc. The problem is of course that the "significant improvement" being heard is caused by either a type of placebo effect or different master versions, as stated by NeoG and many others throughout this thread. And bizarrely, as exemplified by this current thread appreciating MQA, many audiophiles simply don't care that they're comparing different master versions rather than what they're claiming to compare, go figure!!
  
 G


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## DavidPiska

Thank you Gregorio


----------



## globobock

This might be already discussed, but I just fell upon this link and it is interesting what happened when you talk directly to engineers (Sony in this case) - they are basically repeating what gregorio has been telling us since page 1, in that there is no audible difference on hi-res audio IF the source is good. there is even no audible diff between 248bit and flac (1400bit)...
  
http://www.avhub.com.au/news/sound-image/did-sony8217s-chief-sound-architect-just-tell-me-they-can8217t-hear-or-measure-the-difference-between-upsampled-248k-files-and-true-high-res-audio-wow-437891
  


> _Note, importantly, that Koji Nageno had said some rare files could be differentiated: “...Some risk maybe for deep bit-depth and high frequency sound — in that kind of case it has some possibility to make a loss…” But “normal music, sound wave check and listening check, there's no difference.” Hence their use of the term “near high-res audio” — nearly, but not always. There are exceptions._


----------



## dimedrol

from my humble experience, if you don't like the quality of a recording, it is never because it's lacking in the sample rate. It's either the recording is bad or simply not to your liking or your hardware is adding significant distortion. Increasing sample rate does nothing to those two causes. Now, you can theoretically specifically record music to benefit from the increased sample rate (ie make it so it would sound worse in 16bit), but I've never heard of such recordings, from my experience sound engineers work so that their music sounds as good as possible on a broad selection of audio systems.


----------



## gregorio

globobock said:


> http://www.avhub.com.au/news/sound-image/did-sony8217s-chief-sound-architect-just-tell-me-they-can8217t-hear-or-measure-the-difference-between-upsampled-248k-files-and-true-high-res-audio-wow-437891


 
  
 Thanks for the link. The article is funny, it's like "Newsflash! The chief engineer at Boeing states that pigs can't fly.".  The only difference in this case is that Boeing hasn't spent years and millions of dollars manufacturing and marketing flying pigs. For this reason, that Sony engineer is probably either already unemployed or has been locked in his lab and will not be allowed to speak publicly ever again!
  


dimedrol said:


> Now, you can theoretically specifically record music to benefit from the increased sample rate (ie make it so it would sound worse in 16bit), but I've never heard of such recordings, from my experience sound engineers work so that their music sounds as good as possible on a broad selection of audio systems.


 
  
 Not so much actually record but master differently, certainly. In fact, way back earlier in this very thread I remember an exchange I had with a representative of a distributor, I can't remember which distributor but I think it might have been HD Tracks or Linn, where they stated they do (or did at that point in time) routinely make their 16bit versions poorer quality! Of course, that's not exactly what they stated, they stated that they make their recordings as good as possible and that in the case of their 16bit versions they add a significant amount of compression. This is because, they say, many of their clients then lossy encode their 16bit version for use in portable devices where more audio compression would sound better. Now that is a reasonable/acceptable response but when I basically said fair enough but instead of distributing a more highly compressed 16bit version and a far more expensive "HiDef" version, why not just sell a second 16bit version but without the additional compression (which would be audibly the same as the HD version). No response, the silence was deafening!
  
 I likewise don't know engineers who don't want to make their recordings sound "as good as possible" but if a client wants a HD version and a more compressed 16bit version, it's my job to give the client what they ask for and I've done exactly this on a number of occasions. I'm not "selling my soul" because there are numerous situations where a more compressed version will sound better. However, decent quality listening environments/equipment is not one of those situations and therefore it's entirely possible to use one of the 16bit versions I've created to demonstrate that HD is better than 16bit. In fact, it was encountering a client of mine doing exactly this which prompted me to start this thread!
  
 G


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## TheTrace

I just wanna say thank @gregoio for creating this thead and everyone that contributed to this info. Lol


----------



## Petyot

gregorio said:


> Thanks for the link. The article is funny, it's like "*Newsflash! The chief engineer at Boeing states that pigs can't fly*.".  The only difference in this case is that Boeing hasn't spent years and millions of dollars manufacturing and marketing flying pigs. For this reason, that Sony engineer is probably either already unemployed or has been locked in his lab and will not be allowed to speak publicly ever again!


 
  
  
 Coffee all over my keyboard!! This is funny


----------



## globobock

gregorio said:


> Thanks for the link. The article is funny, it's like "Newsflash! The chief engineer at Boeing states that pigs can't fly.".  The only difference in this case is that Boeing hasn't spent years and millions of dollars manufacturing and marketing flying pigs. For this reason, that Sony engineer is probably either already unemployed or has been locked in his lab and will not be allowed to speak publicly ever again!


 
 Glad I could contribute . The hilarious thing is, that the articles author didn't really know what to do with this revelation, since his original purpose was inquiring about Sonys claim of lossless Bluetooth transmittion of HiRes audio.
  
 In defence of the Sony Chief Engineer - if you are an engineer at heart and your day-to-day work is based on hard facts and telling things what they are, you can keep up your marketing face only for so long... And maybe this was something he has carried with him for quite some time already and it needed to be spilled out. Heck who knows, maybe he is a member of this forum 
  
 With his creds, I don't think he has to worry much about unemployment though - that think about getting locked in his lab is something more realistic.
 Anyone attended this?
http://www.aes.org/events/140/presenters/?ID=4417
  
 


> AES Paris 2016 Presenter or Author
> 
> 
> 
> ...


----------



## globobock

gregorio said:


> Not so much actually record but master differently, certainly. In fact, way back earlier in this very thread I remember an exchange I had with a representative of a distributor, I can't remember which distributor but I think it might have been HD Tracks or Linn, where they stated they do (or did at that point in time) routinely make their 16bit versions poorer quality! Of course, that's not exactly what they stated, they stated that they make their recordings as good as possible and that in the case of their 16bit versions they add a significant amount of compression. This is because, they say, many of their clients then lossy encode their 16bit version for use in portable devices where more audio compression would sound better. Now that is a reasonable/acceptable response but when I basically said fair enough but instead of distributing a more highly compressed 16bit version and a far more expensive "HiDef" version, why not just sell a second 16bit version but without the additional compression (which would be audibly the same as the HD version). No response, the silence was deafening!
> 
> I likewise don't know engineers who don't want to make their recordings sound "as good as possible" but if a client wants a HD version and a more compressed 16bit version, it's my job to give the client what they ask for and I've done exactly this on a number of occasions. I'm not "selling my soul" because there are numerous situations where a more compressed version will sound better. However, decent quality listening environments/equipment is not one of those situations and therefore it's entirely possible to use one of the 16bit versions I've created to demonstrate that HD is better than 16bit. In fact, it was encountering a client of mine doing exactly this which prompted me to start this thread!
> 
> G


 
  
 This thing about creating worse quality by purpose to drive sales of higher (more expensive) specs is sadly not uncommon and could be something that will push hires sales. I hope not, though.


----------



## pinnahertz

globobock said:


> This thing about creating worse quality by purpose to drive sales of higher (more expensive) specs is sadly not uncommon and could be something that will push hires sales. I hope not, though.


 
 Already seen it done.  Also, I found an artist self-releasing his stuff, who released a bunch of "high quality" versions of his over-compressed and loudness-processed originals.  It sounds just as bad as the 16/44.1 version.


----------



## gregorio

globobock said:


> This thing about creating worse quality by purpose to drive sales of higher (more expensive) specs is sadly not uncommon and could be something that will push hires sales.


 
  
 That's the problem, or problems! Firstly, as I alluded, a worse quality version may only be worse in certain listening environments, in other common (probably far more common) use environments, that worse quality version is actually perceived as better quality. Secondly, a mastering engineer may not know the intention/agenda of their client, he/she might believe they're making a version of higher quality for those other common listening environments, while their client might intend using it to demonstrate the superiority of HiDef and Thirdly, even if the mastering engineer does know of a nefarious intention, it's very unlikely he/she would risk their livelihood by publicly revealing it.
  
 This is, as you say, not uncommon or in fact a particularly new phenomena, it's been going on since pretty much the availability of "HiRes" to the public nearly 20 years ago.
  
 G


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## mackie1001

Of the back of that MQA appreciation thread mentioned earlier I thought I'd grab a free trial of Tidal Hifi and have a play. Straight away it was blatantly obvious that the difference I was hearing was due to it being a totally different master and nothing to do with the magical format at all. I set my vanilla Realtek laptop audio device to 16/44.1 plugged some £50 Sony IEMs in and had a listen. Not high end, revealing kit. Difference was very apparent. The MQA version probably 3+ db louder for starters. So in short, people raving about MQA are listening to different tracks. If they're hobbling the CD quality stuff too then it's just a big con. If they're not and just facilitating good remasters being created then that has some merit but MQA are getting money for really not very much. I'd happily listen to a 16/48 version of the same remaster in a FLAC container and not have the price jacked up by a license fee.


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## dimedrol

I've heard wonderful 16/44 records and awful 24/192 records. If every record out there was as good as the best 16/44 records, there wouldn't even be the need for a higher resolution format. People listen to bad 16/44 records and think that we need to increase the resolution. But it's not the resolution they they don't like. Higher resolution formats will eventually replace 16/44 anyway, but that is unlikely to yield a boost in the sound quality, not with current level of audio equipment.


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## Computerpro3

I was playing around tonight and came across this thread and the 8 bit vs. 16 bit test.  http://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php

 I can pass this repeatedly with 95%+ confidence but I absolutely cannot articulate why, and it's driving me nuts.  If I listen to the whole sample and try to focus on different things to figure it out, I get it wrong.  If I listen for a couple seconds and make a snap judgement "this one is better" I can reliably pass.  Best I can figure out is some kind of treble muddiness but I have no idea.  Anyone else have a similar experience? 

 Focusrite 2i4 DAC
 Asgard 2 amp
 HD650 headphones

 EDIT:  I can not pass the Gangnam Style one with more than 60% confidence.


----------



## spruce music

computerpro3 said:


> I was playing around tonight and came across this thread and the 8 bit vs. 16 bit test.  http://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php
> 
> I can pass this repeatedly with 95%+ confidence but I absolutely cannot articulate why, and it's driving me nuts.  If I listen to the whole sample and try to focus on different things to figure it out, I get it wrong.  If I listen for a couple seconds and make a snap judgement "this one is better" I can reliably pass.  Best I can figure out is some kind of treble muddiness but I have no idea.  Anyone else have a similar experience?
> 
> ...


 

 You may have discovered for yourself the effect of echoic memory.  It is why quick switching and very short segments give the finest most discriminating blind test results. 
  
 https://en.wikipedia.org/wiki/Echoic_memory
  
 Echoic memory only lasts a few seconds.  No more than 15 and probably less.  The auditory part of your brain holds  a few seconds of raw sensory input in memory.  You can switch instantly and for a few seconds directly compare two full perceptions, the current one and the one in echoic memory.  After that the perception goes to medium term and then longer term memory.  When it leaves echoic memory compression of the full perception occurs.  It is something like longer term memory is an MP3 of the full actual perception. 
  
 So you need to use something that lets you listen to 2 or 3 seconds, switch near instantly and hear the same 2-3 seconds.  I find doing this lets me get high confidence scores on things I can never perceive over say 30 seconds.  Like different resampling algorithms.  If the difference is small enough this is needed, I also most of the time can't figure out why.  Another thing people who claim blind tests don't work don't want to believe.  I can sometimes get reliable repeatable blind results after I am listening and thinking I am just not hearing anything.  Yet some part of my hearing does perceive a difference I subjectively cannot pick up on.


----------



## pinnahertz

spruce music said:


> You may have discovered for yourself the effect of echoic memory.  It is why quick switching and very short segments give the finest most discriminating blind test results.
> 
> https://en.wikipedia.org/wiki/Echoic_memory
> 
> Echoic memory only lasts a few seconds.  No more than 15 and probably less.  The auditory part of your brain holds  a few seconds of raw sensory input in memory.  You can switch instantly and for a few seconds directly compare two full perceptions, the current one and the one in echoic memory.  After that the perception goes to medium term and then longer term memory.  When it leaves echoic memory compression of the full perception occurs.  It is something like longer term memory is an MP3 of the full actual perception.


 
 David Clark, in his AES paper, "High-Resolution Subjective Testing Using a Double-Blind Comparator" outlines the design of his hardware comparator.  He designed for a uniform 50ms gap between choices so if X=A the A>X switch would not reveal itself because of the absence of a switching gap. I have one of his units. I've found that reducing the gap has a minor but noticeable improvement on test resolution so long as the switch delay between all choices remains consistent. Using other switching methods, such as a fast but seamless crossfade, can be even more revealing. However, 2 seconds of gap would reduce the test resolution substantially. 


spruce music said:


> So you need to use something that lets you listen to 2 or 3 seconds, switch near instantly and hear the same 2-3 seconds.  I find doing this lets me get high confidence scores on things I can never perceive over say 30 seconds.  Like different resampling algorithms.  If the difference is small enough this is needed, I also most of the time can't figure out why.


 
 The highest resolution in ABX comparisons is to exactly synchronize both samples and switch without restarting the sample. Switching can be performed at any point during a test sample of any length. Unfortunately, many software ABX comparators preclude this type of comparison, which unfortunately does limit their resolution.  I have only a hypothesis of why this is, which I believe may be because repeating a short sample segment is highly unnatural, and contributes to the pressure of the test.  Playing a continuous sample and switching at any point lets the listener hear the DUT in a natural context, reducing the clinical nature of the test, and I believe, improving accuracy.  Sorry, I have no data to back up my hypothesis. I mean to test it some day, if some day ever comes. 


spruce music said:


> Another thing people who claim blind tests don't work don't want to believe.  I can sometimes get reliable repeatable blind results after I am listening and thinking I am just not hearing anything.  Yet some part of my hearing does perceive a difference I subjectively cannot pick up on.


 
 Most objections to the ABX/DBT protocol relate to it being "high pressure", a decision is demanded, and comparison time is limited. While the first part of that is true for data collection purposes, in practice there is no time limit. In fact, Clark's ABX comparator was designed to retain its X selections for a very long time permitting ABX tests lasting months.
  
 BTW, as you may notice from his AES paper's title, ABX/DBT is actually a subjective test, not an objective test. Objective testing would be measurement.


----------



## spruce music

pinnahertz said:


> David Clark, in his AES paper, "High-Resolution Subjective Testing Using a Double-Blind Comparator" outlines the design of his hardware comparator.  He designed for a uniform 50ms gap between choices so if X=A the A>X switch would not reveal itself because of the absence of a switching gap. I have one of his units. I've found that reducing the gap has a minor but noticeable improvement on test resolution so long as the switch delay between all choices remains consistent. Using other switching methods, such as a fast but seamless crossfade, can be even more revealing. However, 2 seconds of gap would reduce the test resolution substantially.
> The highest resolution in ABX comparisons is to exactly synchronize both samples and switch without restarting the sample. Switching can be performed at any point during a test sample of any length. Unfortunately, many software ABX comparators preclude this type of comparison, which unfortunately does limit their resolution.  I have only a hypothesis of why this is, which I believe may be because repeating a short sample segment is highly unnatural, and contributes to the pressure of the test.  Playing a continuous sample and switching at any point lets the listener hear the DUT in a natural context, reducing the clinical nature of the test, and I believe, improving accuracy.  Sorry, I have no data to back up my hypothesis. I mean to test it some day, if some day ever comes.
> Most objections to the ABX/DBT protocol relate to it being "high pressure", a decision is demanded, and comparison time is limited. While the first part of that is true for data collection purposes, in practice there is no time limit. In fact, Clark's ABX comparator was designed to retain its X selections for a very long time permitting ABX tests lasting months.
> 
> BTW, as you may notice from his AES paper's title, ABX/DBT is actually a subjective test, not an objective test. Objective testing would be measurement.


 
  
 I have found switching that repeats the segment works better for me rather than switching an ongoing bit of music.  And I never meant a 2 second delay as that reduces discriminating ability of the listener substantially. 
  
 Yes the testing segments can go on as long as one wishes.  Everytime it has been tried shorter is better, shorter is much better, very short is the best.  So if someone wishes to convince themselves fine.  They are only going to convince themselves they heard something sighted and dismiss the test when blinded they don't.
  
 I think there is some truth to the high pressure claim.  You are asking people to listen in a way they aren't comfortable with and in informal tests I believe audiophiles think it is a challenge to their audiophile manhood.  It is like anything you haven't done much.  With repetition and experience it is old hat.  In the case of ABX testing it is tedious rather than pressure filled.  So people need convincing to do it a bit with little or nothing on the line until they aren't uptight about it.  Then you can get good results.
  
 It is a subjective test.  Though with brain scans and such it may be more than that.  Already some years back simple monitoring of auditory nerve activity has been possible for academic testing.  That is one of the things done to show bone conducted ultrasonics are heard though usually an octave lower is how they are perceived.


----------



## PETEBULL

> Originally Posted by *globobock*
> 
> 
> 
> This thing about creating worse quality by purpose to drive sales of higher (more expensive) specs is sadly not uncommon and could be something that will push hires sales.


 
 1. Download Hi-Rez content
 2. Downsample it into 16/44.1
 3. Upload it into your regular player
 4. ???
 5. Profit !!!


----------



## globobock

petebull said:


> 1. Download Hi-Rez content
> 2. Downsample it into 16/44.1
> 3. Upload it into your regular player
> 4. ???
> 5. Profit !!!


 

 I have a Sony DAP (the small A17) which has a down-sampling feature. Basically, what changes is the loudness.


----------



## globobock

computerpro3 said:


> I was playing around tonight and came across this thread and the 8 bit vs. 16 bit test.  http://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php
> 
> I can pass this repeatedly with 95%+ confidence but I absolutely cannot articulate why, and it's driving me nuts.  If I listen to the whole sample and try to focus on different things to figure it out, I get it wrong.  If I listen for a couple seconds and make a snap judgement "this one is better" I can reliably pass.  Best I can figure out is some kind of treble muddiness but I have no idea.  Anyone else have a similar experience?
> 
> ...


 

 CMIIW, but on 8bit against 16bit, the difference should only be on the loudness levels, right? Not on the details?


----------



## PETEBULL

globobock said:


> I have a Sony DAP (the small A17) which has a down-sampling feature. Basically, what changes is the loudness.


 
 You got it wrong. If your player supports hi-res, you have no reason to downsample unless it doesn't handle it well. I meant that you can downsample files through a converter and use on conventional 16/44.1 hardware without buying hi-res DAC.


----------



## VNandor

As far as I know, if you downsample from 16bit to 8bit the only thing that will change is the noisefloor. It goes way up. If there's more difference it it means there were additional processing made.





globobock said:


> CMIIW, but on 8bit against 16bit, the difference should only be on the loudness levels, right? Not on the details?


----------



## castleofargh

petebull said:


> globobock said:
> 
> 
> > I have a Sony DAP (the small A17) which has a down-sampling feature. Basically, what changes is the loudness.
> ...


 
 the sony DAP doesn't allow to use DSPs with highres files. IDK if it's a processing matter or if the DSPs are simply working at a given rate, but that's how it goes on that DAP. so as a cheap alternative, sony has a downsapling option to "enable" DSPs even with those highres files. but of course at that point it makes more sense to import lower res files on the DAP from the start IMO.
 now the loudness part is a mistake from our friend, the option is indeed a downsampling one, the bit depth and/or the loudness aren't affected.


----------



## gregorio

globobock said:


> CMIIW, but on 8bit against 16bit, the difference should only be on the loudness levels, right? Not on the details?


 
  


vnandor said:


> As far as I know, if you downsample from 16bit to 8bit the only thing that will change is the noisefloor. It goes way up. If there's more difference it it means there were additional processing made.


 
  
 If doing a bit reduction (only!) then there should be no difference in loudness and VNandor is correct, the only difference will be a far higher noise floor, which would likely affect the amount of details you can hear. However, how noticeable that far higher noise floor would be and how much detail you therefore wouldn't hear would depend on the actual audio/music plus the amount/distribution of the dither noise-shaping. In practice, with some material and an appropriate noise-shaping algorithm, it might be quite difficult to tell apart an 8bit version from a 16bit one.
  
 If converting from hires to say 16/44.1, then we're not ONLY doing a bit reduction, we're doing a resampling process as well. In this case there will often be a change in loudness (level) as a good sample rate converter should take into account inter-sample peaks and reduce the level to avoid clipping, typically these SRCs will only reduce the level by about 0.2dB or so but that can be enough in some cases to screw up a blind or DBT!
  
 G


----------



## Computerpro3

> CMIIW, but on 8bit against 16bit, the difference should only be on the loudness levels, right? Not on the details?


 
 I have no idea what the technical difference between the two files is, or if the test is even valid.  All I know is that I can reliably pass it.  I cannot do the same for the Gangam Style test.


----------



## Deltrus

computerpro3 said:


> I have no idea what the technical difference between the two files is, or if the test is even valid.  All I know is that I can reliably pass it.  I cannot do the same for the Gangam Style test.


 
 16 bit to 8 bit is not 16 bit to 24 bit... I hope you understand this?


----------



## Computerpro3

deltrus said:


> 16 bit to 8 bit is not 16 bit to 24 bit... I hope you understand this?


 
 Gee, I thought they were all the same and we were just using different numbers for fun.


----------



## VNandor

computerpro3 said:


> I have no idea what the technical difference between the two files is, or if the test is even valid.  All I know is that I can reliably pass it.  I cannot do the same for the Gangam Style test.


 
 That's what I call a screenshot! Is that aliasing I see? 
	

	
	
		
		

		
		
	


	



  
 Anyways going by the look of the test, the only thing that changes is the bit depth. The youtube version of gangam style "looks" compressed (the peak meter of windows' mixer is being around at 0dB and doesn't move a lot). This might explain why 8 bit sounds close to the 16 bit version. Less "dynamic range" takes less bits to encode properly. I could take closer look if you linked to the test but all I could do is to perform a null test to see if the difference between the two versions is just noise. People with better softwares and better understanding might be able to confirm that the only difference is just the bit depth or there's something else going on as well.


----------



## RRod

vnandor said:


> That's what I call a screenshot! Is that aliasing I see?
> 
> 
> 
> ...


 
  
 8-bit truncation errors are ~48dB down from 0dBFS, assuming no dither. Many people can't seem to pick it out in a loud clip such as the NY example, but it's not surprising if someone listening intently at high volume might be able to do it.


----------



## globobock

Allow me a question of the naivest order...
  
 In your studio mixing experiences, what is the actual "True relevant" frequency range for music? I know that the human ear can th(theoretically) reach 20K, but in reality 17K is where most reach their limits.
  
 Based on innerfidelity graphs, most headphones start to roll off at 10K.
  
 Is there a table that shows the freq range for vocals and different instruments?


----------



## dimedrol

google helps a lot with instruments frequency:

 http://www.independentrecording.net/irn/resources/freqchart/main_display.htm


----------



## Brooko

globobock said:


> Is there a table that shows the freq range for vocals and different instruments?


 
  
 http://www.independentrecording.net/irn/resources/freqchart/main_display.htm


----------



## StanD

This link should scare the cr@p out of every audiophile worrying about the high freq roll off in headphones. Apparently women are superior.
https://www.osha.gov/pls/oshaweb/owadisp.show_document?p_table=STANDARDS&p_id=9741


----------



## gregorio

vnandor said:


> The youtube version of gangam style "looks" compressed (the peak meter of windows' mixer is being around at 0dB and doesn't move a lot).


 
  
 A fact which many audiophiles appear completely unaware of is that pretty much all the main streaming service apply loudness normalisation. For about the last two years everything uploaded to YouTube is automatically loudness normalised to the equivalent of about 13.5LUFS and they've since been working on applying this loudness normalisation to content uploaded previously. Youtube is the worst of the streaming services as far as I'm aware, Apple Radio is normalised to the equivalent of about 16.5LUFS for example. I say "about" because no one outside of each of the companies know exactly how they're achieving their normalisation but particularly with YouTube (with such a high normalisation level) the process must often include compression.
  


rrod said:


> 8-bit truncation errors are ~48dB down from 0dBFS, assuming no dither.


 
  
 Truncation errors occur in the LSB so with 8bit that would be at ~42dB down from 0dBFS and it doesn't really matter whether we assume dithering or not. I'm sure you probably realise this but just thought I'd be a bit pedantic about it!
  
 Quote:


globobock said:


> In your studio mixing experiences, what is the actual "True relevant" frequency range for music? I know that the human ear can th(theoretically) reach 20K, but in reality 17K is where most reach their limits.


 
  
 Even near perfect human hearing starts rolling off steeply from about 12kHz AND ON TOP OF THIS, most acoustic instruments also produce decreasing level output (of harmonics) from around 8kHz or lower. IME, there is very little relevance beyond about 16kHz. Vinyl is also particularly attenuated/inaccurate by 16kHz and until about 15-20 years ago TV broadcast was limited to 15kHz. Having said this, I do check what's going on above 16kHz (just in case) but I have to check with a spectrogram as I can't hear that high any more.
  
 G


----------



## RRod

gregorio said:


> Truncation errors occur in the LSB so with 8bit that would be at ~42dB down from 0dBFS and it doesn't really matter whether we assume dithering or not. I'm sure you probably realise this but just thought I'd be a bit pedantic about it!


 
  
 Perhaps I am misusing the term "truncation error" then. If you truncate a 16-bit file down to 8-bit and then pad it back up, the difference in the two signals will peak at -48dB. My point with dither was that a difference file made in the same way will have higher peaks if dither was used.


----------



## Computerpro3

"
_A fact which many audiophiles appear completely unaware of is that pretty much all the main streaming service apply loudness normalisation. For about the last two years everything uploaded to YouTube is automatically loudness normalised to the equivalent of about 13.5LUFS and they've since been working on applying this loudness normalisation to content uploaded previously. Youtube is the worst of the streaming services as far as I'm aware, Apple Radio is normalised to the equivalent of about 16.5LUFS for example. I say "about" because no one outside of each of the companies know exactly how they're achieving their normalisation but particularly with YouTube (with such a high normalisation level) the process must often include compression."_
  
 I do believe Tidal has an option to disable this.  I have been impressed with the quality vs. Spotify.  The difference is fairly noticeable on well recorded classical music.


----------



## gregorio

rrod said:


> Perhaps I am misusing the term "truncation error" then. If you truncate a 16-bit file down to 8-bit and then pad it back up, the difference in the two signals will peak at -48dB. My point with dither was that a difference file made in the same way will have higher peaks if dither was used.


 
  
 I'm not sure if you're misusing the term, I think so, I'm trying to get my head around the implications of your test. If you truncate to 8bit, you'll get truncation error in the 8th bit and truncation error is correlated to the signal. Applying dither in theory randomises (decorrelates) this error, so what you'll end up with when using dither is the same amount of error (level) but distributed evenly as white noise rather than as spikes at particular frequencies. I say in theory because in practise dither is often applied at a somewhat higher level than the level of the truncation error, particularly in the case of noise-shaped dither.
  
 If I'm understanding your test, below -48dB in the 8bit padded version you'll just have digital silence (zeros in the last 8bits) as compared to non-zero data in some of the 8 LSBs in the original, an obvious numerical difference. However, truncation error will be occurring in the 8th bit and although not obvious numerically (because like the original it will contain non-zero values at times), you should get a difference measurable up to -42dB, the same as if you applied a perfect 1 LSB dither. With noise-shaped dither you'd almost certainly get a difference file which is higher, probably around -38dB but of course, it would depend on the actual settings of the dither applied. Dither is not such a complex process to get one's head around but like many things in audio, it's a bit of a rabbit hole when we start looking in detail at it's practical application. For example, we have to think about the application of dither in quite different terms when reducing bit depth to 16bit as compared to reducing it to 8bit.
  


computerpro3 said:


> I do believe Tidal has an option to disable this.


 
  
 Good to know. Apple, Youtube and the others I'm aware of don't have such an option. Hopefully they'll all (including Tidal) eventually come into line with the AES recommendations and standardise to -15LUFS and make sure it can't be turned off!
  
 G


----------



## RRod

gregorio said:


> I'm not sure if you're misusing the term, I think so, I'm trying to get my head around the implications of your test. If you truncate to 8bit, you'll get truncation error in the 8th bit and truncation error is correlated to the signal. Applying dither in theory randomises (decorrelates) this error, so what you'll end up with when using dither is the same amount of error (level) but distributed evenly as white noise rather than as spikes at particular frequencies. I say in theory because in practise dither is often applied at a somewhat higher level than the level of the truncation error, particularly in the case of noise-shaped dither.
> 
> If I'm understanding your test, below -48dB in the 8bit padded version you'll just have digital silence (zeros in the last 8bits) as compared to non-zero data in some of the 8 LSBs in the original, an obvious numerical difference. However, truncation error will be occurring in the 8th bit and although not obvious numerically (because like the original it will contain non-zero values at times), you should get a difference measurable up to -42dB, the same as if you applied a perfect 1 LSB dither. With noise-shaped dither you'd almost certainly get a difference file which is higher, probably around -38dB but of course, it would depend on the actual settings of the dither applied. Dither is not such a complex process to get one's head around but like many things in audio, it's a bit of a rabbit hole when we start looking in detail at it's practical application. For example, we have to think about the application of dither in quite different terms when reducing bit depth to 16bit as compared to reducing it to 8bit.


 
  
 I see what you are saying. The reason I think about it in terms of the difference is that the 8 bits you have left after pure truncation are unaltered, so they don't have any error in terms of their own values. The errors are in the 8 bits that are now zeros that might have had content, and the loudest they can get is -48dB (that is, the decimal values for the difference will be in [-128,128]). Agree with you on the actual complexity of dither. I bring it up in an 8bit context because I find that 8bit non-shaped dither can actually audibly stand out more than pure 8bit truncation errors in the version that is applied by SoX, for example. Sorry for boring everyone to death ^_^


----------



## gregorio

rrod said:


> [1] The reason I think about it in terms of the difference is that the 8 bits you have left after pure truncation are unaltered, so they don't have any error in terms of their own values.
> [2] The errors are in the 8 bits that are now zeros that might have had content, and the loudest they can get is -48dB (that is, the decimal values for the difference will be in [-128,128]).


 
  
 OK, I think I understand your reasoning now. However, I don't believe it's valid in practice and the end result is that you appear to be confusing truncation error with quantisation error, which is quite common because truncation error is a form of quantisation error. 
	

	
	
		
		

		
		
	


	




 If you hack off the last 8 bits of a 16bit file then yes, the remaining 8bits will be numerically identical to the 8 MSBs in the 16bit file and the only numerical difference would be between what was in the 8 LSBs of the 16bit original and nothing (the 8 zeros in the 8 LSBs of our padded 8bit file). However, you seem to be forgetting/ignoring what these numerical values actually represent, which is effectively co-ordinates, which when fed into a sinc function will recreate a continuous wave form. In the case of quantisation error, the difference between the actual (original) value and our available (quantisation) values results in an error signal which is fairly uniformly distributed (in both time and frequency) and isn't significantly correlated to the input signal. Feeding this into a sinc function will result in a perfect recreation of our waveform plus a fairly random amount of all other frequencies (of nearly equal intensity), IE. A perfect signal + fairly constant (nearly white) noise. Truncation distortion on the other hand has a non-uniform distribution and is significantly correlated with the input signal. Feed this into a sinc function and we'll get quite different results, the error is concentrated into fewer frequencies and time related with the signal (not constant). IE. A perfect signal plus some spurious tones. It doesn't really sound like that though, these tones are related to the input signal but not harmonically related, so it actually sounds somewhat similar to digital clipping distortion. As an example, let's say it were possible to make two recordings which were absolutely identical in all respects except that one had a bit depth of 16bit and the other 8bit. Using your reasoning, if we truncate the 16bit version to 8 bits the result would be bit perfect identical with our 8bit recording but this isn't the result we'd get, they wouldn't be bit perfect identical and what came out of the DAC would be significantly different.
  
 1. Due to the above, this statement is in fact incorrect. The 8 bits you've got left after truncation maybe numerically identical to the first 8 MSBs of the 16bit original but it definitely does have "error in terms of their own values", it has truncation error!
  
 2. I see where you're coming from (numerically) but this statement isn't correct either. What you're effectively saying is that the loudest that nothing (digital silence) can get in a truncated 8bit file is -48dB, which is true in a sense but is unrelated to the loudest that the error can get by truncating those 8 bits. If we're looking at the actual error caused by hacking those 8 bits off, rather than just at the 8 bits which no longer exist, then with truncation we could end up with error peaks as high as -36dB or so and what's more, it could sound higher than it's level suggests due to our ears' sensitivity to this type of non-harmonic distortion.
  
 G


----------



## RRod

So indeed, if I mock up reconstruction using SoX (resample to some ungodly high rate and keep things at 32 bits), the difference between the reconstructions of the original file and the 8-bit version end up peaking at -42. I picked a hell of a day to quit drinking. Thanks for that; I'll keep trying to digest what's happening, because I'm totally not grokking it.


----------



## castleofargh

rrod said:


> So indeed, if I mock up reconstruction using SoX (resample to some ungodly high rate and keep things at 32 bits), the difference between the reconstructions of the original file and the 8-bit version end up peaking at -42. I picked a hell of a day to quit drinking. Thanks for that; I'll keep trying to digest what's happening, because I'm totally not grokking it.


 
 it's sad that we're not water brothers, not even beer brothers. makes communicating harder.
 I tend to naturally think the way you do/did, while knowing it's wrong, which is how my brain seems to enjoy irony the most. keeping the flawd logic while adding the extra "oh BTW you remember that it's wrong right?"
	

	
	
		
		

		
			





.  but yeah practical tests tend to agree with Greg(he probably altered our reality just to be right, you know him). I'm guessing we're having the same faulty logic saying that if I remove X bits, I'm only losing the accuracy of those removed bits. in term of data it's true.
  
 but thinking differently, the LSB is almost always wrong compared to the actual signal amplitude it's coding, and adding 1 more bit decreases that error, making it the size of that new LSB. but an error still. the last bit gives the magnitude of the error.  does that make sense?(it's totally what Greg is saying but from a dumbed down brain).


----------



## gregorio

rrod said:


> So indeed, if I mock up reconstruction using SoX (resample to some ungodly high rate and keep things at 32 bits), the difference between the reconstructions of the original file and the 8-bit version end up peaking at -42. I picked a hell of a day to quit drinking. Thanks for that; I'll keep trying to digest what's happening, because I'm totally not grokking it.


 
  
 Let's try an analogy: Let's say we take an adult man and amputate one of his legs above the knee. The difference between the man before and after the amputation isn't just the difference between the gap (nothing) which now exists and the part of his leg which has been amputated, there's quite a big difference in what's left of the man himself: For starters he's got a severe wound to deal with and it's also going to cause other changes/differences in his body, metabolic (and probably numerous other) changes for example. Furthermore, there are also going to be significant differences between this adult amputee and the same man had he been born with a congenital defect which resulted in exactly the same gap/nothing above the knee. Likewise in digital audio, there's a difference between running out of resolution/bits when recording and chopping off resolution/bits which did once exist. In fact, truncation error is double the quantisation error in terms of the RMS of the error signal.
  
 I've looked around for a few mins and can't find a quote of the actual math to back this up but I seem to remember that the RMS of quantisation error is just under 0.3 LSB (and therefore about 0.575 LSB for truncation). The LSB in the context of this discussion being the 8th bit, not say the 9th or 16th bit which has been truncated (and then padded with zeros).
  
 I also looked around for an actual example, so you can hear for yourself, which will hopefully help you to get your head around it. I came up with this example (which is actually a link from the previous article posted). While we can't in practise create an 8bit recording (with only quantisation error) for comparison with an 8bit version truncated from a 16bit version, this example does demonstrate well the difference between a 16bit original and the various different ways of arriving at 8bit versions; truncation, dithering and noise-shaped dithering. However we get to 8bit from 16bit though, there is ALWAYS going to be error in those remaining 8bits, regardless of the fact that they're identical to the 8 MSBs in the 16bit original.
  
 One last point. While the same math applies to bit reductions of any depth, there are additional factors to consider when doing a bit reduction from 24bit to 16bit. Additional factors which are commonly ignored and which significantly change the result!
  
 Quote:


castleofargh said:


> ... but yeah practical tests tend to agree with Greg(he probably altered our reality just to be right, you know him).


 
  
 Hmm, it's true that I in effect do often try to alter the perceived reality but I don't think I'm doing it just to be right, I'm doing it to get closer to what is really going on under the hood. Having said this, my perception of what is going on under the hood has it's limits and is not perfect, so maybe in effect it is just about me being wanting to be right, hmmm?! On the other side of the coin, many of the questions/discussions revolve around issues which I investigated as long as 20 years or so ago and in many cases have continued to improve my understanding ever since.
  
 The fact is that under the hood, digital audio is ultimately of course all math, much of which requires requires a quite highly educated mathematician to fully understand. Furthermore, even a great mathematician who would find digital audio math simplistic, wouldn't have the years of audio engineering experience necessary to be sure they're actually taking into account ALL the math relevant to a particular issue/topic of discussion. Us pro audio engineers are not mathematicians though, we're only ultimately concerned with the perceptual results of employing the math. So like the more educated consumers, we have to rely on layman's terms and analogy, although with experience we should have a far better understanding of what underpins those layman's terms and analogies. Ultimately though, if we desire to go further down the rabbit hole, we have to trust/consult others, of which there are incredibly few who are willing to speak publicly, have enough independence from marketing, a deep enough understanding of the math and a broad enough understanding of practical audio to stand a good chance of considering ALL the relevant math. Bob Katz, Paul Frindle, Dan Lavry and just a few others have fit the bill for me over the course of many years.
  
 So when I see a misapprehension due to the inevitable, inherent inaccuracies of layman's terms and analogies, I'll try to come at the issue from a different angle and with some different analogies, to hopefully create a more comprehensive understanding. This approach could easily be seen as "trying to alter your reality"! I could in theory just quote the math but I don't think that would help most here and besides, I commonly don't know the math (because it's proprietary) and even when I do, I often don't understand it comprehensively enough to discuss it in mathematical terms. So while I might sometimes allude to the underlying math, I try to avoid going too far down that hole.
  
 G


----------



## castleofargh

it was a kind of follow up joke. with Stranger in a strange land, fullness and what one can do with it...  
	

	
	
		
		

		
		
	


	



 soz.


----------



## limpidglitch

Is this what you are talking about, essentially the difference between sample peak and true peak?
  
 sox -n -c 1 -r 48k -b 16 noise16.wav synth 60 whitenoise gain -1 
 sox noise16.wav -b 8 noise8.wav -D 
 sox noise8.wav -b 16 noise16pad.wav 
 sox -m noise16.wav -v -1 noise16pad.wav diff.wav 
 sox diff.wav -r 192k diff-ups.wav 
 sox diff.wav -n stats 2>&1 | grep "Pk lev dB" 
# Pk lev dB -48.16 
 sox diff-ups.wav -n stats 2>&1 | grep "Pk lev dB" 
# Pk lev dB -42.18


----------



## RRod

@gregorio I don't find the amputation example too revealing, as I keep thinking (probably wrongly) "how have things changed if you reattach the limb?" But I can see somewhat of a problem from the math perspective, in that truncation and rounding a different operations. If we wanted to round off 126 to 2 places, we really want 130 but truncation+pad gives us 120. But I'm still not sure if that's really relevant, and even if it is I still need to flesh out how it works in reconstruction. As you stated, being a math guy I am perfectly fine if you give me a proof, but I have zero instinct when it comes to storing numbers on a computer.
  
 Quote:


limpidglitch said:


> Is this what you are talking about, essentially the difference between sample peak and true peak?
> 
> sox -n -c 1 -r 48k -b 16 noise16.wav synth 60 whitenoise gain -1
> sox noise16.wav -b 8 noise8.wav -D
> ...


 
  
 Yes, so my question is if this is just two ways of talking about the same issue. Is the fact that a sinc will sometimes clip intimately related to the peak value of our errors? I'm all jumbled...


----------



## Arpiben

castleofargh said:


> it was a kind of follow up joke. with Stranger in a strange land, fullness and what one can do with it...
> 
> 
> 
> ...


 
  
 Rather than a joke I will consider it more like a multi feet slaughtery 
	

	
	
		
		

		
		
	


	



 Since 16 bits to 8 bits' truncation can be considered for the sampled signal as:

an apodization (foot removal) when the 8 MSB are kept into 8 bits
an apodization with 'sticks' when the 8 MSB are kept into 16 bits and the 8 LSB are filled with 0.
 Add one more apodization during FFT windowing (Blackman Harris) for the fun.
  
 Indeed, IMHO, it was a great post from @gregorio


----------



## StanD

arpiben said:


> Rather than a joke I will consider it more like a multi feet slaughtery
> 
> 
> 
> ...


 

 At the very least a skilled surgeon would round the LSB rather than hack it off.


----------



## gregorio

limpidglitch said:


> Is this what you are talking about, essentially the difference between sample peak and true peak?


 
  
 Nope, that's a different issue unrelated to the one currently under discussion. Intersample peaks is an issue of upsampling/reconstruction rather than of bit reduction/reconstruction. Reducing the bit depth without resampling does not affect the value of the MSBs.
  


rrod said:


> [1] I don't find the amputation example too revealing, as I keep thinking (probably wrongly) "how have things changed if you reattach the limb?"
> 
> [2] But I'm still not sure if that's really relevant, and even if it is I still need to flesh out how it works in reconstruction.


 
  
 1. That is "probably wrongly". If you reattach the limb (perfectly) things haven't changed in the slightest (maybe not in this analogy but if we removed and then replaced those 8 LSBs). I can see how you could infer from this that all the change therefore occurs only in those 8 LSBs but this is wrong because those 8 MSBs which we're left with are in effect no longer a coherent number (as far as a sinc function is concerned). Likewise our amputee isn't just the identical man he was with the only difference being in the removed limb, his metabolism and the rest of his body have been affected by the loss of that limb. Let's try another analogy, let's say we have a 4 cylinder car engine and we disable two cylinders, do we now have a car which is identical to before except it has half the horse power? We can re-enable those cylinders and the car will work perfectly again but disabling two cylinders has an effect on that car beyond just those two missing cylinders. In fact, if we could even get the engine to start in the first place, it will probably rip itself to pieces pretty quickly because it's out of balance. We can (and do) build 2 cylinder engines which work absolutely fine but there's a huge difference between a 2 cylinder engine and a 4 cylinder engine with only two working cylinders/pistons, even though the number of working cylinders are exactly the same. That's what's happening here, the 8 remaining bits are exactly the same as the 8 MSBs in our 16bit original, but this 8 bit file is supposed to be a 16bit file and those remaining 8 bits are effectively now "out of balance".
  
 2. Absolutely, I mentioned before that we HAVE to consider reconstruction, how those numbers are going to relate to a reconstructed signal! For reconstruction, we need a series of numbers and and to consider what will happen when the sinc function "joins the dots", then the issue becomes one of statistics, of probability distributions. Trying to take a single number and imagining what happens if we just reduce it's accuracy is not going to give us the full picture of what's going on.
  
 BTW, rounding error is essentially what we get with quantisation error. We can also round truncation error but the results are still very different to each other. If we round the error causing truncation distortion we still get truncation distortion, although presumably somewhat less.
  
 Did you have a listen to the demo I linked to?
  
 G


----------



## RRod

gregorio said:


> Spoiler: Nope, that's a...
> 
> 
> 
> ...


 
  
 Yeah I've gone through audiocheck's stuff before. This paragraph from there seems to be related to my issues:
  


> By down-converting the 16-bit file into 8-bit, every sample now gets truncated to one of 256 possible values (the original had 65,536 possible values). Severe quantization distortion occurs. The loss of clarity below -36 dBFS and the absence of any signal below -48 dBFS are the typical limitations of 8-bit audio files.


 
  
 So we see both the -48 value and the higher -36 that you mentioned. But I can hear stuff happening way before the -36dB mark, so I'm still at a loss for how you make a # for "where does the bit depth start to suck because of truncation". But I guess what I'm hearing proves your point: the errors happen way above -48dB.
  
  
 The car analogy doesn't really do it for me, sadly.


----------



## gregorio

rrod said:


> [1] I can hear stuff happening way before the -36dB mark, so I'm still at a loss for how you make a # for "where does the bit depth start to suck because of truncation".
> 
> [2] But I guess what I'm hearing proves your point: the errors happen way above -48dB. ... The car analogy doesn't really do it for me, sadly.


 
  
 1. Ah, OK. I'm not sure that there is a calculable # (peak value)! There's two issues at play here: A. Quantisation error is quite evenly distributed and quite decorrelated from the signal, therefore the RMS value, which we can easily calculate is useful and gives us a fair indication of it's likely peak value (very roughly -42dB). This isn't the case with truncation error though, it's not evenly distributed and it's correlated to the signal, so although we know it's RMS value, it's peak value, being input signal dependent, could be far higher than the RMS would suggest. It's possible that there is a way to calculate it's peak value but I personally don't know it. My ~-36dB peak figure was no more than a reasoned guess to be honest! B. The result of truncation error is non-harmonically related spikes, which sounds a bit like odd-harmonic distortion and to which human hearing is particularly sensitive, so regardless of it's actual peak level, it sounds more noticeable than it's values would suggest (even if we knew it's values!). In the link provided, I can discern distortion even in the first example (0dB), although it's obviously more noticeable in subsequent examples/levels.
  
 2. Shame, I thought the engine analogy worked quite well, because the error isn't in the two cylinders removed, it's in what's left of the engine (the 8 MSBs), which probably wouldn't even start. It's good though that the audio demos helped, at least you know it's there now. All you've got to do now is develop your own understanding of how/why it's that way, unless someone else can figure out a more helpful analogy than I've managed to come up with! 
  
 G


----------



## RRod

gregorio said:


> 1. Ah, OK. I'm not sure that there is a calculable # (peak value)! There's two issues at play here: A. Quantisation error is quite evenly distributed and quite decorrelated from the signal, therefore the RMS value, which we can easily calculate is useful and gives us a fair indication of it's likely peak value (very roughly -42dB). This isn't the case with truncation error though, it's not evenly distributed and it's correlated to the signal, so although we know it's RMS value, it's peak value, being input signal dependent, could be far higher than the RMS would suggest. It's possible that there is a way to calculate it's peak value but I personally don't know it. My ~-36dB peak figure was no more than a reasoned guess to be honest! B. The result of truncation error is non-harmonically related spikes, which sounds a bit like odd-harmonic distortion and to which human hearing is particularly sensitive, so regardless of it's actual peak level, it sounds more noticeable than it's values would suggest (even if we knew it's values!). In the link provided, I can discern distortion even in the first example (0dB), although it's obviously more noticeable in subsequent examples/levels.
> 
> 2. Shame, I thought the engine analogy worked quite well, because the error isn't in the two cylinders removed, it's in what's left of the engine (the 8 MSBs), which probably wouldn't even start. It's good though that the audio demos helped, at least you know it's there now. All you've got to do now is develop your own understanding of how/why it's that way, unless someone else can figure out a more helpful analogy than I've managed to come up with!
> 
> G


 
  
 Well I think your 1) is quite helpful in getting there, as I can certainly understand that the reconstruction of truncation errors that, by happenstance, end up to be "noise-ish" can have a much different peak value than the reconstruction of errors that look more like a square wave. So in essence the -48dBFS sample difference maps to different actual peak values depending on the nature of the signal. I guess that's my great sin here: treating samples as though they were actual signal.


----------



## gregorio

rrod said:


> [1] I can certainly understand that the reconstruction of truncation errors that, by happenstance, end up to be "noise-ish" can have a much different peak value than the reconstruction of errors that look more like a square wave.
> [2] So in essence the -48dBFS sample difference maps to different actual peak values depending on the nature of the signal.
> [3] I guess that's my great sin here: treating samples as though they were actual signal.


 
  
 1. Yes, although two points: "happenstance" should actually be "statistical probability distribution", IE. Will definitely be quite near to perfect white noise. And secondly, truncation error sounds rather squarish but doesn't look much like a square wave.
  
 2. I'm having difficulty understanding this. By "-48dBFS sample difference" do you mean the 9th and subsequent bits which have been removed? If so, then obviously they can't "map" to anything because they no longer exist. If you mean the 8th bit, then the relationship between the 8th bit and the 9th bit no longer exists (because there is no 9th bit) which will cause the sinc function to create spurious non-harmonic tones as a consequence of this error.
  
 3. Essentially yes, or rather, treating a sample as a signal instead of treating them as a bunch of samples and the analogue signal they'll be converted into. This is an easy trap to fall into because a stream of digital samples is effectively a signal, it's just not related to an analogue signal until it's been decoded by a sinc function. This has several consequences, for example, I was confused by your statement #2 because of your use of "-48dB". -48dB is the quantisation noise floor dictated by 8 bit, however -48dB is not the limit of the signal we can encode in 8 bit, in theory the limit of 8 bit (or any other bit depth) is any signal level down to minus infinity(!), the question is instead: At what point can we still detect that signal within the noise floor? As the linked demo shows, with TDPF dithering we can resolve down to -54dB and if we noise-shape the dither, down to -66dB. The Nyquist-Shannon Theorem is true at all bit depths, which is why it doesn't specify a bit depth! I think you're confusing sample value with what a sinc function will actually render in response to a stream of sample values.
  
 G


----------



## VocaloidDude

Wonderful article. I laughed my ass off at the part about dying instantly if you had the equipment to reproduce 24bit audio.


----------



## RRod

gregorio said:


> 1. Yes, although two points: "happenstance" should actually be "statistical probability distribution", IE. Will definitely be quite near to perfect white noise. And secondly, truncation error sounds rather squarish but doesn't look much like a square wave.
> 
> 2. I'm having difficulty understanding this. By "-48dBFS sample difference" do you mean the 9th and subsequent bits which have been removed? If so, then obviously they can't "map" to anything because they no longer exist. If you mean the 8th bit, then the relationship between the 8th bit and the 9th bit no longer exists (because there is no 9th bit) which will cause the sinc function to create spurious non-harmonic tones as a consequence of this error.
> 
> ...


 
  
 On 2) I'm imagining the reconstruction of the difference signal itself, but I probably need to stop doing that. On 3) I tend to think of multiplying the sinc function by each sample value and then adding up all the resulting scaled sincs. I do wonder if underlying my confusion is the sneaky reality that truncation is not linear; that is, I haven't been taking the "distortion" part of "truncation distortion" seriously enough.


----------



## gregorio

rrod said:


> On 2) I'm imagining the reconstruction of the difference signal itself, but I probably need to stop doing that. On 3) I tend to think of multiplying the sinc function by each sample value and then adding up all the resulting scaled sincs. I do wonder if underlying my confusion is the sneaky reality that truncation is not linear; that is, I haven't been taking the "distortion" part of "truncation distortion" seriously enough.


 
  
 2. I'm still not sure what you mean by "difference signal itself"?
 3. Not sure I understand this either! 
	

	
	
		
		

		
		
	


	






  
 G


----------



## StanD

Let's approach thins from a perspective of visualization. Imagine a complex waveform of only a single instrument playing a single note. This would not be a simple sine wave but have all sorts of squiggles deviating from the fundamental tone frequency sine wave due to the various harmonics and their continual phase shifts. What do you think will happen due to the truncated 8 bit quantization? Intuitively I would suspect that much of this would be lost due to the lack of resolution made worse by truncation. What say you gents?


----------



## pinnahertz

stand said:


> Let's approach thins from a perspective of visualization. Imagine a complex waveform of only a single instrument playing a single note. This would not be a simple sine wave but have all sorts of squiggles deviating from the fundamental tone frequency sine wave due to the various harmonics and their continual phase shifts. What do you think will happen due to the truncated 8 bit quantization? Intuitively I would suspect that much of this would be lost due to the lack of resolution made worse by truncation. What say you gents?


 
 If you were to look at that waveform on something like an oscilloscope, it's unlikely you'd see any difference at all  unless you were to look at only a small portion of the waveform. There just not enough resolution in a scope display if you display a full cycle of a 0dBFS signal. 
  
 The visualization part of this doesn't work well.


----------



## gregorio

stand said:


> [1] What do you think will happen due to the truncated 8 bit quantization?
> [2] Intuitively I would suspect that much of this would be lost due to the lack of resolution made worse by truncation.


 
  
 1. I'm not sure what you'd see on a scope. In a spectogram I'd expect to see a peak representing the fundamental, another lower peak for the first harmonic, a lower peak for the second harmonic and then dozens of other lower peaks again, some below the fundamental, which represents the truncation distortion. Among these dozens of peaks will be the higher harmonics of the instrument but how many you'd be able to visually identify by their amplitude would depend.
  
 2. The lack of resolution itself wouldn't be a problem, you wouldn't loose anything that hadn't already been lost in the original. But how many harmonics would actually be identifiable would depend on where in the frequency spectrum those harmonics are, their amplitudes, the amount of dither and the amount and distribution of any noise-shaping of the dither. The amount lost due to lack of resolution could, under the right conditions, be very little or even none. So it could be that's it's not a case of loss "made worse by truncation" but a case of pretty much the only loss being due to truncation error, although reducing down to just 8 bits, the "right conditions" for this to be the case would be fairly unlikely.
  
 G


----------



## StanD

Perhaps it's time for something you can't mention in too many places in head-fi, carefully constructed ABX listening tests. Will a stradivarius still sound the same?


----------



## old tech

stand said:


> Perhaps it's time for something you can't mention in too many places in head-fi, carefully constructed ABX listening tests. Will a stradivarius still sound the same?


 
 https://en.wikipedia.org/wiki/Player_preferences_among_new_and_old_violins
  
 Perhaps use a modern violin.


----------



## nvfan

Isn't the highest resolution technically something like 22bit, and 24bit is virtually impossible with modern technology?


----------



## old tech

fabianvalluy said:


> Isn't the highest resolution technically something like 22bit, and 24bit is virtually impossible with modern technology?


 
 It's not just the possibility but also pointless.  24bit resolution is below the noise level generated by resistors and other electronics.


----------



## StanD

stand said:


> Perhaps it's time for something you can't mention in too many places in head-fi, carefully constructed ABX listening tests. Will a stradivarius still sound the same?


 
  
  


old tech said:


> https://en.wikipedia.org/wiki/Player_preferences_among_new_and_old_violins
> 
> Perhaps use a modern violin.


 

 Modern is good, I just used Stradivarius in a humorous context.


----------



## Intensecure

stand said:


> Modern is good, I just used Stradivarius in a humorous context.



Actually, there is an interesting study done blind testing musicians and Stradivarius violins..Not quite what you were inferring but still quite amusing..  We are all fallible..
http://www.thestrad.com/blind-tested-soloists-unable-to-tell-stradivarius-violins-from-modern-instruments/


----------



## StanD

intensecure said:


> Actually, there is an interesting study done blind testing musicians and Stradivarius violins..Not quite what you were inferring but still quite amusing..
> 
> 
> 
> ...


 
 I'd like to see a proper blind test of a $99 Schiit Modi 2 DAC vs. some megabuck DACs that have a religious following. Even 24/32 bit on the most worshipped DACs vs 16 bit on the Schiit. I suspect may of the purists will require medication once proven that they can't tell a difference.


----------



## pinnahertz

old tech said:


> It's not just the possibility but also pointless.  24bit resolution is below the noise level generated by resistors and other electronics.


 
 Only at the A/D or D/A, in other words, where analog and digital transition to or from each other.  Internal data structures can and do go to 64 bit floating point.  Adobe Audition does everything at 32 bit FP.
  
 The best A/D I'm aware of has a noise floor down at about 22 bits (German company "Stage Tec").  They cascade more than one A/D to get that to happen and convert at 32 bits, then use DSP to change gain and spit out 24 bit words. But you'd pretty much classify their stuff as a bit exotic.


----------



## castleofargh

fabianvalluy said:


> Isn't the highest resolution technically something like 22bit, and 24bit is virtually impossible with modern technology?


 

 and that's just looking at the DAC. for the entire recording/playback list of processes and gears, even 16bit final resolution to my ears is highly optimistic. I mean when do we get a headphone that has flat FR and no distortions at -90db? and how often do we listen to louder than 110db singers performing in anechoic chambers?


----------



## pinnahertz

stand said:


> I'd like to see a proper blind test of a $99 Schiit Modi 2 DAC vs. some megabuck DACs that have a religious following. Even 24/32 bit on the most worshipped DACs vs 16 bit on the Schiit. I suspect may of the purists will require medication once proven that they can't tell a difference.


 
 It's a dead end. The purists don't acknowledge ABX as a valid test protocol.


----------



## castleofargh

pinnahertz said:


> stand said:
> 
> 
> > I'd like to see a proper blind test of a $99 Schiit Modi 2 DAC vs. some megabuck DACs that have a religious following. Even 24/32 bit on the most worshipped DACs vs 16 bit on the Schiit. I suspect may of the purists will require medication once proven that they can't tell a difference.
> ...


 

 true, also how hard is it to get a colored DAC and then claim it is the only transparent one because I like more songs on it? ^_^


----------



## TheoS53

Found this on Wikipedia:

24-bit digital audio has a theoretical maximum SNR of 144 dB, compared to 96 dB for 16-bit; however, as of 2007 digital audio converter technology is limited to a SNR of about 123 dB[12][13][14] (21-bit ENOB) because of real-world limitations in integrated circuit design. Still, this approximately matches the performance of the human auditory system.[15][16] (While 32-bit converters exist, they are purely for marketing purposes and provide no practical benefit over 24-bit converters; the extra bits are either zero or encode only noise.)[17][18]


----------



## StanD

theos53 said:


> Found this on Wikipedia:
> 
> 24-bit digital audio has a theoretical maximum SNR of 144 dB, compared to 96 dB for 16-bit; however, as of 2007 digital audio converter technology is limited to a SNR of about 123 dB[12][13][14] (21-bit ENOB) because of real-world limitations in integrated circuit design. Still, this approximately matches the performance of the human auditory system.[15][16] (While 32-bit converters exist, they are purely for marketing purposes and provide no practical benefit over 24-bit converters; the extra bits are either zero or encode only noise.)[17][18]


 
 The great P.T. Barnum was overheard saying, "There's a sucker audiophile  born every minute." He went on to become the most successful marketing expert on DACs.


----------



## Arpiben

rrod said:


> Yeah I've gone through audiocheck's stuff before. This paragraph from there seems to be related to my issues:
> 
> 
> So we see both the -48 value and the higher -36 that you mentioned. But I can hear stuff happening way before the -36dB mark, so I'm still at a loss for how you make a # for "where does the bit depth start to suck because of truncation". But I guess what I'm hearing proves your point: the errors happen way above -48dB.
> ...


 
  
 Allow me to try another way since previous analogies didn't work for you.
 Basically I truncated two 44.1 kHz/16 bit files (no dither) into 44.1 kHz/16bit keeping only the 8 MSB:

1 Hz sine 44100 samples/1s in order to maximize sample distribution
10 kHz sine 44100 samples/1s
  
 Truncation was performed with excel  with help of audacity sample export/import function.
 For sure real audio signals are much more complex than simple sine. For sure there were some roundings in the process.
 But the purpose was trying to show the truncation issue.
  

1Hz
 
Time domain:
 

Frequency Domain:
 
  
 2. 10 kHz:
  

Frequency Domain:
 
  
 With the truncated signals (8 bit MSB) you will notice the noise level increase at frequency domain as well as some added clipping for the first signal.
 Hope it helps.
 Rgds.


----------



## pinnahertz

arpiben said:


> Allow me to try another way since previous analogies didn't work for you.
> Basically I truncated two 44.1 kHz/16 bit files (no dither) into 44.1 kHz/16bit keeping only the 8 MSB:
> 
> 1 Hz sine 44100 samples/1s in order to maximize sample distribution
> ...


 
 Wouldn't you need a reconstruction filter in your excel truncation?


----------



## RRod

I think we are losing sight of what was the original issue: "how loud are the errors you get from truncating 16 bits to 8?"
  
 G's answer, as I'm interpreting it, is "it depends", which I agree with now that I'm thinking about reconstruction rather than just the samples themselves. My end-goal in all this is to be able to say "if you're hearing something when you truncate to 8-bits, then it can only be *this* loud, since it has to be lower than the peaks." We don't hear peaks, of course, but being able to bound something by the peak would have been nice. Since we can't, it would be nice to know if the RMS for X samples has some bounding properties that might be useful.


----------



## Yuri Korzunov

arpiben said:


> Allow me to try another way since previous analogies didn't work for you.
> Basically I truncated two 44.1 kHz/16 bit files (no dither) into 44.1 kHz/16bit keeping only the 8 MSB:
> 
> ...
> ...


 
  
 If you remove (put in zero) 8 lesser significant bits, signal lose a bit energy, comparing 16 bit original. Hence *there overload is impossible*.
  
 After removing the 8 bits energy of signal distributed in rest band as quantization noise. But total energy of 8 bit (signal + noise) is decreased comparing 16 bit on energy of lesser 8 bits.
  
 So check, please, rounding or calculations or plot drawing function.


----------



## Arpiben

pinnahertz said:


> Wouldn't you need a reconstruction filter in your excel truncation?


 
 Yes if I didn't lost sight of the original issue as @RRod replied:
_I think we are losing sight of what was the original issue: "*how loud* are the errors you get from truncating 16 bits to 8?"_
 I stopped at FFT(Blackman-Harris) added noise when dealing with sample truncation only.
 How this affects the interpolation/reconstruction I have not the skills to do it.


----------



## Arpiben

yuri korzunov said:


> If you remove (put in zero) 8 lesser significant bits, signal lose a bit energy, comparing 16 bit original. Hence *there overload is impossible*.
> 
> After removing the 8 bits energy of signal distributed in rest band as quantization noise. But total energy of 8 bit (signal + noise) is decreased comparing 16 bit on energy of lesser 8 bits.
> 
> So check, please, rounding or calculations or plot drawing function.


 

 I will check, thanks.


----------



## Yuri Korzunov

arpiben said:


> I will check, thanks.


 

 Me seems, that points of overload was restored for drawing by truncated data via interpolation. I can't see it exactly in screenshot resolution.
  
 If it so, interpolation as oversampling can create virtual points with level higher, that original samples.


----------



## gregorio

rrod said:


> I think we are losing sight of what was the original issue: "how loud are the errors you get from truncating 16 bits to 8?"
> 
> [1] G's answer, as I'm interpreting it, is "it depends", which I agree with now that I'm thinking about reconstruction rather than just the samples themselves.
> [2] My end-goal in all this is to be able to say "if you're hearing something when you truncate to 8-bits, then it can only be *this* loud, since it has to be lower than the peaks." We don't hear peaks, of course, but being able to bound something by the peak would have been nice. [3] Since we can't, it would be nice to know if the RMS for X samples has some bounding properties that might be useful.


 
  
 1. Just to be clear, my answer of "it depends" was based on two points: ...
 2. We always have to be careful with the term "loud" because loudness is a perception rather than a property. We have to be particularly careful about this term when talking about bit reduction because in effect, any correlation between level and loudness is often inverted. IE. The method of bit reduction which produces the highest RMS level of artefacts is the method with the least "loud" artefacts.
 3. The RMS value of truncation error is known, (if memory serves) it is: 1 divided by the square root of 3, LSB. This result is a constant regardless of the input signal. How this RMS level is distributed cannot be known though, because it's signal dependent. I don't know for sure that there is no way to calculate the freq distribution and peak value of truncation error but if there is, you would obviously have to know the input signal, IE. The result is not constant, unlike the RMS amount and unlike quantisation error (which is nearly constant).
  
 G


----------



## Arpiben

gregorio said:


> 1. Just to be clear, my answer of "it depends" was based on two points: ...
> 2. We always have to be careful with the term "loud" because loudness is a perception rather than a property. We have to be particularly careful about this term when talking about bit reduction because in effect, any correlation between level and loudness is often inverted. IE. The method of bit reduction which produces the highest RMS level of artefacts is the method with the least "loud" artefacts.
> 3. The RMS value of truncation error is known, (if memory serves) it is: 1 divided by the square root of 3, LSB. This result is a constant regardless of the input signal. How this RMS level is distributed cannot be known though, because it's signal dependent. I don't know for sure that there is no way to calculate the freq distribution and peak value of truncation error but if there is, you would obviously have to know the input signal, IE. The result is not constant, unlike the RMS amount and unlike quantisation error (which is nearly constant).
> 
> G


 
  
 Completing your post regarding RMS value with the help of some previous readings since my memory is not serving me as much as yours,
  
  

  
*Error function:*
  
  
 With proper choice of dither function, dither noise and quantization error will be uncorrelated from each other and therefore the total error noise power will be additive:
  

  

  
 Q: Quantization width   R:Range   B: bits
  

  
  
 As you rightly mentionned the total error function is signal dependent. Using triangular nonsubstractive dither makes the power spectrum of  the error independent of the input by whitenning the error function. For deeper readings in quantization & dithering one may check Lip****z & R.Wannamaker related documents
  
 Thanks @gregorio for having enlightened me with your posts.
  
 N.B. Edited but still not able/allowed to write Lip ****z corectly, sorry.


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## Traveller (Apr 29, 2017)

So how many bits am I looking at with my "GOTG2" setup... ?


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## Mr Rick

To roughly quote Rhett Butler:  " Frankly, my dear, I don't give a bit !"


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## Yuri Korzunov

traveller said:


> So how many bits am I looking at with my "GOTG2" setup... ?


 
  
 10 bit?


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## LajostheHun

That's probably generous.


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## protoss

how about 24bit vs 32bit?   (32 bit 384khz)


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## LajostheHun

How about is? It's just a big ass file, waste of storage.


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## HAWX (May 1, 2017)

Hi people!

First of all, In my opinion, with todays technology, more than 16 bits and 44.1 kHz does not provide any descernable diffrence. My further understandings are only about the digital audio format, not about If "High Res" is necessary for music reproduction. You can think about the signal is produced by a computer, and decoded by other computer, no playback in real life.

What I learnt about more than CD quality, from this forum + a nice person in this forum:
1) With 24 bit, the minimum loudness step (or say loudness presicion) is increased, although we can not descern it.
2) 24 bit also allows more dynamic range, which I think unnecessary for music reproduction.


3) With higher sampling rates minimul frequency decimal we can go is less, in other words higher frequency presicion, altough standart CD quality already exceeds human hearing in reproduction.
4) Higher sampling rates also allows a file to contain more high pitched wave informations.

Ok, so the 2 and 4 are obvious ones, no need to argue about them.

About 1 and 3, please only contrubute If you have solid information. I can't say they are 100% true. But please don't tell me go watch xiph videos, or we don't need to talk about things we can't percieve with the audotory system, or any kind of side tracking, please. I have read this threads half of it and checked most of the external links given.

And my new question from thsi video; 
Mark Waldrep, AIX records talks about High Res audio, and at some point ( 41:00 ) he says "192 kHz reduces the timing and delays are below the human treshold.."
5)So my question is Does high sampling rate also reduces the delays and improves the timing (not asking wheather we discern it or need it)?

Edited for the errors and clarity.


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## gregorio

protoss said:


> how about 24bit vs 32bit?   (32 bit 384khz)



If you've read the original post then you'd realise that however many more bits beyond 16 you go is just wasted. For the consumer distribution of music, even 16bits is more than enough.



HAWX said:


> Mark Waldrep, AIX records talks about High Res audio, and at some point ( 41:00 ) he says "192 kHz reduces the timing and delays are below the human treshold.."
> 5)So my question is Does high sampling rate also reduces the delays and improves the timing (not asking wheather we discern it or need it)?



Mark Waldrep does NOT in fact say that, he states that someone at Meridian told him that but that he doesn't agree!

5. It improves the timing and delays (and everything else) above the start of the filter of lower sample rates. At 44.1 for example, the filter typically starts around 20kHz to 21kHz, so a 96kS/s sample rate improves everything between about 20kHz-21kHz and say 45kHz (or wherever the start of the filter is). A 192kS/s sample rate should in theory improve everything above the filter start of 96kS/s, say between 45kHz and about 90kHz.

G


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## castleofargh

HAWX said:


> Hi people!
> 
> First of all, In my opinion, with todays technology, more than 16 bits and 44.1 kHz does not provide any descernable diffrence. My further understandings are only about the digital audio format, not about If "High Res" is necessary for music reproduction. You can think about the signal is produced by a computer, and decoded by other computer, no playback in real life.
> 
> ...



for you questions, there might be a need to precisely locate where in the playback chain you're looking at the data.
because if increase fidelity and just cover that added fidelity with noises or filters, then what the increased bit depth or sample rate does is irrelevant. 

for 1) because of the nature of waves, a tone with less bits can always be looked at as being less precise in amplitude, or as always being correct+some noise. same idea as having 2 instruments playing at the same time. you can imagine the second ruining the fidelity of the first instrument, or you can imagine 2 perfectly fine instruments together. which is closer to the way we listen to sounds. so increased bit depth in that context doesn't change the instrument but simply reduces the extra noise that comes with it .
when you change the volume level digitally, each sample has a value and you change that value so that the DAC will now read everything as a lower amplitude(quieter). it means attributing a new value to each sample and in a discrete system you have to go for the closest approximation. in 24bit you'll have an available approximation that is closer than in a 16bit sample. I guess that is what you might mean by loudness precision? 
of course any extra noise or distortion added by the DAC/amp/headphone end up being much louder that -144db below signal, so in practice you can say that those benefits are most likely buried under bigger errors anyway.

for 3)  a 2khz wave absolutely does now need to be recorded at 192khz to be better. the erroneous intuition that adding more points equals to more precision is mostly irrelevant because that's not how we reconstruct a signal. 
well it is if you use a NOS DAC with no filter, but fidelity on those things is bad as they disregard the very math that made digital audio possible so they become aliasing generators. 

for 5) timing can mean a all lot of things. correct pitch, jitter, phase, how high in frequency we can go... 
if you consider more than the audible range, then you can argue that high sample rate does a lot of things obviously. but if you stick to the audible range, stuff like the filters used, oversampling or reclocking might become as relevant or more relevant than highres music. you need to clearly define what you're talking about and where in the playback chain you're looking at it. by not doing that on purpose, a lot of unicorns are sold by talented marketing guys every year.


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## protoss

gregorio said:


> If you've read the original post then you'd realise that however many more bits beyond 16 you go is just wasted. For the consumer distribution of music, even 16bits is more than enough.
> 
> 
> 
> ...



Strange. I always thought and still do, that 32bit will have a lower distortion rate with perfect accuracy and higher bits of information store into the file than 24bit and 16bit. 

I know that humans could only hear upto 124db. But 32bit has upto 200db into its files. Even thou we can not hear that 80db. I in theory will say 32bit will have a better studio quality than 16 and 24 bit. 

I will also bet just like VHS to DVD to Bluray. We will eventually get that technology to decipher the audio out of that 32bit in our headphones. And later we will never look back


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## castleofargh

protoss said:


> Strange. I always thought and still do, that 32bit will have a lower distortion rate with perfect accuracy and higher bits of information store into the file than 24bit and 16bit.
> 
> I know that humans could only hear upto 124db. But 32bit has upto 200db into its files. Even thou we can not hear that 80db. I in theory will say 32bit will have a better studio quality than 16 and 24 bit.
> 
> I will also bet just like VHS to DVD to Bluray. We will eventually get that technology to decipher the audio out of that 32bit in our headphones. And later we will never look back


now take your db talk and try imagining them being db spl. how loud do you figure the band will be playing when recording? 120db spl? certainly not. 
what is the noise level in the studio? let's say they have a real quiet one and it's only 20db spl. anything below is ruined by that noise. 
what is the noise level and dynamic of the microphones used?  here is are a few examples but you can go look at the specs of many professional microphones. http://www.gras.dk/dynamic-range
now let's have fun, what is the dynamic range of your DAC? spoiler it isn't even 24bits and probably never will be. 
what is the distortion level of your DAC, amp and headphone? no please, don't cry, it's going to be ok. 
how loud do you listen to the song?  120db spl? I hope for you that you don't.

24bit files are already questionable for playback purpose. 32bit files are just stupid. there is nothing to gain from having them. not anything. 


about "I know that humans could only hear upto 124db" this doesn't mean what you think it means. they take a person, put him in an anechoic chamber and test the quietest sound he can notice, which is very low only because he's in that anechoic chamber.  then they crank up the volume level up to the point where it's physically painful for the guy and here you go 120 of possible dynamic. 
but the instantaneous dynamic range of the human hear is on average closer to 60db. that's the range we can detect between the loudest and quietest sound at the same time in the song. once the sound reaches a given level, the protection mechanism dampens the eardrum making it less sensitive, so you can listen to louder sounds, but you no longer have the ability to notice the close to 0db spl you could do in total silence. 
and of course it's not like you'll be listening to music in your own anechoic chamber, so don't dream about that 124db value as if it is what you'll notice in a song. for starters you'd have to listen at 124db+whatever noise level in your room to really get 124db of fidelity in the signal. good luck with that too. 
humans are extraordinary creatures, but still very human. the more is better philosophy becomes irrelevant when you apply it only to 1 element of a long chain. 

the race over video still kind of make sense on a huge screen with high contrast and gamut, because the eye still has the ability to notice the improvement if the screen can produce them. same thing can't be said for 16 vs 24 bit where people fail blind tests. if we rely on such tests, we can say with high confidence that 16bit is already transparent to the human hear when listening to music.


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## bigshot

In the recording studio, dynamic range and signal to noise are most important with vocals, because that's where the heavy duty compression is used. A flat recording has dynamics to spare, but when you start compressing vocals to make consonants clear over backing instrumentals, noise can get pulled up along with the subtleties of phrasing. That's why the cleanest performing amps and processors in a studio are the microphone pre's and the noise gates on the vocals. 24 bit allows a great deal of latitude to pull up sound buried in the mix. The sound files themselves aren't really the problem. The problem is how clean the signal coming in from the mic is.


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## protoss

castleofargh said:


> now take your db talk and try imagining them being db spl. how loud do you figure the band will be playing when recording? 120db spl? certainly not.
> what is the noise level in the studio? let's say they have a real quiet one and it's only 20db spl. anything below is ruined by that noise.
> what is the noise level and dynamic of the microphones used?  here is are a few examples but you can go look at the specs of many professional microphones. http://www.gras.dk/dynamic-range
> now let's have fun, what is the dynamic range of your DAC? spoiler it isn't even 24bits and probably never will be.
> ...



Well said. I will let others argue with you on this one.

cheers


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## StanD

protoss said:


> Well said. I will let others argue with you on this one.
> 
> cheers


Why would we argue with him? We agree with him.


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## protoss

StanD said:


> Why would we argue with him? We agree with him.



Just saying. Eventually someone will come and stir something up with different facts and ideas.Who knows.


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## TheTrace

Alternative facts, makes the forums go 'round.


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## gregorio

protoss said:


> [1] Strange. I always thought and still do, that 32bit will have a lower distortion rate with perfect accuracy and higher bits of information store into the file than 24bit and 16bit.
> [2] I know that humans could only hear upto 124db.
> [3] But 32bit has upto 200db into its files. [3b] Even thou we can not hear that 80db.
> [4] I in theory will say 32bit will have a better studio quality than 16 and 24 bit.
> ...



1. No, perfect accuracy is possible with any bit depth. All you get with higher bit depths is a lower digital noise floor. You did read the OP?

2. Yes and a human can go for about 2 months or so without food. A human can also run the 100m in 9.58 secs but who would buy a running machine which could only operate at that speed and who would choose not to eat for two months just for the entertainment value? The limits of the human body are extraordinary but you seem to be forgetting that music products are entertainment, so we're not talking about the utter limits that the human body can endure, we're talking about the limits of what's comfortable, which are way, way lower! For this reason you'd be hard pushed to find any commercial recording with a dynamic range greater than about 60dB and most have several times less.

3. Theoretically 32bit has up to about 192dB. I say "theoretically" because in practice this number is utterly ridiculous. If it were possible to create and play back such a file, so you could actually hear what was stored in the lowest of those 32bits, then the peak level would produce roughly the same sound pressure level as the initial blast wave of an atomic bomb. It's pretty safe to say that being instantly vaporized is significantly beyond the comfort level of consumers!
3b. Let's look at it the other way and say that peak level of our theoretical 32bit audio file is within the upper limits of the comfort zone, so the question then becomes; what is 192dB quieter than that? The answer is; that level is significantly lower than the level produced by two hydrogen atoms colliding, a level which probably couldn't even move a single air molecule, let alone move the countless billions of air molecules necessary to actually propagate a sound wave. So, you cannot hear it because there is no sound wave to hear!

4. How do you arrive at that theory? You think we're setting-off and trying to record nuclear weapons in our studios or trying to record a couple of sub-atomic particles colliding?

5. Almost but only if you equate them accurately: VHS would be equivalent to an old cassette, DVD to a vinyl record and the latest 4k UHD, HDR Bluray would be somewhat less than CD (16/44.1).

6. That's indeed true ... I imagine it would be rather tricky to "look back" (or anywhere else), once your head has been vaporized!! Even if it were possible, headphones which could kill you instantly wouldn't exactly be the ideal Christmas gift. It would make an interesting bundle though, along with a TV capable of outputting so much UV light it instantly fries you to a crisp. 

Crispy fried, vapourized audiophiles is quite an attractive proposition in some ways and if it were possible, I'd be sorely tempted to give some of them exactly what they're asking for!   ... I can only assume that you've been completely suckered by marketing and/or audiophile myths/anecdotes and have no idea what you're actually suggesting?!


protoss said:


> Eventually someone will come and stir something up with different facts and ideas.Who knows.


1. Science knows! Sure, some audiophiles (or more typically, those who sell equipment to audiophiles), come up with ridiculous ideas all the time and even sometimes present those ridiculous ideas as "different facts" but they're not really facts, they're just marketing bulls***. The basic facts of digital audio were invented 90 years ago, proven mathematically 70 years ago and no-one since has dis-proven them. In fact, doing so would invalidate the basis of all digital information theory and therefore demonstrate that no computer based technology works. This, along with most of the other facts in this post, were discussed in the OP, are you sure you've read it?

Hopefully, this is all starting to sound as laughably ridiculous to you as it does to us?

G


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## HAWX

Thanks @gregorio for answering no 5, what you say is logical. My English is not native thank you for clairty.

@castleofargh ..... I think I have already said "My further understandings are *only* about the digital audio format, not about* If "High Res" is necessary for music reproduction. You can think about the signal is produced by a computer, and decoded by other computer*, *NO PLAYBACK IN REAL LIFE*." regarding my understandings. You're still saying "spesify where you are looking at the playback chain" and limitations of dacs amps reconstruction and bla bla, *I'm not even talking about the playback of the file.*

I will repeat my understandings with asking questions and examples about 1 and 3.
1) For a given computer generated tone, (not captured with electronic devices) how presice the tone will be assigned to approximate dBFS without any dither comparing, 24 vs 16 bits? Let's say normally tone should be assigning about -50.5555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 dBFS. (100 digits after the decimal) 
I think the 24 bit assignation will have more digit after the decimal point thus leading to more precise assignation where the precision of a number is the total number of significant decimal (or other) digits.

3) For a given computer generated continuous tone which has the frequency of 10,000.55555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 Hz (200 digits after the decimal), how presice this tone frequency value assignation will be, comparing sample rates of 44.1kHz and 192 kHz, without any dither. (Although I don't think the dithering will make a diffrence) 
Again, I think the higher the sampling rate will be more presice to the original* computer generated* waveform. 

*I'm not asking this to get infromation about the playback fidelty of sampled file to human ears through dac/amp/headphones. My question was about math.*


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## gregorio (May 2, 2017)

HAWX said:


> 1) For a given computer generated tone, (not captured with electronic devices) how presice the tone will be assigned to approximate dBFS without any dither comparing, 24 vs 16 bits?
> 3) For a given computer generated continuous tone which has the frequency of 10,000.55555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 Hz (200 digits after the decimal), how presice this tone frequency value assignation will be, comparing sample rates of 44.1kHz and 192 kHz, without any dither.



1. It doesn't matter whether it's 24bit, 16bit or just 2bit, the precision is *infinite*. IE. -50.555r dB (infinite number of decimal places). This is a basic, proven tenet of digital audio, as explained in the OP. Obviously though, it's going to be far more difficult to discern that precision above the digital noise floor with only 2bits rather than say 16bits. Secondly, in practise we're not going to see such accuracy out of a DAC, just the analogue section is not capable of anywhere near infinite accuracy. The actual level of accuracy varies from DAC to DAC.

3. Again, the accuracy is *infinite* at both 44.1 and 192. This is true of any frequency up to slightly below the Nyquist Point (half the sample rate). This is also covered by the proven basic tenet of digital audio (the Nyquist-Shannon Theorem). So, your example of 10,000.5r Hz can be perfectly encoded (with infinite precision) with any sample rate exceeding about 22kS/s. Also again, we won't get this infinite precision out of a DAC in practise.

G


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## HAWX

Dude, sorry but* I think It is just impossible for any kind of finite file will be carriying infinite information.* I have already looked the original post and many of the external links given in this thread. I need proof, or It would be great to make test in some kind of program wich showes me what I'm talking about.

You are right about 44.1 kHz and 16 bits are more than enough for digital audio playback, which I also agree. But that's not I'm asking for. Thanks.


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## gregorio (May 2, 2017)

HAWX said:


> [1] Dude, sorry but* I think It is just impossible for any kind of finite file will be carriying infinite information.*
> 
> [2] I need proof ...



1. What you think is impossible is completely irrelevant! What's relevant is the established proven facts, which are completely unaffected by your personal suppositions and beliefs!

2. "_Intuitively we expect that when one reduces a continuous function to a discrete sequence and interpolates back to a continuous function, the fidelity of the result depends on the density (or sample rare) of the original samples. The sampling theorem introduces the concept of a sample rate that is sufficient for *perfect fidelity *for the class of functions that are bandlimited to a given bandwidth, such that *no actual information is lost* in the sampling process. ... The theorem also leads to a formula for *perfectly* reconstructing the original continuous-time function from the samples:
If a function x(t) contains no frequencies higher than B hertz, it is *completely determined* by giving its ordinates at a series of points spaced 1/(2B) seconds apart. 
A sufficient sample-rate is therefore 2B samples/second, or anything larger. Equivalently, for a given sample rate fs, *perfect reconstruction* is guaranteed possible for a bandlimit B < fs/2._" - Taken from the Nyquist-Shannon Sampling Theorem page of Wikipedia.

The bold emphasis is mine, the use of "perfect fidelity", "no actual information lost", "completely determined" and "perfect reconstruction" equals infinite information. If you require the actual proof, here's a link to the original 1948 paper by Claude Shannon (http://math.harvard.edu/~ctm/home/text/others/shannon/entropy/entropy.pdf) where the mathematical proof was published. The practical invention of digital audio was based on this proven theorem (which again was mentioned in the OP)!

You're free to believe whatever you like of course but if you're going to dispute the Nyquist-Shannon Theorem you're going to need a whole lot more than your personal belief, A Fields Medal and a Nobel Prize or two would be a good starting point!!

G


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## HAWX

@gregorio Wow, OMG!. Is this message from a professional audio engineer who's job is to know and utilise the digital audio. You have quoted a theorem which can be found on Wikipedia again and said digital audio is based on this theorem again. Didn't know that, thanks.

*Enough talking, Implement your claim as an expereinced professional audio engineer, or give way.*

Please post here the proof of a sine wave which's amplitute is -50.5555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 dBFS and frequency is 10,000.55555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 Hz which has encoded by 44.1 kHz and 16 bit bit depth audio file. 

Oh, and there's no "PERFECT" or "lossless" word, EVEN ONCE in the paper you sent me. You have made fun of me, I will make fun of you as well.

And I beleive you can encode a tone which has a frequency with a presicion to 100 million digits, take pi for instance, which plays for 1 second wav file and still takes 176.4 KB of storage. You will win a perfectly reconstructed Nobel Prize for that, _with infinite fidelety_.


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## castleofargh

HAWX said:


> Thanks @gregorio for answering no 5, what you say is logical. My English is not native thank you for clairty.
> 
> @castleofargh ..... I think I have already said "My further understandings are *only* about the digital audio format, not about* If "High Res" is necessary for music reproduction. You can think about the signal is produced by a computer, and decoded by other computer*, *NO PLAYBACK IN REAL LIFE*." regarding my understandings. You're still saying "spesify where you are looking at the playback chain" and limitations of dacs amps reconstruction and bla bla, *I'm not even talking about the playback of the file.*
> 
> ...


 if we look only at digital values which is missing the point of recording and playing back a sound, then sure you can use more bits to record a bigger number without any limit as to the size of those numbers. let's record Pi ^_^.  




HAWX said:


> Dude, sorry but* I think It is just impossible for any kind of finite file will be carriying infinite information.* I have already looked the original post and many of the external links given in this thread. I need proof, or It would be great to make test in some kind of program wich showes me what I'm talking about.
> 
> You are right about 44.1 kHz and 16 bits are more than enough for digital audio playback, which I also agree. But that's not I'm asking for. Thanks.


how about y=X² ? did we forget about math along the way?
digital reconstruction isn't based only on the precision and number of samples, we also use the fact that an audio signal is composed of sine waves. which have a behavior we know perfectly. with 2 samples we have crap as is, but if we add perfect band limiting where it counts, then Nyquist proved that we could reconstruct the signal.(we don't have perfect band limiting but as you want to stick to digital numbers without doing anything with them, I guess that's fine).
 I imagine that's where you reply that with 64bit samples we could reconstruct an even more precise signal than with 24bit. did I guess right?^_^
and I'd say yes in theory but no in practice. because how do you record the value at such a resolution in the first place? from the microphone? the preamp? the ADC? no real life situation allows for 24bit resolution of a recorded band. we only make containers for that resolution.


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## HAWX

@castleofargh Yeah, that's why I said computer generated. Not real life sound captured by mic.. and now *it's 3rd time saying it again btw..* 
Look I know math in basics. You migh contruct the fuction for the first 100 million digits of the pi number, though It won't be easy as y=X², and solving would not be instant by consumer computers today. We did not forget the math along the way. If Nyquist theorem applies fully as you say, read my post above, accomplishing what I have requested will solve my problem.


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## VNandor (May 2, 2017)

HAWX said:


> @castleofargh Yeah, that's why I said computer generated. Not real life sound captured by mic.. and now *it's 3rd time saying it again btw..*
> Look I know math in basics. You migh contruct the fuction for the first 100 million digits of the pi number, though It won't be easy as y=X², and solving would not be instant by consumer computers today. We did not forget the math along the way. If Nyquist theorem applies fully as you say, read my post above, accomplishing what I have requested will solve my problem.



Real life sound captured by a mic contains a sum of sine waves only, just as a computer generated tone. In fact, any kind of sound can be described as a sum of sine waves. If you think that is not true, I suggest you look up fourier-transform. 
Once a signal is band limited, it can be captured and replayed perfectly with the correct sampling rate.


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## HAWX

> Real life sound captured by a mic contains a sum of sine waves only, just as a computer generated tone. In fact, any kind of sound can be described as a sum of sine waves. If you think that is not true, I suggest you look up fourier-transform.
> Once a signal is band limited, it can be captured and replayed perfectly with the correct sampling rate.


Thanks, I'm saying computer generated because I want to discard recording chain limitations. I know the mathematical theorem, I just want to see an real life example in digital audio format for the conditions I provided above..


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## castleofargh

ok I think I get it, you're not talking about sound at all. your point is that a discrete value is discrete. well congrats on that, you're right. 

what Greg has been trying to say to you is that a 2khz sine wave with noise is a perfect 2khz sine wave, plus some noise. by increasing the bit depth what you effectively do is changing the noise content, not the 2khz sine. and when we use discrete values, the approximation of each sample creates extra noise.
as I've explained not long ago, you're free to look at the resulting amplitude and say the 2khz wave is wrong by the noise value. but it's also true that the 2khz is perfectly reconstructed and then we have some noise. the same way a piano and a singer are still a piano + a singer despite the signal being a single amplitude value.


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## bigshot

VNandor said:


> You won't get a "real life example" that discards recording chain limitations because in real life, recording chain limitations are always there.



But it doesn't mean that those limitations have an effect on human ears. Theory is always great in theory. But the ultimate goal is to make our sound systems sound good to us. That's the best thing to keep focused on.


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## gregorio

HAWX said:


> Oh, and there's no "PERFECT" or "lossless" word, EVEN ONCE in the paper you sent me. You have made fun of me, I will make fun of you as well.



There's no need for me to make fun of you, you're doing a perfectly good job of that all by yourself! ...
Section 19, Page 34 of the Shannon paper states:_ "If a function of time f(t) is limited to the band from 0 to W cycles per second it is *completely determined* by giving its ordinates at a series of discrete points spaced [1/2W] seconds apart in the manner indicated by the following result." _I can't paste the actual equations but you can read them for yourself. Hopefully you understand that "completely determined" effectively means infinite precision (unlimited decimal places)? This is the undisputed proof YOU requested!

Secondly, as a professional audio engineer I own and use industry standard professional audio tools. The signal generator I own is capable of setting a sine wave frequency in 0.1Hz increments and my DAW is capable of amplitude output adjustments in increments of 0.1dB. Scientific laboratory tools are needed for higher precision signal generation, not audio industry tools. Therefore, I cannot even generate the signal you are suggesting in the first place!  

Lastly, you really should read what you're replying to and understand what is being said before jumping to the conclusion that you're being made fun of. If you don't understand then ask but threatening personal attacks is a childish response!

G


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## HAWX (May 4, 2017)

I have asked you several times already. IF you don't know the value you can always say I DON'T KNOW. Really. You are talking too much but the work done is 0. You cost me too much time.

For the dBFS part, I have found it myself.
"For example, if the amplitude of the electrical signal being sampled ranges from -10 volts to +10 volts and we use one byte for each sample, each number does not represent a precise voltage but rather a 0.078125 V portion of the total range. Any sample that falls within that portion will be ascribed the same number. This means each numerical description of a sample's value could be off from its actual value by as much as 0.078125V -- 1/256 of the total amplitude range. In practice each sample will be off by some random amount from 0 to 1/256 of the total amplitude range. The mean error will be 1/512 of the total range.
This is called _quantization error_. It is unavoidable, but it can be reduced to an acceptable level by using more bits to represent each number. If we use two bytes per sample, the quantization error will never be greater than 1/65,536 of the total amplitude range, and the mean error will be 1/131,072." https://docs.cycling74.com/max5/tutorials/msp-tut/mspdigitalaudio.html

IF you give the "_ discrete points " perfectly _(which I wonder)_,_you will obtain the waveform perfectly. But I want to see that wheather the exact same case in digital audio from first hand. That's why I'm here.

@gregorio "Scientific laboratory tools are needed for higher precision signal generation, not audio industry tools. Therefore, I cannot even generate the signal you are suggesting in the first place! "  *Even I, far from audio engineering, can generate more presice tone than that!* *Here you go*: http://onlinetonegenerator.com/

*I am quite sure that just as that site does, a computer program can produce artificial waves for given loudness and frequency, and measure/inspect the samples for their Hz and dBFS. Does anybody know a program can do that?*


----------



## gregorio

HAWX said:


> [1] For the dBFS part, I have found it myself....
> This is called _quantization error_. It is unavoidable, but it can be reduced to an acceptable level by using more bits to represent each number.
> *[2] Even I, far from audio engineering,can generate more presice tone than that!*



1. What do you mean you found it for yourself, that's all in the ORIGINAL POST!!!! How does quantization error stop you from getting infinite resolution and how does it disprove the Nyquist-Shannon Theorem? For the last time, read the damn original post and ask questions if there's something you don't understand rather than trying to use your ignorance as a battering ram!

2. Please, tell me why I, as a professional audio engineer, ever need a signal generator with more precision than a 1/10th of a Hz and more precision than a 1/10th of a dB!

G


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## HAWX

gregorio said:


> 1. What do you mean you found it for yourself, that's all in the ORIGINAL POST!!!! How does quantization error stop you from getting infinite resolution and how does it disprove the Nyquist-Shannon Theorem? For the last time, read the damn original post and ask questions if there's something you don't understand rather than trying to use your ignorance as a battering ram!
> 
> 2. Please, tell me why I, as a professional audio engineer, ever need a signal generator with more precision than a 1/10th of a Hz and more precision than a 1/10th of a dB!
> 
> G



1st)I don't need to discuss it anymore. And I didn't said my findings compromises the Nyquist-Shannon Theorem.

2nd) It is quite sad that YOU DON'T KNOW OR YOU CAN'T EVEN THINK that is possible to create an audio file with more presicion than a tone generator, by using just a plain computer. I have never said you need to have. I expected you already knew or you could understood it by JUST THINKING.. You were the engineer who has extensive amount of experience and information about digital audio, remember?

You have made your point and said what you think, which is infinite presicion. OK. No need to repeat or argue it again and again. Cool?


Now for the other people, I need to see a sample in an audio file which can be assigned by infinite presicion, from first hand, or try it myself. For loudness part, a sample can be assigned to 2^16 value (16 bits), or 2^24 (24 bits) That's it. I am NOT talking about playback capability or noise floor or dither and all that. 

Now for demonstration I need to know "For a given computer generated continuous tone let's say a tone that has to be encoded with the frequency of 10,000.55555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 Hz (200 digits after the decimal), *how presice this tone frequency value assignation* will be, comparing sample rates of 44.1kHz and 192 kHz, without any dither. " Please provide me the results of tone which has assigned to this value in an audio file, in real life. Or recommend be some program where I can create this tone in various sample rates.
 I know this amount of presicion is completely unnecessary in digital audio world. Thanks.


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## Squeaky Duck

C'mon guys, this is a FRIENDLY forum to share ideas and knowledge, not bash on each other.


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## old tech

gregorio said:


> 1.
> 
> 1. Science knows! Sure, some audiophiles (or more typically, those who sell equipment to audiophiles), come up with ridiculous ideas all the time and even sometimes present those ridiculous ideas as "different facts" but they're not really facts, they're just marketing bulls***. The basic facts of digital audio were invented 90 years ago, *proven mathematically 70 years ago* and no-one since has dis-proven them. In fact, doing so would invalidate the basis of all digital information theory and therefore demonstrate that no computer based technology works. This, along with most of the other facts in this post, were discussed in the OP, are you sure you've read it?
> G


The fundamentals were mathematically proved long before that.  The discrete properties of sound were shown to be equivalent to a continuous wave form over a hundred years before Nyquist et al. Indeed, the mathematical properties of sound were sort of understood back in the days of Euclid.


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## old tech (May 2, 2017)

HAWX said:


> @gregorio
> Oh, and there's no "PERFECT" or "lossless" word, EVEN ONCE in the paper you sent me. You have made fun of me, I will make fun of you as well.


On that part you are correct, in a strict sense.  There is no perfect or lossless audio even in the natural world of acoustic physics. Acoustic sound has quantization properties if you drill down far enough into air molecules.  Whether that is relevant to human hearing is another matter.  One thing is certain, sound can never be lossless once it passes through a transducer (ie a change of energy state) converting air pressure to an analog electrical signal and then again, converting the analog signal to acoustic energy.  However, digital processing of the signal between these two necessary transducers is going to be more lossless than any analog processing, particularly processing that involves transmitting that signal over a distance or passing it through more transducers, eg tape heads, T/T cartridges etc.

https://www.st-andrews.ac.uk/~jcgl/Scots_Guide/iandm/part12/page2.html


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## spruce music (May 2, 2017)

If you wish to discuss accuracy of the frequency then 44/16 can do so down to 55 picoseconds which is in a around about way the accuracy limits of depicting frequency.  However, dither will decrease that number further.  24 bit could depict frequency to a finer level of accuracy.  A higher sample rate could as well.

So just for instance 16/44 digitally created could give you 1000 hz and 1000.000055 hz. Would it be possible to determine that difference in the analog world once you construct those signals.  Maybe though probably not.  Go to 24 bit and the noise in the analog realm will make the smaller differences something that can't be determined by the random motion of the air itself.  So certainly at 44/24  (and almost surely dithered 44/16) the ability to digital construct something exceeds the accuracy of its existence physically in air. Random air molecules aren't that stable.

So the practical limit is the bit depth in terms you are asking about.  With enough bit depth you can describe to any level of accuracy desired.  Sample rate increases however are not required. Within the nyquist band you can perfectly reconstruct the wave without needing additional bandwidth.  For audio purposes we have already exceeded the accuracy of what can physically exist.

As for software even something free like Audacity can generate waves I think to 6 decimal places.  That isn't the limit of what is possible just how far the software goes.  Matlab can construct whatever is mathematically possible.


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## spruce music

HAWX said:


> Hi people!
> 
> 
> What I learnt about more than CD quality, from this forum + a nice person in this forum:
> ...



Okay, backing up a bit to your earlier post. 

I gave an example of timing accuracy of digital at 44/16.  The shortcut formula is 
1/( number of levels x sample rate x 2pi).

So 1/ (65536 x 44,100 x 2pi) which is for redbook will give you a number of between 55 and 56 picoseconds. 

Higher number of bits will mean smaller steps between bits is possible and this increases the timing accuracy.  Why?  Sampled systems can reconstruct or interpolate the wave between samples accurately.  That is how a given sinewave can start in between samples and the time it began is accurately reconstructed in between samples.  Imagine a 1 khz sine wave that starts exactly during one sample.  The following sample instead of reading a zero reads some non zero amount.  Now imagine you move the starting point 10 microseconds after the first sample or less than half a sample time at 44,100 hz rates.  Now your second sample will read non-zero, but the level of that reading is lower than in the first instance.  For any given sinewave there is only one group of samples that fit.  So as further samples are taken only a starting time between samples will fit and could be reconstructed the same way.  So if it were redbook, lets us say you move the start time forward by only 1 picosecond.  The following sample in infinite precision would also be less, however the amount less will be an amount smaller than the step between the the least significant bit and the next one so it would be missed as having moved.  There is a small area of indeterminability about exactly when that sine wave began.  Now if I changed to 44/24 sampling the new timing accuracy is .2 femtoseconds or about .0002 picoseconds.  In this new instance moving the start of the sine wave forward by 1 picosecond will result in a change  which is larger than the least significant bit in a 24 bit system and the reconstruction of that sampled wave can also be done at higher timed accuracy at least in theory. 

Now yes, this is highly over-simplified and hopefully gets the point across without being out and out misinformation. 

So why the bit about perfect reconstruction.  Well the theorem demonstrates that and it is true.  The assumptions however are for infinitely long sample times, infinitely steep brickwall filtering and infinite sampling precision.  Other mathematicians have worked out that the same ideas about perfect reconstruction work with somewhat less stringent infinities involved. 

So people on this subforum get a bit touchy whenever people, which they regularly do, come in and try to tell us how the Shannon-Nyquist theorem isn't really true.  It is true.  It has been proven, and practical shortcomings of real world implementation have been rigorously worked out as well.  Things really do work that way.  So with reasonable sample rates and 24 bits the genuine accuracy of the digital system is in excess of what can be physically created due to noise and real world fluctuations that swamp quantization noise or timing limitations.


----------



## bigshot

old tech said:


> Whether that is relevant to human hearing is another matter.l



Since we all hear with human ears, I would say that relevance to human hearing is the only thing that really does matter. And since we all listen to music on our sound systems, I'd say that fidelity in music is more important than in abstract waveforms like square waves. Audiophiles can go off the deep end with "what ifs". It's better to focus on what makes music sound better on our stereos.


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## spruce music

bigshot said:


> Since we all hear with human ears, I would say that relevance to human hearing is the only thing that really does matter. And since we all listen to music on our sound systems, I'd say that fidelity in music is more important than in abstract waveforms like square waves. Audiophiles can go off the deep end with "what ifs". It's better to focus on what makes music sound better on our stereos.



Yes staying grounded in reality keeps one from going off the rails like a lunatic.


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## StanD

spruce music said:


> Yes staying grounded in reality keeps one from going off the rails like a lunatic.


Are you suggesting that the Internet forums are crowded with lunatics?


----------



## spruce music

http://audiosciencereview.com/forum/index.php?threads/finally-i-can-sell-my-stillpoints.1604/page-3

Try reading this thread starting at post #57.  Fractional picosecond jitter levels audible according to one fellow.  He proclaims clocks of 1 picosecond horrible and not fit for audio use.  Meets with a skeptical response. 

Seems kinda loony to me.


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## castleofargh

keep an open mind, maybe he trained in a secret temple in Tibet? as a kid in the theater on Wednesday I've seen many documentaries where individuals develop super human senses in such temples.
plus there are all the people living near Smallville or Central City, around radioactive spiders, toxic wastes... they don't register at a statistical level, but that doesn't mean they don't exist. 

 I remember seeing him in some video of RMAF or some show like that with flames on his shirt, and I remember thinking that if I had that shirt I would probably experience life differently too.


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## HAWX

spruce music said:


> Okay, backing up a bit to your earlier post.
> 
> I gave an example of timing accuracy of digital at 44/16.  The shortcut formula is
> 1/( number of levels x sample rate x 2pi).
> ...



TOP.
Big thanks for the in depth explanation. I know theorem will perfectly work If we measure thte samples coordinates perfectly, which will not be possible in real life thus leading to limitations.
I also do beleive for digital audio, 44.1/24 (or 44.1/16 dithered, altough I don't have extensive knowledge about dithering) is enough for today, and will be enough for many years to come, If not forever.

Thank you very, very much for clarification. That is one solid reply. Got the answer I wondered. Respects, sir.


----------



## StanD

spruce music said:


> http://audiosciencereview.com/forum/index.php?threads/finally-i-can-sell-my-stillpoints.1604/page-3
> 
> Try reading this thread starting at post #57.  Fractional picosecond jitter levels audible according to one fellow.  He proclaims clocks of 1 picosecond horrible and not fit for audio use.  Meets with a skeptical response.
> 
> Seems kinda loony to me.


That made for some interesting science fiction.


----------



## gregorio (May 4, 2017)

spruce music said:


> So the practical limit is the bit depth in terms you are asking about.  With enough bit depth you can describe to any level of accuracy desired.  Sample rate increases however are not required. Within the nyquist band you can perfectly reconstruct the wave without needing additional bandwidth.



This isn't really correct, or rather it's only partially correct. For it to be correct, the Nyquist-Shannon Theorem would have to be incorrect! You've got the sampling rate part right but not the bit depth part. From what you've said and from our last discussion, I suspect you've got all the information (or nearly all) but you haven't quite managed to fully join all the dots (excuse the pun) or put another way, you haven't fully appreciated all the implications. The part of the story which you appear not to fully appreciate is the statistical nature of digital audio, you appear to think about sample values as individual absolute values rather than as a series of probabilistic ordinates. A sinc function does of course take in the absolute sample values but effectively processes them statistically and part of this statistical process is dither.
It appears you only partially appreciate the purpose, usage and implications of dither. Let me re-phase what I stated in the OP: While the output of a DAC is a continuous waveform, the transfer curve "would be" the infamous stair-step (defined by the quantisation levels/steps). I say "would be" because that's not what happens in practise (if it did, it would NOT satisfy Nyquist-Shannon). What happens in practice is dither: The random modulation of the signal between adjacent quantisation steps/levels, this results in our stepped transfer curve becoming perfectly linear (statistically)! Let's take a worst case scenario, 1 bit. With one bit we only have two values: 0 (which is digital silence) and 1 (which is overload status). If we look at the result of quantisation in terms of these absolute individual values (a stream of individual values) then 1 bit of data gives us precisely zero resolution! However, if we look at these values as a statistical group (including dither) rather than as individual values, then we don't only have values of digital silence and digital overload but also *every* value in-between (IE. Infinite resolution)! The proof is SACD, which otherwise simply couldn't work at all (as it would have zero resolution).

Consider this quote from Hugh Robjohns (2011 SOS forum):

_"The myth of 'digital resolution' also bears a comment here. In undithered systems higher recording levels results in lower quantisation distortion, and some very early digital recording systems were not adequately dithered. ... However, this problem went away nearly thirty years ago, too. A properly dithered system (*as all now are and have been for decades*) has no quantising distortion at all. None. Consequently, the audio 'resolution' is 100% perfect regardless of recording level....

'Audio resolution' does not vary with recording level, nor the wordlength of the system. Only the absolute level of the noise floor varies with wordlength. And while we're at it, only the recorded audio bandwidth varies with sample rate, too."
_
I've highlighted that part because there appears to be a tendency to consider dither as somewhat of an optional extra which can be applied to digital audio, rather than as an essential ingredient. The application of dither is not optional, a dithering quantiser is employed in all ADCs (as I mentioned in the OP). Where some of the confusion about the application of dither probably lies is not in the quantisation process but any subsequent operator chosen re-quantisation process. As my OP and Hugh Robjohns mentions, resolution is infinite at any bit depth, IE. ABSOLUTELY ALL the information is there (as Nyquist-Shannon states). However, getting at all that information is the tricky part because some of it is buried in the resultant dither noise. The issue then is purely one of noise and has nothing to do with resolution. Just saying "noise" is not as simple as it first appears either though, because we've got all sorts of different noise, at various different points/places and with various statistical implications. For example we've got thermal noise, which tends to have a Gaussian distribution, acoustic noise which doesn't have a definable distribution, etc.

For the above reasons, your previous explanation/example of greater bit depth improving timing is not correct.



HAWX said:


> [1] For the dBFS part, I have found it myself. ... This is called _quantization error_. It is unavoidable
> 
> 2nd) It is quite sad that YOU DON'T KNOW OR YOU CAN'T EVEN THINK that is possible to create an audio file with more presicion than a tone generator, by using just a plain computer.
> 
> Now for demonstration I need to know "For a given computer generated continuous tone let's say a tone that has to be encoded with the frequency of 10,000.55555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555555 Hz (200 digits after the decimal), *how presice this tone frequency value assignation* will be, comparing sample rates of 44.1kHz and 192 kHz, without any dither.



1. That's utter nonsense, of course quantisation error is avoidable! For example, if you don't quantize (or re-quantize), you're obviously going to avoid any quantisation error! Is it just me or is this not ridiculously obvious?

2. Huh, what are you talking about? I'm saying both in the OP and my responses to you that an audio file has infinite resolution, you're the one arguing that it doesn't!

3. Going back to point #1. In your computer, get a 16bit signal generator and generate your sine wave, then write it to a 16bit wav file. Where's the quantisation error going to come from? How is there going to be any sort of error at all? The audio file can store anything you can generate, PERFECTLY! Maybe you want to use say a 64bit float signal generator though, in which case you would need to dither the output to a 16bit file to get infinite resolution. You can't say "no dither" because by doing so you are breaking the rules of digital audio which make it linear in the first place. It's like saying, what's the performance of an electric car if you don't give it any electricity?!

G

Edited


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## castleofargh

would you please be so kind as to edit your post and remove the really unnecessary personal comments.


----------



## spruce music

gregorio said:


> This isn't really correct, or rather it's only partially correct. For it to be correct, the Nyquist-Shannon Theorem would have to be incorrect! You've got the sampling rate part right but not the bit depth part. From what you've said and from our last discussion, I suspect you've got all the information (or nearly all) but you haven't quite managed to fully join all the dots (excuse the pun) or put another way, you haven't fully appreciated all the implications. The part of the story which you appear not to fully appreciate is the statistical nature of digital audio, you appear to think about sample values as individual absolute values rather than as a series of probabilistic ordinates. A sinc function does of course take in the absolute sample values but effectively processes them statistically and part of this statistical process is dither.
> It appears you only partially appreciate the purpose, usage and implications of dither. Let me re-phase what I stated in the OP: While the output of a DAC is a continuous waveform, the transfer curve "would be" the infamous stair-step (defined by the quantisation levels/steps). I say "would be" because that's not what happens in practise (if it did, it would NOT satisfy Nyquist-Shannon). What happens in practice is dither: The random modulation of the signal between adjacent quantisation steps/levels, this results in our stepped transfer curve becoming perfectly linear (statistically)! Let's take a worst case scenario, 1 bit. With one bit we only have two values: 0 (which is digital silence) and 1 (which is overload status). If we look at the result of quantisation in terms of these absolute individual values (a stream of individual values) then 1 bit of data gives us precisely zero resolution! However, if we look at these values as a statistical group (including dither) rather than as individual values, then we don't only have values of digital silence and digital overload but also *every* value in-between (IE. Infinite resolution)! The proof is SACD, which otherwise simply couldn't work at all (as it would have zero resolution).
> 
> Consider this quote from Hugh Robjohns (2011 SOS forum):
> ...



I don't believe I wrote anything that was incorrect if dither were not in use.  Perhaps you overlooked where I said I knew it was over-simplified.  I also don't know what discussion convinced you I don't know how dither works or what the results are or what that means.  Also despite you acting as if no dither isn't an option, plenty of software will allow you to do things without dithering.  It isn't a wise choice to make, but it happens. 

Trying to get the point across to someone who doesn't yet know is better done in important chunks in my opinion.  Otherwise it seems like so much indecipherable magic. 

Now if the previous rambling about this stuff made sense, and you don't get the seemingly magical claims for dither by Gregario, perhaps this will help. 

http://bitperfectsound.blogspot.com/2013/09/dither-some-hard-data.html

And the prior article that in simple terms explains what is going on. 

http://bitperfectsound.blogspot.com/2013/09/dither.html


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## HAWX (May 5, 2017)

For the orignal post writer @gregorio

Quantisation error is not avoidable in real life digital audio sampling. It's even distrubtion is possible by dither which reduces It's side effect, the quantisation noise, overall. But the assigning presicion is still reduced. Want to see from first hand? Try downsapmling the PCM audio file to 1 bits as you said. You hear quantisation noise of course. But what I'm mentioning is; 2^1=2, you have only 2 values to assign the sample points. The tones (asside from quantisation noise) on the file either will be copmletely silient (assigned to zero), or they will be at the same loudness. This proves that there is an error, and higher the bit depth=less the error will be.

As an engineer in diffrent field, I would be ashamed after a situation like this, instead of making word tricks and still trying to look like I am right. If somebody come and asked me about calculating the square root of 5, with 100 digits afther to decimal I would not say "I use industry standart calculator which is only capable of calculating 8 digits after the decimal." or "Why do I need to calculate for that amount of presicion?". I am the engineer and I am expected to provide a solution to the asker or simply give way. I would say: 
1) A good, simple, short,  I  DON'T  KNOW.  (Seems like somebody can't say this words and try a bit hard to know everything) + You might find the answer or useful infromation from this source or making a search on this ..... site. (Recommendation)
2) Here you go: http://www.ttmath.org/online_calculator (the exact thing they need)
3) There is no good third option and rambling and talk about a little diffrent topic to show how knowledge they got as some people do, which is basically time and focus loss. I haven't done that on purpose and will try to not do it in the future as an engineer.

Also I would not make jokes about the asker, because I know there is a possibility of the asker is working on some kind of digital coding that I don't or even can't think of, or a field I know and in that field is requeired to calculate with very high presicion. And I actually do know for air to air missles 16 digits after the decimal is not enough even in 1km (short) shots you have potential to miss the target which is in some kind of turn by 1.5 meter from just the growth of the mathematical  summing of the error.
Even If someone is really asking an insane number of presicion, I would try to show my interest in a good way like "wow that's super high! How/Why will you do with it !?". Not making fun of him/her. *Next time, please don't try to mask the capability of your knowledge by going 3rd option and making some word tricks.* Cause some people will understand what you are doing and they will have a judgement about your quality.



gregorio said:


> 3. Going back to point #1. In your computer, get a 16bit signal generator and generate your sine wave, then write it to a 16bit wav file. Where's the quantisation error going to come from? How is there going to be any sort of error at all? The audio file can store anything you can generate, PERFECTLY! Maybe you want to use say a 64bit float signal generator though, in which case you would need to dither the output to a 16bit file to get infinite resolution. You can't say "no dither" because by doing so you are breaking the rules of digital audio which make it linear in the first place. It's like saying, what's the performance of an electric car if you don't give it any electricity?



That's exactly can't saying the 1) "I don't know what will happen If the dither is not used" option but going the 3rd option and blaming me for asking a stupid question like "what's the performance of an electric car if you don't give it any electricity?". Really.

If the humanity were able to put and has process power to extract infinite amount of information when requested, instantly from a just a digital audio file, we wouldn't need supercomputers for simulation, we wouldn't need terabites of hdd, copmanies wouldn't invest millions of dollars to more calculation capable processing devices, and there would not be prizes for the calculation of the first xxx digits of the mathematical term "pi", for a spesific time and so on..

You always backed your thoughts with the Nyquist-Shannon theorem. Digital audio is not 100% equal to Nyquist-Shannon theorem. It's logic and operating principle is the same. The diffrence is that you (or should I say we?, lol) have predefined values for taken sapmles in computer world and also for digital audio. And these values are simply not perfect. Endless presicion values will requeire infinite amount of data. Just a tone's first 10 Million digits after decimal will cost you 10Million bytes which is about 10 megabytes which already exceeds  the 1 second wav file size.

......

*I am both happy and sad for the explosion of you and your 7 year old original post which you always directed as an religious book, even in diffrent sites.* I, as a person who don't have considerable expereince in electrical or signal processing or strong claim in digital audio like many here, am copmletely happy that my logic discerned you and some professional looking sources (eg:xiph's) explanations and many media and tutors on various sites are repeating what these sources say over and over again, can't be the whole story and there needs to be some imperfections other than dynamic range with bit depth and limitations for the taken sample point's requenciny presicion to some extent, with all the pressure from you and this fomus head-fi forum's various members who mostly identify themselves as real life audio engineers/tutors or people who gave their many years on digital and analog audio and understood almost all the concept by heart.

I am sad that how all this years, this  flawed, insuffcient explanation is standed strong from the most of the very best of these forum members (except @spruce music from what I saw while I was here, and there might be some others I didn't saw or realised, not including them) who supposed to have tremendous amount of knowledge, and how this flawed explanation faked many people many people. *IMPORTANT If you haven't read my older posts I am not talking about wheather the 16 bit audio is not enough for digital audio or not, I am talking about It's not perfection and It's explanation on the  original post and in the writers statements)*


----------



## castleofargh

@HAWX sorry but just no. 
you say to forget about recording and playback boundaries that would render your conditions impossible, then you argue about how 1 bit would sound... come on. disregard the DAC but care for how it sounds?

we're talking about the specific case of reconstructing band limited sine wave signal, you bring up computation processing???? Nyquist theorem is about sine waves, not about solving complex operations. this makes no sense and you end up talking about how you have the higher ground and are happy to have found the flaw in many people's logic. look in a mirror. 

 if you cared more about the theorem instead of repeating that you understand it, you would see how the finite limit you so dearly desire to demonstrate is right there before your eyes. band limiting! it is the hard limit that cannot be avoided and gives, if not a finite resolution(might have to define that more clearly), at least a finite content. if you move the band limiting, you lose data, or create aliasing. both with the direct consequence of non perfect reconstruction of the signal. if it is not band limited we cannot reconstruct it perfectly, even in theory. 
if you decide to assign less bit to code the signal, you increase the noise floor. interpret that however you like but that's all you do. the signal is still there and perfect in all the amplitude above the noise floor. 

and of course there are very real problems with practical application that would show obvious limits in resolution for multiple reasons, but you're the one insisting so much on disregarding real life encoding and playback boundaries. Greg has simply been following your conditions better than yourself. don't blame him for that. 
if you make up a theoretical case, stay theoretical. if you talk about real analog signal, you cannot disregard all the reconstruction steps in real life.

now as I said before, if all your posts were really about claiming that discrete values are discrete, you could have just said so and win the internet with a nice captain obvious meme. but if you're trying to demonstrate something else, I'm still sincerely wondering what it is.


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## HAWX

@castleofargh "you say to forget about recording and playback boundaries that would render your conditions impossible, then you argue about how 1 bit would sound... come on. disregard the DAC but care for how it sounds?"

I have given an axample about how the 1 bit sampling will loose the presicion of the sample's loudness. You don't have to play it back, but If you do, you will understand what I mean. You think that I'm going to say that 16 bit sounds better than 1 bit so that 24 bit does sound better than 16 bit? No, try to understand my explanation about how a sample's have lost the presicion of their loudness values relative to each other. Every sample will be assigned to either same loudness level or just 0. I have been talking about the REAL LIFE DIGITAL AUDIO SAMPLING whole time. You don't have to play the sound in real life, If you know how to measure it (I seriously don't) you will see what I mean. But If you decide to play it back you will have limitations of the analog chain I know *but you can still see* *WHAT PART I SPESIFCALLY MEAN FOR THE BIT DEPTH.*

"interpret that however you like but that's all you do. the signal is still there and perfect in all the amplitude above the noise floor. " That is, wrong. That's why I have given 1 bit example. You don't seem to understand what is the purpose of that example.

 "Nyquist theorem is about sine waves, not about solving complex operations." OK. You take a sample, and measure It's corrdinates. And you look It's coordinates and reconstruct it. Right? Now can you please tell me, how do you take perfect samples, please. Because what I am saying in short term is, that's not possible.

I know everything is band limited. Even If you have complete perfection for a  very short band limited area, you will still need infinite amount of data thus storage. There is no other way.

Rest of your writing is even more irrelevent. Please try to understand my writing fully.


----------



## gregorio (May 5, 2017)

spruce music said:


> [1] I don't believe I wrote anything that was incorrect if dither were not in use.
> [2] Also despite you acting as if no dither isn't an option, plenty of software will allow you to do things without dithering.  It isn't a wise choice to make, but it happens.



1. It's maybe just a misunderstanding, due to the fact that I didn't find it clear that your example and statements excluded dither and a few statements appeared not quite right, for example:
"_If you wish to discuss accuracy of the frequency then 44/16 can do so down to 55 picoseconds which is in a around about way the accuracy limits of depicting frequency.  However, dither will decrease that number further.  24 bit could depict frequency to a finer level of accuracy._" -
Dither will indeed increase that number further, to infinity. Undithered 24bit will give greater accuracy/resolution than undithered 16bit but dithered 16bit has more resolution than undithered 24bit. Dithered 16bit and dithered 24bit have exactly the same (infinite) resolution! lol ... And:
"_With enough bit depth you can describe to any level of accuracy desired._" - In an undithered system infinite resolution would require infinite bit depth, which is why it's always applied in the quantisation process.

2. It's not an option in the quantisation process. Any recorded/acquired signal will be dithered as all ADCs automatically dither and signals generated in the computer will either not incur quantisation or the processor generating the signal would be expected to internally dither if it does. It's only in the mixing/summing of those dithered signals where a "re-quantisation" process is likely to occur and where dithering that/those re-quantisations is optional. Therefore, I can't think of any commercial digital audio recording scenario in which dither is completely absent, although this doesn't exclude the possibility that at some points in the chain dither hasn't been applied to re-quantisation processes (typically because it's irrelevant) or hasn't been applied correctly.


HAWX said:


> Even If you have complete perfection for a  very short band limited area, you will still need infinite amount of data thus storage. There is no other way.



This is another way, it's called dither!!! That's why dither is a fundamental requirement of digital audio, because without it digital audio is not linear and to make it linear you would need infinite bit depth. That's what the original post explains and why the whole post is effectively based on dither. So, saying that if you eliminate dither the original post is incorrect is completely nuts! Of course it would be incorrect if you exclude dither, which is precisely why the OP does NOT exclude dither!!! Is it just me or are we well past the point of surreal? 

G


----------



## RRod

I'm a bit confused by all this. Was there some point in time where people were noticing all kinds of quantization artifacts from their favorite un-dithered CD releases?


----------



## StanD

RRod said:


> I'm a bit confused by all this. Was there some point in time where people were noticing all kinds of quantization artifacts from their favorite un-dithered CD releases?


If dithering is done in the ADC (Analog to Digital Converter) then the content of the CD is already dithered.


----------



## HAWX

gregorio said:


> Dither will indeed increase that number further, to infinity.


I don't agree. But I don't have in depth knowledge about dither to oppose or show a direct proof.



gregorio said:


> In an undithered system infinite resolution would require infinite bit depth


I have been saying basically the same thing, I agree.



gregorio said:


> 1.
> Therefore, I can't think of any commercial digital audio recording scenario in which dither is completely absent, although this doesn't exclude the possibility that at some points in the chain dither hasn't been applied to re-quantisation processes (typically because it's irrelevant) or hasn't been applied correctly.


I can think of non recorded, completely computer generated signals mixed together for elctronic music in computer. However, in real life recordings dither should be applied in the end, which everybody knows.


----------



## bigshot

HAWX said:


> [Please try to understand my writing fully.




I'd like to, but I don't see anything to indicate that what you are talking about would be remotely audible for human ears listening to a CD of music in their living room. Can you elaborate a little more on the audibility of what you are talking about? Theories are great, but I like listening to music better.


----------



## HAWX

bigshot said:


> I'd like to, but I don't see anything to indicate that what you are talking about would be remotely audible for human ears listening to a CD of music in their living room. Can you elaborate a little more on the audibility of what you are talking about? Theories are great, but I like listening to music better.



From my original post: "...... (not asking wheather we discern it or need it)?"
Sorry If you are seeking for discenrable diffrence by our hearing. Answer is obviously, no.


----------



## RRod

StanD said:


> If dithering is done in the ADC (Analog to Digital Converter) then the content of the CD is already dithered.



Well now we're just into the semantics of whether or not you are deliberately adding the randomness or letting natural phenomena do it. Either way, there was a time when people weren't working in high-bit processing chains that automatically dither, and yet I've never seen a single review from the 80s that said "man, these little crackly sounds at the end of reverb tails really annoy me!"


----------



## castleofargh

@HAWX 
we never said we could perfectly encode all sounds without any noise, in fact all of us have been insisting on noise a all lot and you do not seem to care or to get the implications.

also you can encode something in 1 bit and retrieve more than 6db of dynamic from it. delta sigma DACs and DSD work because that is a reality. when you encode in 1bit you create a crippling noise that is added to the signal 6db below. if you stop there then sure it looks bad. but given the right tool and sample rate, you can apply noise shaping to move that crippling noise into another frequency range. and what's left below the first 6db is not nothing. what's left is the original signal in all its glory, down to the point where some other noise is again added to it for whatever reason like the limit of the noise shaping ability or thermal noise or whatever. that's why I've been insisting on looking at things as proper signal+some noise. because the right signal is always there. just not always alone. ^_^


----------



## spruce music

gregorio said:


> 1. It's maybe just a misunderstanding, due to the fact that I didn't find it clear that your example and statements excluded dither and a few statements appeared not quite right, for example:
> "_If you wish to discuss accuracy of the frequency then 44/16 can do so down to 55 picoseconds which is in a around about way the accuracy limits of depicting frequency.  However, dither will decrease that number further.  24 bit could depict frequency to a finer level of accuracy._" -
> Dither will indeed increase that number further, to infinity. Undithered 24bit will give greater accuracy/resolution than undithered 16bit but dithered 16bit has more resolution than undithered 24bit. Dithered 16bit and dithered 24bit have exactly the same (infinite) resolution! lol ... And:
> "_With enough bit depth you can describe to any level of accuracy desired._" - In an undithered system infinite resolution would require infinite bit depth, which is why it's always applied in the quantisation process.
> ...



Is it just me or are we well past the point of surreal?


Yes you are.  

One issue repeatedly when explaining dither to someone who doesn't know is the confusion that infinite resolution would also mean infinitely low noise.  I understand what you mean, but what is the reply by Hawx about this?  It is exactly what I see over and over, they don't understand what you mean by infinite resolution.  So you stated it and the other fellow didn't understand it.  So you have conveyed pretty much no useful information to the person asking a question. 

Sometimes being too ticky picky to the Tee correct is an impediment to getting the point across.  I get that this idea should have long ago died about resolution of digital audio, but I also see that simply stating it is so doesn't get the job done helping people understand it.  Yes I was sloppy in my explanation, but it was in a way that to someone new to it they wouldn't even see the difference.  Yes it would have been better to be more careful. 

To reiterate, get someone to understand how digital sampling works without dither because it is simpler.  They can see the relation between sampling resolution and timing and noise.  Then next show how the dither works and they usually have the light bulb go off and see how you have finessed the resolution issue of limited bit depth and then they understand why it works that way.  Then with this idea in their mind in a way that is rationally understandable they are no longer snookered by some of the snake oil issues related to this situation.  Simply saying this is how it works trust me and don't question it doesn't work well with lots of people.  

You have the same problem with the ADC statements you made to someone who doesn't know how it works.  What you wrote will explain exactly nothing to them.  While it is all true and correct you haven't provided anything to someone who doesn't already know.


----------



## StanD

RRod said:


> Well now we're just into the semantics of whether or not you are deliberately adding the randomness or letting natural phenomena do it. Either way, there was a time when people weren't working in high-bit processing chains that automatically dither, and yet I've never seen a single review from the 80s that said "man, these little crackly sounds at the end of reverb tails really annoy me!"


Some folks like to split hairs or have overactive imaginations regarding audio tech, I think they are called ........... You can fill in the blanks.


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## gregorio (May 6, 2017)

HAWX said:


> [1] I don't agree. But I don't have in depth knowledge about dither to oppose or show a direct proof.
> [2] I can think of non recorded, completely computer generated signals mixed together for elctronic music in computer.
> [3] However, in real life recordings dither should be applied in the end, which everybody knows.



1. By saying you don't agree you ARE "opposing"! Did you not understand from the OP what dither does?
2. No, it would have to be even more specific than that, for 3 reasons: Firstly, the vast majority of electronic music is not purely electronically generated, it's at least partly based on samples (which have been recorded) and therefore had dither applied. Secondly, even if it is purely synthesized, it would need to be synthesised at a different bit depth, in order to incur a quantisation process and therefore any quantisation error and Lastly, if the soft-synth does incur a quantisation process it would have to do so without applying dither internally. While one could in theory create a piece of music which fulfilled all these conditions, I'm not sure in practise if there have ever been any commercial releases which do.


RRod said:


> Well now we're just into the semantics of whether or not you are deliberately adding the randomness or letting natural phenomena do it. Either way, there was a time when people weren't working in high-bit processing chains that automatically dither, and yet I've never seen a single review from the 80s that said "man, these little crackly sounds at the end of reverb tails really annoy me!"



It appears that some here still don't fully appreciate how integral dither is to digital audio, how integral it's always been and that it's ALWAYS been automatically applied. Some of the very earliest ADC stages generated dither with an analogue dither unit/stage (although they couldn't produce very accurate dither) but since the early '80's accurate digital TDPF dither was automatically applied by all ADC stages. So, the reason you've never seen a single review from the 80's about reverb tails cutting in and out is because there have never been any digital recordings without dither!

I think the reason for some of the confusion is that the situation changed at the start of the 90's, with the introduction of bit depths greater than 16bit, initially 20bit. Dither was still integral and automatically applied during the quantisation process (as it had always been) but working at 20bit introduced the new concept of a "re-quantisation" process, an additional quantisation process required to get from 20bit to CD (16bit). The application of dither for this re-quantisation process was a choice, initially: Truncation (no dither), Triangular PDF or Rectangular PDF. Then, during the early/mid 90's we saw the introduction of a new re-quantisation dithering choice, noise-shaped dither (Sony's Super Bit Mapping and Apogee UV22, for example). I suspect that this choice of re-quantisation dither is what is leading some to believe that dither is some sort of optional bolt-on extra just added at the end of the mastering process, rather than realising that it's integral to sampling theory (digital audio) and that all quantisers are dithering quantisers! (Though not necessarily all re-quantisers).



spruce music said:


> To reiterate, get someone to understand how digital sampling works without dither because it is simpler.  They can see the relation between sampling resolution and timing and noise.  Then next show how the dither works and they usually have the light bulb go off and see how you have finessed the resolution issue of limited bit depth and then they understand why it works that way.



Your response just seems to be extending the surreality even further! I did explain how digital audio works without dither, I did go on to explain how dither works and I did explain that we end up with a perfect measurement (infinite resolution) plus some noise, that the quantisation errors have been converted to noise, and I did go into considerable detail about the effects of that noise. My original post is pretty much ENTIRELY based on exactly what you're now suggesting 8 years later!?

G


----------



## HAWX

castleofargh said:


> also you can encode something in 1 bit and retrieve more than 6db of dynamic from it. delta sigma DACs and DSD work because that is a reality. when you encode in 1bit you create a crippling noise that is added to the signal 6db below. if you stop there then sure it looks bad. but given the right tool and sample rate, you can apply noise shaping to move that crippling noise into another frequency range. and what's left below the first 6db is not nothing. what's left is the original signal in all its glory, down to the point where some other noise is again added to it for whatever reason like the limit of the noise shaping ability or thermal noise or whatever. that's why I've been insisting on looking at things as proper signal+some noise. because the right signal is always there. just not always alone. ^_^




Yeah, you got my example. But I think I might have mistake. Practical limit can be 1/4 bit although It doesn't make sense to me, but still gives an indication about what I mean. I need to dig that  further. But I was not talking about the dithered and noise shaped case. By the way does DSD files work exactly on the same logic as PCM? Like say in 11.2MHz can you encode up to 5.6 MHz?


Does anybody has access to "
[SIZE=5][B]Resolution Below the Least Significant Bit in Digital Systems with Dither" [/B]from audio engineering society or somewhere else? Thanks.[/SIZE]

[QUOTE="gregorio, post: 13473223, member: 69811"]1. By saying you don't agree you ARE "opposing"! Did you not understand from the OP what dither does?
2. No, it would have to be even more specific than that, for 3 reasons: Firstly, the vast majority of electronic music is not purely electronically generated, it's at least partly based on samples (which have been recorded) and therefore had dither applied. Secondly, even if it is purely synthesized, it would need to be synthesised at a different bit depth, in order to incur a quantisation process and therefore any quantisation error and Lastly, if the soft-synth does incur a quantisation process it would have to do so without applying dither internally. While one could in theory create a piece of music which fulfilled all these conditions, I'm not sure in practise if there have ever been any commercial releases which do.
[/QUOTE]

1) My English is not top notch. I'm opposing but I don't have any real proof to show. But I will dig futher when I have time.
2) I already stated "I can think of." and gave a broad example. I didn't said there is a recording which have never seen dithering process either, which you examined the conditions.


----------



## sonitus mirus

HAWX said:


> Does anybody has access to "
> [SIZE=5][B]Resolution Below the Least Significant Bit in Digital Systems with Dither" [/B]from audio engineering society or somewhere else? Thanks.[/SIZE]
> 
> 
> ...



http://www.drewdaniels.com/dither.pdf


----------



## HAWX

sonitus mirus said:


> http://www.drewdaniels.com/dither.pdf


WoW  Wasn't expecting that fast. Thank you!


----------



## Yuri Korzunov (May 16, 2017)

castleofargh said:


> also you can encode something in 1 bit and retrieve more than 6db of dynamic from it. delta sigma DACs and DSD work because that is a reality. when you encode in 1bit you create a crippling noise that is added to the signal 6db below. if you stop there then sure it looks bad. but given the right tool and sample rate, you can apply noise shaping to move that crippling noise into another frequency range. and what's left below the first 6db is not nothing. what's left is the original signal in all its glory, down to the point where some other noise is again added to it for whatever reason like the limit of the noise shaping ability or thermal noise or whatever. that's why I've been insisting on looking at things as proper signal+some noise. because the right signal is always there. just not always alone. ^_^



Low bit number and noise shaping for signal-noise-ratio increasing is band reserve matter.


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## Yuri Korzunov (May 16, 2017)

protoss said:


> how about 24bit vs 32bit?   (32 bit 384khz)



Both bit and sample rate is potential noise matter.

If any parameter increasing is applied, need check result. There are many variables for univocal answer.


----------



## gregorio

Yuri Korzunov said:


> Both bit and sample rate is potential noise matter.



That sounds suspiciously like one of the most common tricks used in audiophile marketing BS! 

While your statement is literally true, does that "potential" actually apply or is it completely out of context? It definitely does apply in certain areas of the digital audio chain, for example, pro ADCs have for decades absolutely relied on manipulating/exchanging bit depth and sample rate for noise (and filtering) reasons. But, does your statement apply to distribution file formats? Answer: Absolutely not! It's not even vaguely close to being applicable to distribution file formats!

G


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## Yuri Korzunov

gregorio said:


> But, does your statement apply to distribution file formats?



What purpose of distibution?


----------



## gregorio

Yuri Korzunov said:


> What purpose of distibution?



I'm assuming the poster was referring to distribution to consumers. However, I can't think of any circumstances where distributing audio files at 32bit would make any difference, even in the case of distribution between audio professionals for further processing.

G


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## Yuri Korzunov

gregorio said:


> I can't think of any circumstances where distributing audio files at 32bit would make any difference, even in the case of distribution between audio professionals for further processing.



Which parameters difference you mean?


----------



## gregorio

Yuri Korzunov said:


> Which parameters difference you mean?



Any parameters!


----------



## Yuri Korzunov

I don't know such parameter.


----------



## castleofargh

Yuri Korzunov said:


> Low bit number and noise shaping for signal-noise-ratio increasing is band reserve matter.


sure, I wasn't trying to entertain the idea that noise shaping was magically creating bits at no cost. even alchemists didn't believe in making stuff out of nothing. 
DSD turns sample rate into more dynamic in a given frequency range, MQA uses some bit values to store more samples. but tricks only go so far.
my point was simply to say that a 1 bit encoding didn't necessarily condemned the final signal to have only 1 bit of dynamic.


----------



## Yuri Korzunov (May 16, 2017)

castleofargh said:


> sure, I wasn't trying to entertain the idea that noise shaping was magically creating bits at no cost. even alchemists didn't believe in making stuff out of nothing.
> DSD turns sample rate into more dynamic in a given frequency range, MQA uses some bit values to store more samples. but tricks only go so far.
> my point was simply to say that a 1 bit encoding didn't necessarily condemned the final signal to have only 1 bit of dynamic.



Noise shaping is system with limited stability. Because it have feedback.
When the system work about overload limit, it can come to unstable state: silence or contant sine(s). For back to normal state need reset noise shaper.

If noise shape have sloping transient from lowest to maximal level noise (for dithering, as example), we work far stability limit. But 1 bit on 2.8 MHz is hard in implementation for providing noise level close to 24 bit PCM.


----------



## gregorio

Yuri Korzunov said:


> But 1 bit on 2.8 MHz is hard in implementation for providing noise level close to 24 bit PCM.



Again, what you say is true but also again, what difference does it make? To put it in terms of the quote; Why do you need a consumer distribution file format to provide "noise level close to 24 bit PCM"? 

Yes, 1 bit DSD has problems, limitations and difficulties in implementation but AGAIN, just stating this is typical of marketing BS, because it ignores context, the fact that other aspects of DSD implementation are far easier/cheaper. And yes, there is potentially some instability but in practice that was overcome (beyond audibility) over 20 odd years ago. So in effect this is just like a typical audiophile marketing "red herring", a real problem that we/our product solves, which is simply a lie because either the problem, though real, does not actually affect digital audio or it's a problem which does affect digital audio but was already solved years/decades ago and should already be built into ANY competent product! 

G


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## Yuri Korzunov

gregorio said:


> Again, what you say is true but also again, what difference does it make? To put it in terms of the quote; Why do you need a consumer distribution file format to provide "noise level close to 24 bit PCM"?
> 
> Yes, 1 bit DSD has problems, limitations and difficulties in implementation but AGAIN, just stating this is typical of marketing BS, because it ignores context, the fact that other aspects of DSD implementation are far easier/cheaper. And yes, there is potentially some instability but in practice that was overcome (beyond audibility) over 20 odd years ago. So in effect this is just like a typical audiophile *marketing* "red herring", a real problem that *we/our product solves, which is simply a lie *because either the problem, though real, does not actually affect digital audio or it's a problem which does affect digital audio but was already solved years/decades ago and should already be built into ANY competent product!



What is marketing definition?


----------



## Arpiben

Dealing with noise shaping, the audio layman I am, always wondered about the added value of high order Sigma Delta Modulators in consumer DACs.
For sure,in theory, a 5 bit 17th order shaper with OSR=2048 may increase digital dynamic range around 50bits.
In a mixing or heavy DSP environment I may find some advantages. In a DAC perspective, I failed to understand the reasons of such range.
Marketing BS probably?


----------



## Yuri Korzunov

Arpiben said:


> Dealing with noise shaping, the audio layman I am, always wondered about the added value of high order Sigma Delta Modulators in consumer DACs.
> For sure,in theory, a 5 bit 17th order shaper with OSR=2048 may increase digital dynamic range around 50bits.
> In a mixing or heavy DSP environment I may find some advantages. In a DAC perspective, I failed to understand the reasons of such range.
> Marketing BS probably?



Order is not target there.
Order of noise shaper filter need for achieving steeper amplitude response of noise shaper filter.
It can increase signal/noise ratio in wider band of low frequencies.
But it can reduce stability of sigma-delta modulator to overload.

Need carefully increase steepness of the filter.
Easier way is increasing of sample rate.
It allow to decrease steepness of the filter to increase signal/noise ratio in wider band and keep the stability.


----------



## Arpiben

Yuri Korzunov said:


> Order is not target there.
> Order of noise shaper filter need for achieving steeper amplitude response of noise shaper filter.
> It can increase signal/noise ratio in wider band of low frequencies.
> But it can reduce stability of sigma-delta modulator to overload.
> ...



Relaxing filtering, reducing costs and avoiding instability all right,fine. But do we really need for example,discrete shapers running at 104MHz with high orders in a DAC ? That was/is my wonder.
Some pretend that by increasing noise shaping characteristics you increase depth perception....


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## Yuri Korzunov (May 26, 2017)

Arpiben said:


> Relaxing filtering, reducing costs and avoiding instability all right,fine.



17th order is too much, in my opinion for 104 MHz.
I'd refer to -170 ... -200 dB level, because it is give ability of "transparent for user" work with audio stuff.
Therefore enough 7...10th order approximately.
Need also account that precision used for calculations. If there 24 bit is used, we can decrease filter order for providing noise level comparable with quantization noise.
Here I can't call certain figures (need model the case). May be 5th order is enought.
Reducing precision and order may be reasonable for DAC, because after the DAC no other processings and DAC is limited by electronic component's noise level.


----------



## gregorio

Arpiben said:


> For sure,in theory, a 5 bit 17th order shaper with OSR=2048 may increase digital dynamic range around 50bits.
> ... In a DAC perspective, I failed to understand the reasons of such range. Marketing BS probably?



Typically, commercial music has no more than about 10 bits of dynamic range and DACs typically have about 17 - 20 bits of signal to noise ratio. Another 30 bits (180dB) of digital dynamic range beyond that is useful for only one thing, marketing!

G


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## Yuri Korzunov

gregorio said:


> Another 30 bits (180dB) of digital dynamic range beyond that is useful for only one thing, marketing!



What is marketing definition?


----------



## StanD

Yuri Korzunov said:


> What is marketing definition?


Selling unnecessarily expensive products to unsuspecting consumers that lack the knowledge and are susceptible to being tricked by hype (bigger is better).


----------



## Yuri Korzunov

StanD said:


> Selling unnecessarily expensive products to unsuspecting consumers that lack the knowledge and are susceptible to being tricked by hype (bigger is better).


No.


----------



## StanD

Yuri Korzunov said:


> No.


No what?


----------



## Yuri Korzunov

I wait for definition by Gregorio. He use the word too often, in my opinion.


----------



## spruce music

It would appear related to his quote.  Someone said an arrangement could result in 50 bits digital dynamic range.  With most devices limited by real use noise of about 20 bits, that leaves 30 bits marketing overkill (50-20=30 bits). 

Even 32 bits would appear to offer 192 db of dynamic range.  A number of news chips in new DACs are trumpeting their 32 bit DAC chips.


----------



## gregorio

Yuri Korzunov said:


> I wait for definition by Gregorio. He use the word too often, in my opinion.



As this thread discusses, 16bit is more than sufficient for a distribution file format. In the case of a DAC, to be safe we'd want another bit or so of digital dynamic range, to give some headroom for processing (over-sampling for example). There is even a potential argument for a DAC to accept and operate at 24bit int or 32bit float, for example, in the case of the consumer applying some processing such as EQ and/or HRTFs or adjusting digital volume and making sure truncation artefacts are always well below audibility. But, what is the practical benefit (or even potential practical benefit) of a DAC operating at 32bit int or in the example given, 50bit (or equivalent DR)? The answer is; no practical benefit whatsoever! Where there is no practical benefit, the only reason for operating at bit depths in excess 24/32float is marketing. This then is a definition of marketing; A feature of no practical benefit, included for the sole purpose of misleading consumers by stating/implying that the feature is of practical benefit. This broadly agrees with both @StanD and @spruce music.

G


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## Yuri Korzunov

gregorio said:


> Where there is no practical benefit, the only reason for operating at bit depths in excess 24/32float is marketing.



And what is marketing?


----------



## gregorio

Yuri Korzunov said:


> And what is marketing?



Already answered!

G


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## Yuri Korzunov

gregorio said:


> Already answered!


Give your definition: "Marketing is..."


----------



## gregorio

Yuri Korzunov said:


> Give your definition: "Marketing is..."



Are you trolling or did you not bother to read the post to which you are responding?

G


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## Yuri Korzunov

gregorio said:


> Are you trolling or did you not bother to read the post to which you are responding?



Why you avoid to give dfinition of "marketing" term? While I don't see correct definition of the term.


----------



## gregorio

Yuri Korzunov said:


> Why you avoid to give dfinition of "marketing" term?



I can only conclude from this response that the answer to my last question is: Trolling!

G


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## Yuri Korzunov (May 27, 2017)

gregorio said:


> I can only conclude from this response that the answer to my last question is: Trolling!



It is argument in the discussion?
I want correct using of terms.


----------



## Brooko

gregorio said:


> This then is a definition of marketing; A feature of no practical benefit, included for the sole purpose of misleading consumers by stating/implying that the feature is of practical benefit. This broadly agrees with both @StanD and @spruce music.
> 
> G



Yuri - he clearly stated his definition in his reply to you.  Why are you asking him to repeat it .....


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## Yuri Korzunov

Brooko said:


> Yuri - he clearly stated his definition in his reply to you. Why are you asking him to repeat it .....



Brooko,

Misleading is not marketing purpose. Misleading is illegal action. I don't like it too.

Marketing is one of terms, that have many definitions.

We can refer to Oxford dictionary, as example: https://en.oxforddictionaries.com/definition/marketing

But I heard very short and very exact, in my opinion, definition: "*Marketing is any relations with customers*".

It is spreading of any information, feedback, tech support, solving issues, product design direction, looking for new customers, work with customers, etc.
If you want long and successful relations with customers, misleading is not the best way for the marketing.


----------



## gregorio

Yuri Korzunov said:


> Marketing is one of terms, that have many definitions.



Correct and I gave one of those definitions. The definition of marketing I gave was the one to which my posts were refering, as you requested! Note that I stated "*a* definition of marketing" and not "*the* definition of marketing".

G


----------



## Yuri Korzunov

gregorio said:


> Correct and I gave one of those definitions.



Marketing is tool for bringing of ideas, products to people, not for "misleading".
Marketing is very useful thing for both manufacturers and customers.


----------



## Brooko

Yuri - I work in Sales & Marketing.  But that makes no difference.

You asked Greg specifically for his definition of Marketing - and he gave it in relation to what people claiming the benefits of 32 bit audio are claiming (and his version in this case is one I agree with).  He wasn't talking about that being the only definition of Marketing - just for this particular case.  At least that's how I read it anyway.

They are making claims for a standard that has no benefit - not that I can see anyway.  All seems to be is the classic case of bigger is better, the higher the number - the better it must be - right?  Except when there are no benefits.  Snake-oil is still snake-oil - no matter how your market it.  Unfortunately the gullible will still also be the gullible.  Its human nature.


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## Yuri Korzunov (May 27, 2017)

Brooko said:


> Yuri - I work in Sales & Marketing.  But that makes no difference.
> 
> You asked Greg specifically for his definition of Marketing - and he gave it in relation to what people claiming the benefits of 32 bit audio are claiming (and his version in this case is one I agree with).  He wasn't talking about that being the only definition of Marketing - just for this particular case.  At least that's how I read it anyway.
> 
> They are making claims for a standard that has no benefit - not that I can see anyway.  All seems to be is the classic case of bigger is better, the higher the number - the better it must be - right?  Except when there are no benefits.  Snake-oil is still snake-oil - no matter how your market it.  Unfortunately the gullible will still also be the gullible.  Its human nature.



Brooko, me seems most peoples now is no so gullible. Because currently many information around.

For learning 32 bit advantages, need take certain scheme, algorithm, measurements and learn all it in complex. Otherwise we have endless discussion like "what better DSD or PCM?"

There may be case when 44 kHz/16 bit implemented better when 24 bit/192 kHz. But is is not reason consider, that 44 kHz/16 bit is enough. Because other 192 kHz/ 24 bit device may have better features, than first 44/16 one due wider technical abilities for developers.

Also any audio implementation stumbled about "threshold of audibility". One person claim that it is enough, other claim that not. And may both are right. I don't know how to correct compare 16 and 24 bit even.

As other example "enough or not", let me consider ultrasound. We can check right at home that we don't hear sound after 16 ... 20 kHz (with proper equipment, of course).
But there is researches about brain response to ultrasound. And I don't know what will tomorrow.


----------



## StanD

Yuri Korzunov said:


> It is argument in the discussion?
> I want correct using of terms.


You could look that up yourself, the Internet has dictionaries, etc,, however, the below should define marketing in the context of this discussion.
"The development and implementation of a *promotional strategy.*" In this case the deceptive promotion of products that offer no tangible value, done for the purpose of sales/revenue.


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## Yuri Korzunov

StanD said:


> "The development and implementation of a *promotional strategy.*"



Promotional strategy is not mean "misleading".


----------



## StanD

Yuri Korzunov said:


> Promotional strategy is not mean "misleading".


Not necessarily, however, in the context of this discussion it surely is. All too often marketing is full of hype which is intended to mislead people. Sad isn't it.


----------



## Yuri Korzunov

StanD said:


> All too often marketing is full of hype which is intended to mislead people



"Too often" is how many in numbers and how it calculated?


----------



## StanD

Yuri Korzunov said:


> "Too often" is how many in numbers and how it calculated?


I'm not here to do a scientific study of marketing practices. If you are willing to fund that. I'm willing to give it a go. From a subjective standpoint, haven't you seen enough on the forums and adverts?


----------



## Yuri Korzunov

StanD said:


> I'm not here to do a scientific study of marketing practices. If you are willing to fund that. I'm willing to give it a go. From a subjective standpoint, haven't you seen enough on the forums and adverts?



Not all is so as look subjectivelly.
I red somewhere, that 97% of car drivers consider their skills above middle level (I don't remember exact figures, but there was funny proportion) 
Me seems, most of people doubt in advertising states.


----------



## StanD

Yuri Korzunov said:


> Not all is so as look subjectivelly.
> I red somewhere, that 97% of car drivers consider their skills above middle level (I don't remember exact figures, but there was funny proportion)
> Me seems, most of people doubt in advertising states.


Perhaps that 97% is the result of marketing and adverts for sports cars. Same thing for overpriced boutique DACs, perhaps 97% of audiophiles believe they can hear the difference. Never underestimate the power of imagination.


----------



## thasneakershop

StanD said:


> Perhaps that 97% is the result of marketing and adverts for sports cars. Same thing for overpriced boutique DACs, perhaps 97% of audiophiles believe they can hear the difference. Never underestimate the power of imagination.


yep probably just placebo


----------



## bigshot

I read about a study once a long time ago. They asked people if they thought they were going to heaven. 80% said yes they were. Then they asked them what percentage of other people did they think were going to heaven with them. They estimated 20%.


----------



## Yuri Korzunov

StanD said:


> Perhaps that 97% is the result of marketing and adverts for sports cars.



I don't thought about the topic same way.



StanD said:


> Same thing for overpriced boutique DACs, perhaps 97% of audiophiles believe they can hear the difference. Never underestimate the power of imagination.



I'd replace "belive" word to "hear".
I suppose, that people who "hear" (subjective approach), that one unit's sound better other unit's sound, are not more trusting to ads, than people who "like numbers" (objective approach).
If somebody "hear", that sound better, it is not result of advertizing always.

As developer, I rely to numbers only. Because, it is base, that may be used for further dialog and control of results.
However, numbers are nothing without accounting of psychoacoustics. And the psychoacoustics may be changed with time. 
Also "subjective" perception of better sound in unbelievable case (from objective point of view) may have pure technical reasons. Bugs, as example.

I'd use both approaches (subjective and objective) for checking of results two ways.


----------



## StanD

Yuri Korzunov said:


> I don't thought about the topic same way.
> 
> 
> 
> ...


I don't listen to crickets.


----------



## StanD

StanD said:


> Perhaps that 97% is the result of marketing and adverts for sports cars. Same thing for overpriced boutique DACs, perhaps 97% of audiophiles believe they can hear the difference. Never underestimate the power of imagination.





thasneakershop said:


> yep probably just placebo


Guided by chatter of the forum mobs, the adverts making the usual unusual claims, more bits are better, special filters, etc. contribute to the imaginative rendition of SQ.


----------



## Yuri Korzunov

StanD said:


> I don't listen to crickets.



But may be you heard one enough famous hypothesis why FLAC and WAV sound differently?


----------



## castleofargh

in the end it's a matter of perspective:
some tend to wait for evidence to believe in something.
some wait for evidence that it doesn't exist to stop believing. 
and some will reject anything that doesn't agree with them because thinking they are right matters more to them than the truth. 

and people are free to do what they like. but when we're discussing facts, only facts matter! where are the facts about 24bit sounding better than 16bit? many empty claims, a few uncontrolled anecdotes, people who fail to even make sure the 2 files are the same master, that one guy who will record cheat to pass his abx(haxorz exist in all games). and that's about it. 
24bit can store data with a lower noise floor, that is a fact and can be verified by anybody. but 24bit files sound different from 16bit, now as far as I know this is not a fact.


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## StanD (May 28, 2017)

castleofargh said:


> in the end it's a matter of perspective:
> some tend to wait for evidence to believe in something.
> some wait for evidence that it doesn't exist to stop believing.
> and some will reject anything that doesn't agree with them because thinking they are right matters more to them than the truth.
> ...


+1
Ultraviolet light exists, however, I can assure that I cannot see it. Is that like I need a higher sample rate? I can market special glasses, someone will buy it.


----------



## bigshot

The special glasses would have to convert the ultraviolet frequencies to a frequency in the visible spectrum if you are going to see it. It's the same with ultrasonic content. You'd have to transpose the ultrasonic sound down a few octaves to be able to hear it. But then you aren't hearing ultrasonic frequencies. You could transpose it in other ways than even harmonics, but that would just sound like distortion.


----------



## Yuri Korzunov

castleofargh said:


> where are the facts about 24bit sounding better than 16bit? many empty claims, a few uncontrolled anecdotes, people who fail to even make sure the 2 files are the same master, that one guy who will record cheat to pass his abx(haxorz exist in all games). and that's about it.
> 24bit can store data with a lower noise floor, that is a fact and can be verified by anybody. but 24bit files sound different from 16bit, now as far as I know this is not a fact.



Comparison 16 and 24 bit technically impossible.
Somebody suggets multiple 16 bit to 256 and compare both source and multiplied stuff at single 24 bit DAC.
But in both cases meanful (non-zero) music stuff played back in different bits. So we have different distortions. And result depend on implementation of the DAC.
Discuss right now, there are the distortions are audible or not, have no sense. Because need to learn each DAC separatelly.



StanD said:


> Ultraviolet light exists, however, I can assure that I cannot see it. Is that like I need a higher sample rate?



Higher sample rates were not used to ultrasound playback. It is analog filter low steepness matter, begining from ADC.


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## Yuri Korzunov (May 29, 2017)

bigshot said:


> The special glasses would have to convert the ultraviolet frequencies to a frequency in the visible spectrum if you are going to see it. It's the same with ultrasonic content. You'd have to transpose the ultrasonic sound down a few octaves to be able to hear it. But then you aren't hearing ultrasonic frequencies. You could transpose it in other ways than even harmonics, but that would just sound like distortion.



Some times ago I wrote the article "Ultrasound as Ultraviolet. Audio as Optics" even http://samplerateconverter.com/content/ultrasound-ultraviolet-audio-optics http://samplerateconverter.com/content/ultrasound-ultraviolet-audio-optics


----------



## StanD

Yuri Korzunov said:


> Higher sample rates were not used to ultrasound playback. It is analog filter low steepness matter, begining from ADC.


With that kind of attitude you will never become a proper audiophile and be welcome in certain forums. This can be head despite any of your arguments, ask anyone.
You will also disappoint the marketing department which is preparing adverts for sales of new ADCs


----------



## StanD

bigshot said:


> The special glasses would have to convert the ultraviolet frequencies to a frequency in the visible spectrum if you are going to see it. It's the same with ultrasonic content. You'd have to transpose the ultrasonic sound down a few octaves to be able to hear it. But then you aren't hearing ultrasonic frequencies. You could transpose it in other ways than even harmonics, but that would just sound like distortion.


I preferred the glasses that were advertised in comic books, the ones you could use to see whatever it was that you were looking for.


----------



## TYATYA

Bit depth? You guys use hybrid SACDs and feel the different when reading each layer? Yes! Different.

But Flac and mp3 256, I can classify. For a dts track, if transfer to wav, I hear the decrease of bass impact. 

For same bitrate, mp3 128 has more distort than aac64  sample. 

Number and spec somehow nomeaning in SQ some case


----------



## Yuri Korzunov

StanD said:


> With that kind of attitude you will never become a proper audiophile and be welcome in certain forums.



I suppose, that I'm audiophile - I love good sound  As manufacturer, I look for reasons of good sound.
While I don't stumbled such forums.



StanD said:


> This can be head despite any of your arguments, ask anyone.
> You will also disappoint the marketing department which is preparing adverts for sales of new ADCs



Advertizing strategy must show: why and who really need a product. It give confidence in manufacturer and easier sales.


----------



## StanD

Yuri Korzunov said:


> I suppose, that I'm audiophile - I love good sound  As manufacturer, I look for reasons of good sound.
> While I don't stumbled such forums.
> 
> 
> ...


Staying in business means creating more stuff to sell to unsuspecting consumers. The marketing plan is to hoodwink them. Simple as that. Even if one can make a better mousetrap, things cannot keep improving without an end in sight, after all we cannot. As humans we have limitations.


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## Yuri Korzunov (May 30, 2017)

StanD said:


> Staying in business means creating more stuff to sell to unsuspecting consumers. The marketing plan is to hoodwink them. Simple as that. Even if one can make a better mousetrap, things cannot keep improving without an end in sight, after all we cannot. As humans we have limitations.



It is too hard way of business. To implement it need many efforts and money.

There all simple: look for demands, create or modify product for satisfying the demand.
Customers ask about its demands via email, them write it at forums, social networks, etc. There no need invent a things, that you can sell.


----------



## gregorio

TYATYA said:


> Bit depth? You guys use hybrid SACDs and feel the different when reading each layer? Yes! Different.



Yes, it often is different but that's not the question. The question is, why is it different? Is it different because SACD or other "hires" formats are better or is it different because they've deliberately used different masters for the different layers? The answer is the latter, when using the same master no one can tell the difference between SACD (or other hires formats) and CD.

This isn't necessarily true of the lossy formats like MP3 but even then, if using one of the better versions it's virtually impossible to tell the difference.

G


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## Yuri Korzunov

gregorio said:


> Is it different because SACD or other "hires" formats are better or is it different because they've deliberately used different masters for the different layers?



And different hardware or use different circuits/processings of single harware unit.


----------



## gregorio

Yuri Korzunov said:


> And different hardware or use different circuits/processings of single harware unit.



Are you saying Sony deliberately crippled the "circuits/processings" of their SACD players when playing back the CD layer? Even if you are and it is true, that doesn't mean there is any audible difference between 16/44.1 and the SACD format, in fact there is overwhelming evidence that there isn't.

G


----------



## StanD

Yuri Korzunov said:


> It is too hard way of business. To implement it need many efforts and money.
> 
> There all simple: look for demands, create or modify product for satisfying the demand.
> Customers ask about its demands via email, them write it at forums, social networks, etc. There no need invent a things, that you can sell.


Apparently you haven't seen the adverts (many on this site) nor the numerous humorous articles that plague both the Internet and written media. I also get printed catalogs in the mail that are full of crazy statements designed to pick one's pockets.


----------



## Yuri Korzunov

gregorio said:


> Are you saying Sony deliberately crippled the "circuits/processings" of their SACD players when playing back the CD layer?



There may be different circuits. Different circuits, as rule, give different distortions.




gregorio said:


> Even if you are and it is true, that doesn't mean there is any audible difference between 16/44.1 and the SACD format, in fact there is overwhelming evidence that there isn't.



Do you have proper laboratory test results for such state? Give it us, please.


----------



## Yuri Korzunov

StanD said:


> Apparently you haven't seen the adverts (many on this site) nor the numerous humorous articles that plague both the Internet and written media. I also get printed catalogs in the mail that are full of crazy statements designed to pick one's pockets.



I said about business model, that proper for me.
In my opinion, business (people who do it) and customers are partners.
Total strategy must be [win - win].


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## bigshot

Just give me good music then. I'll be a happy customer. Sound quality is secondary, especially theoretical sound quality.


----------



## Yuri Korzunov

bigshot said:


> Just give me good music then. I'll be a happy customer. Sound quality is secondary, especially theoretical sound quality.



Sometimes my friends-musicians ask me to check their records.
It is possibly to listen at laptop instantly.
However, for better undersatnding details of the records need listen at "big home system". There may be absolutelly other perception of the similar music.

Personally, I have threshold of quality for listening. As example, I will not listen for pleasure at mobile phone speaker.
Laptop also is not suitable enough for this purpose.


----------



## gregorio

Yuri Korzunov said:


> Do you have proper laboratory test results for such state? Give it us, please.



"_Audibility_ of a CD-Standard A/DA/A _Loop_ Inserted into High-Resolution _Audio Playback". _Meyer and Moran, JAES Volume 55 Issue 9 pp. 775-779; September 2007



Yuri Korzunov said:


> I said about business model, that proper for me.



I was not referring to your personal business model, I don't even know what your product is or how you market it, I was referring to audiophile marketing in general.

G


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## Yuri Korzunov (May 31, 2017)

gregorio said:


> "_Audibility_ of a CD-Standard A/DA/A _Loop_ Inserted into High-Resolution _Audio Playback". _Meyer and Moran, JAES Volume 55 Issue 9 pp. 775-779; September 2007



I haven't access to the stuff.

Updated (I found the stuff at one site, I hope it is correct):
1. There compared direct DSD sound and the sound passed thru PCM DAC.
FIrst issue there that actually compare "how the DAC damage sound of the SACD-player".
It is not comparison of high resolution and 16/44 exactly.
Other the DAC or SACD-player implementation may give other results.




gregorio said:


> I was not referring to your personal business model, I don't even know what your product is or how you market it, I was referring to audiophile marketing in general.



"Marketing in general" don't exists.


----------



## StanD

Yuri Korzunov said:


> I said about business model, that proper for me.
> In my opinion, business (people who do it) and customers are partners.
> Total strategy must be [win - win].


I was discussing that the common marketing practices in the Audio (and many other) business reeks with deception.


----------



## Yuri Korzunov

StanD said:


> I was discussing that the common marketing practices in the Audio (and many other) business reeks with deception.



No "common marketing practices". Need consider each case separatelly.


----------



## StanD

Yuri Korzunov said:


> No "common marketing practices". Need consider each case separatelly.


Are you kidding? Read the adverts, articles, etc. I just love the garbage I get in the mail trying to sell me incredibly expensive cables that are supposed to do magic. Not to mention DACS, Amps, etc, with equally outrageous claims. Yes there is a common marketing practice, plainly seen by the overwhelming adverts, web pages, etc. If you wish to ignore all of this, that's fine for you, not for me. My audio junk mail goes straight from the letterbox to the garbage can, after I take a quick read and laugh.


----------



## Yuri Korzunov

StanD said:


> Are you kidding? Read the adverts, articles, etc. I just love the garbage I get in the mail trying to sell me incredibly expensive cables that are supposed to do magic. Not to mention DACS, Amps, etc, with equally outrageous claims. Yes there is a common marketing practice, plainly seen by the overwhelming adverts, web pages, etc. If you wish to ignore all of this, that's fine for you, not for me. My audio junk mail goes straight from the letterbox to the garbage can, after I take a quick read and laugh.



Do you really think, that all vendors are "bad boys"?


----------



## StanD

Yuri Korzunov said:


> Do you really think, that all vendors are "bad boys"?


Common practice mean most, not all. In this case very few play fair.


----------



## Yuri Korzunov

StanD said:


> Common practice mean most, not all. In this case very few play fair.



Do you like, if somebody will sell you something that solve your issues for reasonable price? I'd like.

Some publications in social networks, instantly generate comments: "where can I buy it?"
In this case I consider advertizing as useful information. I supose future marketing (including advertizing) should move in that direction.

Probably, more correctly targeted advertising will reduce annoying of people.


----------



## gregorio

Yuri Korzunov said:


> [1] It is not comparison of high resolution and 16/44 exactly.
> [2] Other the DAC or SACD-player implementation may give other results.



1. Yes it is, they took the SACD layer and compared that layer to the same version truncated through a 16/44.1 DAC. No difference could be detected.
2. They ran tests with other equipment. Still no difference could be detected!



StanD said:


> Are you kidding?



That's the obvious conclusion, although I phrased the question differently, asking if he was trolling.

G


----------



## Yuri Korzunov

gregorio said:


> 2. They ran tests with other equipment. Still no difference could be detected!



In tests number is matter. How many equipment (vendors/models) was tested?
Also need to know: what is skill of listeners for each test?
As rule, in conclusion statistics of group by skill is shown.



gregorio said:


> That's the obvious conclusion, although I phrased the question differently, asking if he was trolling.



It's not trolling. It's _marketing_!


----------



## gregorio

Yuri Korzunov said:


> [1] How many equipment (vendors/models) was tested?
> [2] Also need to know: what is skill of listeners for each test?



1. They used various setups of high-end audiophile equipment. They did not test all vendors/models, burnt-in vs not burnt-in or any other ridiculous/impossible suggestions! What they did do, was make the test relatively easy to pass, they just truncated to 16bit rather than noise-shaped dithering it. No difference could be detected! There is a response by Meyer & Moran somewhere, which goes into far more detail of all the equipment used. I'm sure it's fairly easy to find with Google.

2. All sorts of people; some pro engineers, some audiophiles, mostly ordinary students, if I remember correctly. None of them could detect a difference!

G


----------



## StanD

Yuri Korzunov said:


> In tests number is matter. How many equipment (vendors/models) was tested?
> Also need to know: what is skill of listeners for each test?
> As rule, in conclusion statistics of group by skill is shown.
> 
> *It's not trolling. It's marketing!*


So then, that's revealing.


----------



## Yuri Korzunov

gregorio said:


> They used various setups of high-end audiophile equipment. They did not test all vendors/models, burnt-in vs not burnt-in or any other ridiculous/impossible suggestions! What they did do, was make the test relatively easy to pass, they just truncated to 16bit rather than noise-shaped dithering it.



"Various" is not suitable term for proofs. There models should be shown obligatorily.



gregorio said:


> No difference could be detected!



Sorry for my English. But "No difference could be detected!" and "No difference be detected!" are different things, isn't it?



gregorio said:


> All sorts of people; some pro engineers, some audiophiles, mostly ordinary students, if I remember correctly. None of them could detect a difference!



However, need official protocol of test.



gregorio said:


> There is a response by Meyer & Moran somewhere, which goes into far more detail of all the equipment used. I'm sure it's fairly easy to find with Google.



Ok. We wait for you show us sources, that open for everybody (not only for AES members).


Also in the test was compared similar audio stuff at one loudness level. I think, for more exact test need check similar musical stuff at different levels: as adjusted in analog form as in digital form.

Anyway thank you for information, that you give already.


----------



## bigshot

Yuri Korzunov said:


> Personally, I have threshold of quality for listening.




That's a shame because you'll never know about the greatness of Enrico Caruso or Arturo Toscanini.


----------



## Yuri Korzunov

bigshot said:


> That's a shame because you'll never know about the greatness of Enrico Caruso or Arturo Toscanini.



Old records are not quality matter.
But "big home system" play it better than mobile phone? Do you agree?


----------



## bigshot

I use my phone to play through my speaker system. It sounds just as good as my Oppo blu-ray player or media server. We're very lucky to live in an era where almost all digital players are as perfect as you can hope for. Speakers are better than earbuds by a large margin though.


----------



## old tech

Yuri Korzunov said:


> "Various" is not suitable term for proofs. There models should be shown obligatorily.
> 
> 
> 
> ...


The methodology, along with various post test Q&As are available.

With regards to the audiophile test subjects, what was interesting is that they were allowed to do the ABX tests in their own homes, on their own stereos and in their own time - thus negating the [misplaced] criticisms of ABX testing you hear from subjectivists.  None were able to pick the high res source from 16/44 at a rate greater than a fair flipping of a coin.

Bear in mind this was just one of the many tests on this subject, like as in the links below:
http://archimago.blogspot.com.au/2014/06/24-bit-vs-16-bit-audio-test-part-i.html
http://archimago.blogspot.com.au/2014/06/24-bit-vs-16-bit-audio-test-part-ii.html


----------



## Yuri Korzunov

old tech said:


> The methodology, along with various post test Q&As are available.
> 
> With regards to the audiophile test subjects, what was interesting is that they were allowed to do the ABX tests in their own homes, on their own stereos and in their own time - thus negating the [misplaced] criticisms of ABX testing you hear from subjectivists. None were able to pick the high res source from 16/44 at a rate greater than a fair flipping of a coin.
> 
> ...



Thank you for links.

1) I respect Archimago. He do many good and very interesting researches.
This experiment carefully done *in its conditions*, that carefully described. Conclusions are done properly.

2) *I don't claim that 24 bit better 16 bit.*

3) I think, that *comparison of 16 and 24 bit is technically impossible.*

4) I consider *proper* double blind test as correct estimation of *subjective* audio quality (that is last criteria, want you it or don't) via *objective* (repeatable result in certain conditions) method.


However, need consider some things.

By link compared playback of 2 formats of files: 24 and 16 bit.

1. There was checked many devices. But DAC use different circuits or different parts of circuits. *So this way compared implementations, not formats as itself*.

2. ABX test is not home experiment.
There need big number of measurements (several thousands though for proper calculations) and *group of certified observers, that carefully control experiment performing*.
Results should be fixed in the protocol, that each of observers sign.
Also need exact schemes for each experiment.

3. I don't know what to do foobar ABX plugin, when prepare files for comparison (you can see progress before it run).

4. There was not checked *depending of quality* for 16 and 24 bit on *level* for each of tested files.

_Example:
If you test dithering and signal have high loudness, very probably dither give nothing.
If you reduce loudness into file 40 ... 80 dB and increase amplifier loudness, there, probably, difference appear (depend on dither implementation and experiment conditions)._

5. Need control SPL (sound pressure level) in the listener ear point.

6. Different spectral distribution in test files can cause diffrent resuts. This issue need to learn.

7. All measurement tools should have proper precision, be certified and checked by certified organization.



*Resume:*
There are many subtle details for exact experiment.

16 or 24 or 32 bit is not our aim.

It is only our tools for implementations of *subjective improvemnents*, that should be *proved via objective methods*.

If 24 bit don't improve something in certain case, don't use it in the case.

Experiment conditions are first thing that need to know


----------



## bigshot

If you want to measure subjective improvements, a bottle of wine is the best piece of audio equipment you can buy.


----------



## Yuri Korzunov

bigshot said:


> If you want to measure subjective improvements, a bottle of wine is the best piece of audio equipment you can buy.



For me, this way have opposite results.


----------



## StanD

If you are ABX comparing with foobar or an equivalent app, using the same audio chain , then the volume is going to be equal. This would be the typical use case.
It is not hard to convert a 24 bit file to a proper 16 bit file. If you cheat, then you are fooling yourself. If you are conducting the test for yourself, then you have no interest in fooling yourself. Obviously, this case is not for publishing to a scientific journal but instead to satisfy personal curiosity or use in discussions.


----------



## Yuri Korzunov

StanD said:


> If you are ABX comparing with foobar or an equivalent app, using the same audio chain , then the volume is going to be equal. This would be the typical use case.
> *It is not hard to convert a 24 bit file to a proper 16 bit file*.



Do you know exact border here between "proper" and "unproper" and have exact proofs?




StanD said:


> If you cheat, then you are fooling yourself. If you are conducting the test for yourself, then you have no interest in fooling yourself. *Obviously, this case is not for publishing to a scientific journal but instead to satisfy personal curiosity or use in discussions.*



Why need discussion about non-exact experiments at a forums?

To get some hidden details, that need to know for more deep knowledges.

It is infinite process: when we get the hidden details, we discovery new hidden details.

Why you partisipate in the discussion?


----------



## gregorio

Yuri Korzunov said:


> 1. There was checked many devices. But DAC use different circuits or different parts of circuits. *So this way compared implementations, not formats as itself*. ...
> [2] Example: If you reduce loudness into file 40 ... 80 dB and increase amplifier loudness, there, probably, difference appear (depend on dither implementation and experiment conditions).



1. So, what's the implication of what are you saying: A. That 24bit and 16bit can/could be audibly distinguished and B. Different circuits are audibly distinguishable but that hundreds of ABX tests are invalid because both together (A+B) are not audibly distinguishable?

2. Yes, there are a number of ridiculous test conditions and test signals which can be generated to demonstrate audible differences. However, this thread addresses the entire range of commercial music/audio content and the range of consumer listening equipment/conditions, it does NOT address audio material or listening conditions which consumers either don't have the equipment to replicate or no sane consumer would ever want to replicate! Again, your "example" is *exactly* the type of irrelevant, misleading statement so typical of audiophile marketing BS. And please, don't use that as a cue to start your "what is marketing" nonsense again. Everyone here, including I suspect even you, knows precisely what I mean by "marketing BS"!

The rest of the points in your post are likewise either irrelevant or just semantics to try and defend/support some agenda you're trying to promote. It's not even clear what your agenda is, whether it's to do with some product you're selling or just trolling?

G


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## Yuri Korzunov

gregorio said:


> 1. So, what's the implication of what are you saying: A. That 24bit and 16bit can/could be audibly distinguished and B. Different circuits are audibly distinguishable but that hundreds of ABX tests are invalid because both together (A+B) are not audibly distinguishable?
> 2. Yes, there are a number of ridiculous test conditions and test signals which can be generated to demonstrate audible differences.



There not only number matter.
If you want calculate statistical values, let's collect big numbers.
Also you need consider all points, that I called.




gregorio said:


> However, this thread addresses the entire range of commercial music/audio content and the range of consumer listening equipment/conditions, it does NOT address audio material or listening conditions which consumers either don't have the equipment to replicate or no sane consumer would ever want to replicate! Again, your "example" is exactly the type of irrelevant, misleading statement so typical of audiophile marketing BS. And please, don't use that as a cue to start your "what is marketing" nonsense again. Everyone here, including I suspect even you, knows precisely what I mean by "marketing BS"!
> 
> The rest of the points in your post are likewise either irrelevant or just semantics to try and defend/support some agenda you're trying to promote. It's not even clear what your agenda is, whether it's to do with some product you're *selling or just trolling*?



If you claim something, be ready to prove it.
Let's show us *proper* results of researches in conditions of "usual consumers". It will be interesting.
All that I see before, is experements in certain condition. But it is not total proof for all posible cases.
We can look for experiments in other conditions for more full understanding of the topic.
Using of "selling", "trolling", "marketing", "misleading", etc. words can’t replace *objective* arguments.

If you state, what I "promote" something, let us know:
What I «promote» in discussion 16 bit vs. 24 bit?


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## gregorio

Yuri Korzunov said:


> But it is not total proof for all posible cases.



Nothing ever is, but thanks for playing the "let's figure out some marketing BS" game. Ford could market their Fiesta as accelerating from 0-100 faster than a Lamborghini, which given "all possible cases" would be true ... For example, if you drove a Ford Fiesta off a very high cliff!   
As far as any sensible conditions/cases are concerned, there is plenty of supporting evidence and as requested, you've been provided with some of it. Not to mention, that the OP explains why you can't hear any difference in any sensible/reasonable consumer conditions.



Yuri Korzunov said:


> If you state, what I "promote" something, let us know: What I «promote» in discussion 16 bit vs. 24 bit?



You are not reading again, I stated that I don't know what your agenda is, what you're promoting or if you're just trolling! Either way, enough semantics, irrelevant misdirection and logical fallacies!!!

G


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## Yuri Korzunov

gregorio said:


> Nothing ever is, but thanks for playing the "let's figure out some marketing BS" game. Ford could market their Fiesta as accelerating from 0-100 faster than a Lamborghini, which given "all possible cases" would be true ... For example, if you drove a Ford Fiesta off a very high cliff!



There cars have different purposes. There more complex matter.



gregorio said:


> As far as any sensible conditions/cases are concerned, there is plenty of supporting evidence and as requested, you've been provided with some of it. Not to mention, that the OP explains why you can't hear any difference in any sensible/reasonable consumer conditions.



Do you want maximally _*objective*_ approach?



gregorio said:


> You are not reading again, I stated that I don't know what your agenda is, what you're promoting or if you're just trolling! Either way, enough semantics, irrelevant misdirection and logical fallacies!!!



If I try to say my opinion, that different with your opinion, it is «promoting» or «trolling»?


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## gregorio

Yuri Korzunov said:


> [1] There cars have different purposes. There more complex matter.
> [2] Do you want maximally _*objective*_ approach?
> [2a] If I try to say my opinion, that different with your opinion, it is «promoting» or «trolling»?



1. Misdirection! It doesn't matter, a Lamborghini is not faster than a Ford Fiesta under "all possible cases" but to market the Fiesta as faster under your premise would be misleading and any sane adult would recognise it as such.

2. You've been provided with evidence of objective approaches. Maximally or perfectly objective is virtually impossible to obtain, there is no absolute proof only a weight of evidence.

2a. No, it's promoting or trolling because you're apparently pushing some personal agenda which is off topic! The topic of this thread is not your personal, unique definition of marketing, nor is it about inappropriate and irrelevant conditions/cases, artificially manufactured for the sole purpose of demonstrating an audible difference. Sure, a consumer could reduce the file level by 40-80dB and turn their amp up by an equivalent amount and easily hear the difference BUT: How many consumers have amps capable of that and of those, how many have actually applied a 40-80dB digital attenuation to their entire music collection and whack up their amps to compensate? *Surely you must realise this is nonsense?*

G


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## Yuri Korzunov

gregorio said:


> 1. Misdirection! It doesn't matter, a Lamborghini is not faster than a Ford Fiesta under "all possible cases" but to market the Fiesta as faster under your premise would be misleading and any sane adult would recognise it as such.
> 
> 2. You've been provided with evidence of objective approaches. Maximally or perfectly objective is virtually impossible to obtain, there is no absolute proof only a weight of evidence.
> 
> 2a. No, it's promoting or trolling because you're apparently pushing some personal agenda which is off topic! The topic of this thread is not your personal, unique definition of marketing, nor is it about inappropriate and irrelevant conditions/cases, artificially manufactured for the sole purpose of demonstrating an audible difference. Sure, a consumer could reduce the file level by 40-80dB and turn their amp up by an equivalent amount and easily hear the difference BUT: How many consumers have amps capable of that and of those, how many have actually applied a 40-80dB digital attenuation to their entire music collection and whack up their amps to compensate? *Surely you must realise this is nonsense?*



[1] Off topic.

[2] I claim nothing. You claim that is no difference between 16 and 24 bits. Who should show proofs?

[3] That is my profit in question "16 or 24 bit better"?


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## gregorio

Yuri Korzunov said:


> [1] Off topic.
> [2] I claim nothing. You claim that is no difference between 16 and 24 bits. Who should show proofs?
> [3] That is my profit in question "16 or 24 bit better"?



1. Hallelujah brother! Now if only you could apply your sense of observation to your own words ....
2. Read the response!! You asked for evidence and you've been supplied with evidence.
3. I notice you avoided answering the (bold) question. Why was that I wonder?

G


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## Yuri Korzunov (Jun 1, 2017)

gregorio said:


> Now if only you could apply your sense of observation to your own words ....



Do you want to discuss marketing of autos?




gregorio said:


> 2. Read the response!! You asked for evidence and you've been supplied with evidence.



You show us one research. We discussed it. Do you have something else?




gregorio said:


> 3. I notice you avoided answering the (bold) question. Why was that I wonder?



You don’t like suggested way to practically check audible differences between 16 and 24 bit? Ok.


Now answer to my question: what is my profit in question "16 or 24 bit better"?


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## gregorio

@Yuri Korzunov Thanks for the confirmation, my wondering is at an end!!

G


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## Yuri Korzunov

gregorio said:


> Thanks for the confirmation, my wondering is at an end!!



Looks like you will don't answer my question. And I will don't know where my money in "16 vs. 24 bit"


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## Brooko

[Mod Hat On]
Yuri - we're starting to get complaints about your posts in this thread.

Greg has listed his thoughts and his reasoning on 16 vs 24 bit in the first post - suggest you re-read it.  If you then have specific questions related to it - by all means ask.
Greg has answered your many questions several times - you just refuse to engage, often don't reply, or change the subject.  I'm not going to label it as such - but it could be construed as trolling.  I'm giving you the benefit of the doubt here - that it could be language barrier.
You question their assertions when people answer your questions - but you never give your own position or back it with fact.  Given your apparent knowledge in this area (_Member of the Trade: AuI ConverteR 48x44)_ I personally find this telling.  How about you state what you believe - and produce data to back it?
To the others making the complaints, the easiest way to stop this is to disengage.  If Yuri refuses to actually engage properly, and you feel that this is trolling behaviour - simply don't reply.

[/Mod Hat Off]

Personally I unsubbed from the thread a couple of weeks ago.  I recognised the behaviour - and simply decided that it wasn't worth my time staying subbed.  I am a lot happier - and Greg's original post is still there to refer to when people ask pertinent questions in other parts of the forum.


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## castleofargh

@ Yuri I don't know if it's a matter of "lost in translation" or if you just move the accuracy line out of measurable grasp just to make a point? we can't test it perfectly, I get some of the theoretical problems, like how we always just test one DAC, how we tend to turn one bit depth into another even if only with zero padding when we do abx to control the playback system, how many DAC designs don't even really deal with 16 or 24bit in the first place, and stuff like that. but if you start being picky about that level of testing, soon enough nothing can be tested, we know nothing with enough confidence and there is nothing to discuss anyway. math becomes the only possible subject because only math can reach absolute certainty. is that what you think this forum is about?

 remember we typically have to explain to people that digital audio is more than making a line between samples, and that there is such a thing as hearing threshold and noises. for the guy who has no understanding of waves, so most amateur audiophiles, that is the next frontier, the stuff they think has yet to be proved. for those false unknowns, the body of knowledge and the accuracy of the testing methods aren't that much of a concern. after all we can usually measure magnitudes below human thresholds.
IMO the 16/24 bit matter was answered decades ago and only ignorance and marketing make people believe they will hear something new with 24bit. and so we're back on the issues of marketing. if there weren't guys like Sony making BS graphs all the time and giving false expectations to consumers about high res, we could close such topics and focus on stuff that could really benefit audible music. and the maddening part is that Sony while selling high res everywhere, has been one the hissing kings for years with their sort of class D amps. they don't market anything factual, what you see in the ads is the file format the DAC can *read*, never the output resolution. how could we respect Sony's marketing when they do that? why should we trust anything they say from then on? and that's the general state of high res marketing those days, I just name Sony because it's such a big player in a wild range of gears and musics. but people are abused at all levels, files with a big ass container but unspecified content for high price. pseudo science talking about night and day differences while the test is tempered with on purpose. facts taken out of context about non music content to pretend like audibility of highres is proved in general...
 the marketing of high res stinks. of course you can find a few guys who did nothing wrong, and even guys who tell us that they sell some formats only because we ask for them but don't advise buying them. the basic honest guys making a living by selling what the consumer desires. but they seem to be the minority those days.

in practice I see people clipping their music all day long because of settings they apply from fear of losing some least significant bits, or the "bit perfect" illusion taken at face value. and to me this is emblematic of HIFI those days. people lost in surreal numbers, spending money out of fear because marketing just like news media told them what to be scared of. and if it's too low to be heard or measured, it's even better. the most frightening thing is always the one we can't see. let's not buy that DAC because it doesn't play 32bit, let's use only 24bit files or above just in case something something. but at the same time let's be blind to problems a hundred dB louder because we haven't been told to look there. let's get the 4 gigawatt amp from fear of not getting all the dynamic range but as a result we'll now listen to music with a 0.4db channel imbalance caused by the volume knob, and some really high noise floor from the unreasonable gain. this is what we get from high resolution wisdom those days.


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## Yuri Korzunov (Jun 2, 2017)

Brooko said:


> Yuri - we're starting to get complaints about your posts in this thread.
> 1. Greg has listed his thoughts and his reasoning on 16 vs 24 bit in the first post - suggest you re-read it. If you then have specific questions related to it - by all means ask.



@Brooko,

Done.
I don’t saw there noise level of electrical circuits of DAC, as very important detail in understanding of the matter.
There is nothing about dependency of distortion by signal/noise ratio.
There nothing about link between sample rate, FFT length and measured noise level. Using 6 dB per 1 bit is not correct in general case.
Don’t considered analog volume control ways.
There I saw simplified definition of dynamic range. *Dynamic range* is ratio of *energies* of signal and noise in accepted frequency band. It measured this way.

*All these things should be considered in complex. Otherwise we risk to miss something important.*

And I don’t consider any opinion, theory and experiment as last word in knowledges.

Tomorrow, we can get new information. And current theory may be altered.

*I don’t agree, I don’t disagree. I learn the resolution issue.*

Each my doubt aimed for better approach to correct understanding. First for myself.




Brooko said:


> Greg has answered your many questions several times - you just refuse to engage, often don't reply, or change the subject. I'm not going to label it as such - but it could be construed as trolling. I'm giving you the benefit of the doubt here - that it could be language barrier.



I don’t want to answer to questions in incorrect form.

I’d want to see answers to my questions. Especially after some states.

I consider words *«misleading», «marketing», «trolling», «nonsense» etc. unallowable as arguments and definitions in correct discussion* between forum members.

I hope to future *correct* ways of discussion.





Brooko said:


> 1. You question their assertions when people answer your questions - but you never give your own position or back it with fact. Given your apparent knowledge in this area (Member of the Trade: AuI ConverteR 48x44) I personally find this telling. How about you state what you believe - and produce data to back it?



My position «16 vs. 24 bit» is implementation matter. I write it above.

*For promotion of my audio conversion software matter «what is format superior» is not matter:*

16/44 is better? Ok. Let’s convert music library to 16/44 for saving space.
No? 32/768 is the best? Ok. Let’s convert music library to 32/768.
Do you like 1 bit? Ok. 

I participate in discussions, because *explanation of own attitude to other people allow to look to the attitude from new point of view*.
It’s very useful. Some discussions help me to write articles, give new ideas, information.

I hope, that I answer _all_ your quesions?


----------



## Yuri Korzunov

castleofargh said:


> @ Yuri I don't know if it's a matter of "lost in translation" or if you just move the accuracy line out of measurable grasp just to make a point? we can't test it perfectly, I get some of the theoretical problems, like how we always just test one DAC, how we tend to turn one bit depth into another even if only with zero padding when we do abx to control the playback system, how many DAC designs don't even really deal with 16 or 24bit in the first place, and stuff like that. but if you start being picky about that level of testing, soon enough nothing can be tested, we know nothing with enough confidence and there is nothing to discuss anyway. math becomes the only possible subject because only math can reach absolute certainty. is that what you think this forum is about?



@castleofargh,
I think, we can’t achieve 100% confidence. New experiments and theories may give us other vision to well known things.

Zero padding of 16 bit word to 8…24 bits, may use other circuits of DAC, than 0 … 16 bit. As far as I know, in some cases there may be touched different instances of sigma delta modulators at DAC scheme.
However, there may be other unknown me details.




castleofargh said:


> remember we typically have to explain to people that digital audio is more than making a line between samples, and that there is such a thing as hearing threshold and noises. for the guy who has no understanding of waves, so most amateur audiophiles, that is the next frontier, the stuff they think has yet to be proved. for those false unknowns, the body of knowledge and the accuracy of the testing methods aren't that much of a concern. after all we can usually measure magnitudes below human thresholds.



In audio we always stumbled in this «threshold audibility», that is out of our access. I know only experimental statistical (big numbers) way to check it. Because we can’t connect to other person perception.

Numers is matter of precision of results. As rule, measurements should have 3 ... 10 time more precision than measured value.
I.e. we should provide number of tests per participant according precision per participant.
Number of participants according precision per participants.

If results of tests (ABX-voting) show 48%, it is not mean that precision there 1%.
If we add 1 participant in low number of participants, results may be more 50%.

It is desirable for results of each experiment:
1. Be repeatable (we can repeat the experiment and get similar results)
2. Be proved via other experiments (alternative way to check results, check by other approach)




castleofargh said:


> IMO the 16/24 bit matter was answered decades ago and only ignorance and marketing make people believe they will hear something new with 24bit. and so we're back on the issues of marketing. if there weren't guys like Sony making BS graphs all the time and giving false expectations to consumers about high res, we could close such topics and focus on stuff that could really benefit audible music. and the maddening part is that Sony while selling high res everywhere, has been one the hissing kings for years with their sort of class D amps. they don't market anything factual, what you see in the ads is the file format the DAC can read, never the output resolution. how could we respect Sony's marketing when they do that? why should we trust anything they say from then on? and that's the general state of high res marketing those days, I just name Sony because it's such a big player in a wild range of gears and musics. but people are abused at all levels, files with a big ass container but unspecified content for high price. pseudo science talking about night and day differences while the test is tempered with on purpose. facts taken out of context about non music content to pretend like audibility of highres is proved in general...
> the marketing of high res stinks. of course you can find a few guys who did nothing wrong, and even guys who tell us that they sell some formats only because we ask for them but don't advise buying them. the basic honest guys making a living by selling what the consumer desires. but they seem to be the minority those days.





Currently (may be in the future appear new information), in my opinion, correct comparison of resolutions is technically impossible.
As example, by reason that I wrote in the post above (zero paddind case).
It is implementation matter only.

Example:
Probably, that pro audio card 16/44 will sound better than 24/192 cheap one. But it don’t prove superiority of 16/44.

Low obvious difference or no difference is reason of all these discussions.
However, I always wait for new information and keep open mind.

Yesterday I released new article (what want to write many time), where wrote, what I see as most effective way of further improving of audio quality.
Me seems this way we can achieve difference, that obvious almost for everyone.
http://samplerateconverter.com/content/where-limit-audio-quality



If somebody ask me «what target resolution for audio files conversion you recommend», I recommend check each system resolution for «the best sounding mode» (mode=sample rate+bit depth+PCM/DSD) by hearing. Because proper measuring hardware is unavailable, as rule.
As first goal I recommend to take maximal sample rate and bit depth. And sequentially listen all modes.
But it is matter of implementation (especially ADC and DAC analog filters), not format.
If it is interesting, details here https://samplerateconverter.com/content/how-improve-sound-quality


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## castleofargh

ok, first thing first. our posts on HeadFi are not scientific peer reviewed papers. we are not doing research. this section has an unfortunate name but that's it. I agree with several points you make, but they are relevant for research, maybe not for amateur audiophiles. ^_^

I'm fine with the scientific method in general, in fact I love it. any new piece of evidence should of course be included in the new model. and because we think that new evidence can always come up, we must also keep in mind the possibility that we probably don't know all there is to know on a subject or the testing methods related to that subject. maybe someone will have an idea, maybe a new technology will allow to measure something new that will lead to new discoveries. such is the way of science, a never ending game of guesses, discoveries, and experiments to disprove them. 
but this is not a valid reason to fall for an appeal to ignorance and start thinking that something is true simply because we haven't proved it wasn't. or because some people think they felt something special under ridiculous conditions and total lack of controls. when we have little to no evidence of something, it is wrong to make it significant just because we want it to be significant. that's not good science. where is the evidence that 16bit isn't enough at normal listening level? without it why should I care for more? 

also general statements are just that unless clearly specified otherwise. and what is pissing off a few friends here is that you react like you take everything literally. anytime someone says "humans have 2 arms" you post saying it's not true. you're right, not 100% of humans have exactly 2 arms, but the statement was never meant to be read that way. only you did.
it's the same for human thresholds, when we say human's audible range is 20hz-20khz, it's not my audible range, it's not yours, we all know that it's a general statement based on some average data. it's like saying the grass is green. if you start reading the sentence as an absolute claim that all humans can hear 19.9khz but not 20.1khz, don't blame other for thinking you're trolling them.


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## gregorio (Jun 2, 2017)

Yuri Korzunov said:


> [1] I don’t saw there noise level of electrical circuits of DAC, as very important detail in understanding of the matter.
> [2] Using 6 dB per 1 bit is not correct in general case.
> [3] Don’t considered analog volume control ways.
> [4] There I saw simplified definition of dynamic range. Dynamic range is ratio of energies of signal and noise in accepted frequency band. It measured this way.
> ...



I'm not sure if you didn't read or didn't understand castleofargh, particularly his last paragraph. I say this because you have fallen into that exact trap/fallacy! For example:

1. In practice, for the consumer, no it's not an important detail, it's largely a completely irrelevant detail. You can buy DACs for well under $100 whose electrical circuit noise is way below audibility.
2. Again, NO. As far as the consumer is concerned 6dB per bit is plenty correct enough in any general case, it's even correct enough for audio engineers. Yes, it would be more accurate to state that it's more like 6.02dB per bit, that the theoretical dynamic range of a CD (16/44.1) is 98dB rather than 96dB, that with TDPF dither applied to re-quantisation it's more likely to be 93dB or often less in the case of noise-shaped dither. For developers/designers/EEs these more accurate/correct figures are potentially very important but that's a specific case, not the general case!
3. What do you mean? Sure, I didn't consider ridiculous gain staging, if that's what you mean.
4. Dynamic range is a rather vague term with several definitions. The definition I gave was perfectly adequate for the topic.
5. Broadly speaking I would agree but in a number of your postings you're are NOT taking your own advice! Yes, noise sums. In some cases (some mixing circumstances for example) even 24bit is not enough but again, this is a specialist case, not a general case for the consumer or consumer distribution formats! This is another completely typical audiophile marketing trick; take a potential problem in a specific case and create "fear" (as castleofargh put it) in the consumer and therefore the need for some feature or equipment to address this "problem". It's a trick/falsehood/lie because the marketing does not mention that this specific case is one which the consumer will never encounter and therefore, that there's no benefit to the consumer in addressing it!
5a. Again, this sounds exactly like a typical audiophile myth and again, castleofargh already explained this point! The myth typically goes something like: "Science doesn't have absolute proof", "science doesn't know everything", "some new evidence might come to light which changes everything (and maybe proves the audiophile myths correct)". It's nonsense! The sampling theory was developed over 60 years ago, it wasn't just some postulate/idea but a theory backed up mathematically. Over the course of 30 years or so it was developed into a practical technology, a technology has been completely dominant for decades and many billions of them have been sold, the average household in developed counties probably owns about 10 or so DACs. And yet, despite the massive financial incentive, there's been no significant development in the sampling theory for about quarter of a century and even that development can be argued to in fact be of no real significance to the consumer!
6. Oh dear. Again, you have fallen into a typical audiophile trap (again described by castleofargh!). And again you have also failed to take your own advice (#5)! Not only haven't you considered "all the things" but the things you haven't considered are actually some of the most important things!! A fair amount of what you state in your article is therefore invalid, some of it is simply factually incorrect and some of it I can agree with. It's not as bad as many audiophile articles/reviews/testimonials, which are often almost entirely complete nonsense from beginning to end but, due to a serious flaw in your starting premise, a number of subsequent flaws which result from that and a number of incorrect facts, it's still IMO a quite poor article.

G


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## Yuri Korzunov

castleofargh said:


> I agree with several points you make, but they are relevant for research, maybe not for amateur audiophiles.



Accurate look to results of researches is automatical professional skill.
Just I listen «scientists proved», I instantly get mind «what is conditions?».
Because, when you implement algorithm or device, sometimes you can see that the theory don’t work.
It may be because any theory have limits of conditions.
Therefore, need to know conditions, where the claim work.

The topic «24 vs. 16» like an iceberg.
If somebody try show underwater part it is not trolling.
Of course, if nobody interests the underwater part, we can stop the discussion and view only upper part of the iceberg.


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## StanD

Yuri Korzunov said:


> Do you know exact border here between "proper" and "unproper" and have exact proofs?
> 
> 
> 
> ...


The math and concept for conversion is not difficult. You can find the answers with a quick consultation with Professor Google.
Why do I participate in the discussion? Are you the thought police questioning what I can do? Don't be naughty.


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## Yuri Korzunov (Jun 2, 2017)

gregorio said:


> I'm not sure if you didn't read or didn't understand castleofargh, particularly his last paragraph. I say this because you have fallen into that exact trap/fallacy! For example:
> 
> 1. In practice, for the consumer, no it's not an important detail, it's largely a completely irrelevant detail. You can buy DACs for well under $100 whose electrical circuit noise is way below audibility.
> 2. Again, NO. As far as the consumer is concerned 6dB per bit is plenty correct enough in any general case, it's even correct enough for audio engineers. Yes, it would be more accurate to state that it's more like 6.02dB per bit, that the theoretical dynamic range of a CD (16/44.1) is 98dB rather than 96dB, that with TDPF dither applied to re-quantisation it's more likely to be 93dB or often less in the case of noise-shaped dither. For developers/designers/EEs these more accurate/correct figures are potentially very important but that's a specific case, not the general case!
> ...



Ok. I understand _your personal opinion_.

3. Volume control like a shifting of digital level range by voltage axis.


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## Yuri Korzunov

StanD said:


> The math and concept for conversion is not difficult. You can find the answers with a quick consultation with Professor Google.
> Why do I participate in the discussion? Are you the thought police questioning what I can do? Don't be naughty.



Sorry, I nothing understand.


----------



## gregorio

Yuri Korzunov said:


> The topic «24 vs. 16» like an iceberg. ... If somebody try show underwater part it is not trolling.



It is trolling! It's trolling because the "underwater part" is tiny (unlike an iceberg!) and completely irrelevant for a consumer distribution format. And there have been countless formal and informal tests over a period of decades which have provided reliable supporting evidence (at least two of which you've been supplied with) and in that time, no reliable evidence to the contrary!

There has been some interest in that tiny "underwater part", it's already been mentioned in the OP and discussed in considerable detail later in this thread. So, you're just repeating what's already been discussed, which is also a type of trolling!



Yuri Korzunov said:


> Ok. I understand _your personal opinion_.



No, it's not just my personal opinion! castleofargh already explained to you, it's long established knowledge which was put to bed years/decades ago and the contrary position is only perpetuated by marketing depts, those fooled by the marketing and trolls!!!

G


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## Yuri Korzunov

gregorio said:


> No, it's not just my personal opinion! castleofargh already explained to you, it's long established knowledge which was put to bed years/decades ago and the contrary position is only perpetuated by marketing depts



Clear.

I hope to correct communication in the future.


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## VNandor

castleofargh said:


> ok, first thing first. our posts on HeadFi are not scientific peer reviewed papers. we are not doing research. this section has an unfortunate name but that's it. I agree with several points you make, but they are relevant for research, maybe not for amateur audiophiles. ^_^
> 
> 
> also general statements are just that unless clearly specified otherwise. and what is pissing off a few friends here is that you react like you take everything literally. anytime someone says "humans have 2 arms" you post saying it's not true. you're right, not 100% of humans have exactly 2 arms, but the statement was never meant to be read that way. only you did.
> it's the same for human thresholds, when we say human's audible range is 20hz-20khz, it's not my audible range, it's not yours, we all know that it's a general statement based on some average data. it's like saying the grass is green. if you start reading the sentence as an absolute claim that all humans can hear 19.9khz but not 20.1khz, don't blame other for thinking you're trolling them.



I think I've seen that somewhere before. If someone makes a generalizing sweeping statement without noting there are exceptions and the statement is not always true, I think it's fair if someone points out that the generalization is not always applicable. I know this is a lame example but imagine if you analyzed the effects of radiation exposure on humans. Now take your "people have two arms" statement into that context and you will soon conclude that aliens exist. The point I'm trying to make is that there will *always* be that 1% that renders *all* the statements that contain "always, every, all, never ever" and such to be false. Except maybe if we are talking about math. It's possible to make mathematical statements that are always or never true and not depending on the circumstances.

I'm not surprised people are trying to discuss that 1% in details because the first post seems to cover all the main points of digital audio and doesn't leave much room for discussion other than that.


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## Yuri Korzunov (Jun 2, 2017)

I suppose, most of music (significantly more 1% of audiophile music), that noted here, is jazz and classical. The pieces with long quiet fragments.
Me seems, it is important compare 16 vs. 24 bit in different levels of loudness into record (not volume knob position), normalized by sound pressure level (SPL) and at different SPL levels.
Of course, there should be checked number of devices, because it is not format comparison, but comparsion of device implementations in 16 and 24 bit mode.


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## sonitus mirus

Yuri Korzunov said:


> I suppose, most of music (significantly more 1% of audiophile music), that noted here, is jazz and classical. The pieces with long quiet fragments.
> Me seems, it is important compare 16 vs. 24 bit in different levels of loudness into record (not volume knob position), normalized by sound pressure level (SPL) and at different SPL levels.
> Of course, there should be checked number of devices, because it is not format comparison, but comparsion of device implementations in 16 and 24 bit mode.


From a practical use perspective, excluding pathological situations, if you can hear the noise floor in quiet passages of your music, the volume level is almost certainly at a level that would not be safe for your hearing.   An example of a pathological situation might be having a very large room where listeners are both near and far away from the transducer, where the volume level is way too high for those people nearest the transducer.  Yes, having a lower noise floor might be technically achievable using 24-bit vs 16-bit, but the remedy is not practical, and probably not a good idea for anyone.  If you can hear noise floor at any time, turn the volume down or move away for your own safety.


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## Yuri Korzunov

sonitus mirus said:


> if you can hear the noise floor in quiet passages of your music



I meant signal / [noise + distortions] ratio rather. Not audible noise floor.


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## bigshot

You get audible noise and distortion in digital music? What DAC are you using? That isn't the way it's supposed to be.


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## Yuri Korzunov (Jun 2, 2017)

I don't make the experiment. There is need a lot time.
As we understand, there will not big sound difference anyway.
So need big numbers for decreasing impact of "pair" of additional tests or participants.
Places: anechoic room, other kinds of rooms.

I see it so:
1. Create audio files with -40 ... - 50 ... -60 ... -80 dB average loudness in 24 and 16 bit.
Need provide same SPL for 24 and 16 bit mode of DAC with precision 0.1 dB.
As far as I know, ear sensitive to 1... 2 dB difference. So we should provide 3 ... 10 times more precision for measurements.
Listener listen only one simultaneously at one seat place. It need for avoid impact of speaker's pattern to audio perception.

2. Set volume at SPL level #1 (need look for certain values experimental way) for file.
3. Perform number ABX tests between 16 and 24 bit samples.
4. Repeat step #3 for SPL levels #2, #3, .... (step between SPL levels need to found experimental way, may be by learining of ear's features)
5. Repeat steps #2 ... #4 for rest files.
6. Repeat steps #2 ... #5 for rest listeners.
7. Repeat steps #2 ... #6 for rest audio systems (source+DAC+amp+speakers) / components.
8. Repeat steps #2 ... #7 for rest rooms.

Only after careful experiment and analysis of its results, we can suggest, what details is insignificant in common consumer's cases.

[post was updated]


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## StanD

Yuri Korzunov said:


> I don't make the experiment. There is need a lot time.
> As we understand, there will not big sound difference anyway.
> So need big numbers for decreasing impact of "pair" of additional tests or participants.
> 
> ...


Aw C'mon, you can publish the paper and get tons of hate mail from all the disappointed audiophiles.


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## Yuri Korzunov

StanD said:


> you can publish the paper and get tons of hate mail from all the disappointed audiophiles.



May be it show that 24 bit (implementation, not format) is better 

I don't know that will results of the experiment. May be there will 50/50. May be not.

P.S. Just now I will add additional details in previous post.


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## bigshot (Jun 2, 2017)

I think some people are more interested in testing methodologies than sound quality as it relates to music. However, if this is more than just armchair theorizing and you are hearing actual noise and distortion in your system, it's time to get it checked out. Odds are that your system is the problem, not the bit depth.


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## castleofargh

VNandor said:


> I think I've seen that somewhere before. If someone makes a generalizing sweeping statement without noting there are exceptions and the statement is not always true, I think it's fair if someone points out that the generalization is not always applicable. I know this is a lame example but imagine if you analyzed the effects of radiation exposure on humans. Now take your "people have two arms" statement into that context and you will soon conclude that aliens exist. The point I'm trying to make is that there will *always* be that 1% that renders *all* the statements that contain "always, every, all, never ever" and such to be false. Except maybe if we are talking about math. It's possible to make mathematical statements that are always or never true and not depending on the circumstances.
> 
> I'm not surprised people are trying to discuss that 1% in details because the first post seems to cover all the main points of digital audio and doesn't leave much room for discussion other than that.


from http://www.csuchico.edu/phil/sdobra_mat/catclaims.html


> *Realistically...*
> 
> It's one of the messy facts of life that exceptions often arise to complicate most claims that we would like to think of as universal. Complexity theory, a comparatively new sub-discipline in mathematics that has grown out of chaos theory, leads us to expect exceptions to general rules whenever we look closely enough at the details or examine a large enough number of events. This is why a practical approach to categorical logic will include both a knowledge of the deductive techniques that establish what is theoretically true as well as some way to handle exceptions to virtually universal claims without having the whole system break down.
> 
> Aristotle's solution to this problem was to allow generalizations to be true for the most part, thus letting apparent exceptions be until their natures were better understood. The sixteenth century philosopher René Descartes attempted to improve on Aristotle by requiring absolute certainty of the truth of the premises of any argument that would meet the standard of science, but this view has not held up well in our scientific age as we have come to realize that in the world of experience, the possibilities of absolute certainty are few and far between.


so maybe in our amateur audio forum we can allow ourselves to take a step back and agree that the most obvious exceptions are implied. of course if I start saying "*all* humans have 2 arms", and "*nobody* *under no circumstance* can hear a difference between 16 and 24bit" feel free to rub my nose in my false claim. that's fair game IMO.

as for being interested in the 1% exception, of course that's ok. as long as its significance stays clear and does not somehow become described as... let's say the hidden part of an iceberg 
how much audible difference does it make to use 16bit files? the uncertainty stands between almost none under some specific circumstances, and none that we seem to notice the rest of the time. we can't and shouldn't expect improved understanding to suddenly make us all pass abx tests with 80% confidence.

now I understand very well how, when trying to experiment on something, we soon end up with 20 new problems we didn't know existed and 30 new questions we had never thought about. and that's where @Yuri Korzunov iceberg analogy does make sense. but we mustn't mistake that for how audibly significant it is to use 24bit.


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## Yuri Korzunov (Jun 2, 2017)

bigshot said:


> Odds are that your system is the problem, not the bit depth.



Bit depth is mathematical abstraction.
For check it in real world, need implement it in real world.
Thus error of the implementation give error of learning the mathematical abstraction.


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## bigshot (Jun 3, 2017)

Yuri, I have a very simple question for you. Just answer it in one word. Be honest. Do you hear distortion and noise when you play a CD or SACD? It's a yes or no answer. Do you hear error with your ears? That's all I need from you to answer.


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## Yuri Korzunov

bigshot said:


> Yuri, I have a very simple question for you. Just answer it in one word. Be honest. *Do you hear distortion and noise when you play a CD or SACD?* It's a yes or no answer. Do you hear error with your ears? That's all I need from you to answer.



When me seems that I hear something, *I don't know*: it's real or imaginary difference.
Always need check it instrumentally. Bu it is out my possibilities.


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## bigshot

If you can't say yes honestly, I would say that even though you might have curiosity about it, I wouldn't really worry about it. People worry too much about theoretical things they can't hear. That is as much true of golden eared audiophiles as it is sound science folks. I focus on the music and work toward achieving the best sound I can. I don't require it to be perfect to the last decimal point. I just care about pleasing my ears.


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## Yuri Korzunov (Jun 3, 2017)

bigshot said:


> If you can't say yes honestly, I would say that even though you might have curiosity about it, I wouldn't really worry about it. People worry too much about theoretical things they can't hear. That is as much true of golden eared audiophiles as it is sound science folks. I focus on the music and work toward achieving the best sound I can. I don't require it to be perfect to the last decimal point. I just care about pleasing my ears.



I understand your attitude.

But I must exactly control technical aspects of product as manufacturer.
I share the knowledges for somebody, who have additional interest to deeper look to audio issues.


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## gregorio (Jun 3, 2017)

Yuri Korzunov said:


> [1] As we understand, there will not big sound difference anyway.
> [2] Places: anechoic room, other kinds of rooms. ...
> [3] 1. Create audio files with -40 ... - 50 ... -60 ... -80 dB average loudness in 24 and 16 bit. ...
> [4] Thus error of the implementation give error of learning the mathematical abstraction.



1. As we understand it, based on years of experiments, there will be NO sound difference!
2. How many consumers have anechoic rooms? How much commercial music is mixed/mastered in anechoic rooms for playback in anechoic rooms?
3. If the peak level is normalised close to 0dBFS, as has always been standard practice, then it won't make any difference what the average level is, provided the peak level is at a reasonable (not dangerous) SPL. However, there are no commercial recordings which peak near 0dB and have an average level of even -40dB, let alone -80dB. So, in order to satisfy the conditions of your test (and stand a chance of differentiation), there's only two options: A. Manufacture your own test signal, effectively a test signal specifically designed to pass the test or B. Mangle or break any sensible concept of appropriate gain staging.
4. Any competently designed DAC should have "implementation error" well below audibility. Apple iPhones manage it perfectly well with only about $15 or so of DAC chip/audio components. Any DAC which can't achieve this has a serious design flaw or is malfunctioning/broken.

So, these points bring us right back to where we started (again!), to typical audiophile myth/marketing! 24bit will definitely sound better than 16bit, exactly the same as a $1,000 audiophile cable will definitely sound better than a standard (say $15) cable and this could be easily verified with an ABX! We just have to use a standard cable that either; has a serious design flaw, is broken or is inappropriate for the task. In fact, using this technique, we can prove/evidence pretty much anything at all! For example, forget about 24bit vs 16bit, with the correct mangling/breaking of gain staging we could easily demonstrate that 512bit audio is better than 511bit audio. Or, that a Ford Fiesta is quicker than a F1 race car (if the race takes place in say a muddy field or if we use a faulty F1 car).

The topic of this thread has tacit, common sense conditions, it refers to consumers appropriately using competent equipment when listening to commercial audio. It does not cover situations of massively inappropriate use, faulty equipment or signals other than commercial audio content! Just in case I've opened the door for more semantics nonsense, "competent" is defined in point #4 above and "appropriate" is any sensible/reasonable use. And for clarity, lowering the file by say 50dB and then increasing amp gain by 50dB would not be sensible, reasonable or appropriate (in many cases it probably wouldn't even be possible)!

G


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## StanD

gregorio said:


> 4. Any competently designed DAC should have "implementation error" well below audibility. Apple iPhones manage it perfectly well with only about $15 or so of DAC chip/audio components. Any DAC which can't achieve this has a serious design flaw or is malfunctioning/broken.


Oh man, word gets out that you said that and you'll have to change your name and move to the Arctic Circle. Every audiophile will have to talk to their therapists about this.


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## Yuri Korzunov (Jun 3, 2017)

gregorio said:


> 1. As we understand it, based on years of experiments, there will be NO sound difference!
> 2. How many consumers have anechoic rooms? How much commercial music is mixed/mastered in anechoic rooms for playback in anechoic rooms?
> 3. If the peak level is normalised close to 0dBFS, as has always been standard practice, then it won't make any difference what the average level is, provided the peak level is at a reasonable (not dangerous) SPL. However, there are no commercial recordings which peak near 0dB and have an average level of even -40dB, let alone -80dB. So, in order to satisfy the conditions of your test (and stand a chance of differentiation), there's only two options: A. Manufacture your own test signal, effectively a test signal specifically designed to pass the test or B. Mangle or break any sensible concept of appropriate gain staging.
> 4. Any competently designed DAC should have "implementation error" well below audibility. Apple iPhones manage it perfectly well with only about $15 or so of DAC chip/audio components. Any DAC which can't achieve this has a serious design flaw or is malfunctioning/broken.
> ...



For discussion of «years of experiments», currently here we haven’t stuff, besides pair of links, that were discussed above.

Different SPL is accounting of psychoacoustic issues.

If you really ready to correct dialog, pay attention to my phrase above, please:



Yuri Korzunov said:


> Only after careful experiment and analysis of its results, we can suggest, what details is insignificant in common consumer's cases.



However, you can:
1. consider or don't any details, that look for you as ridiculous;
2. accept or don’t any opinions/hypotheses/minds as axioms.


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## bigshot

Yuri Korzunov said:


> I understand your attitude. But I must exactly control technical aspects of product as manufacturer..




What products do you manufacture? Have you invested in independent scientific testing of them?


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## Yuri Korzunov

bigshot said:


> What products do you manufacture?



It's audio file converter software AuI ConverteR 48x44 http://samplerateconverter.com

Now we work under new product, but I don't want disclose it now. 



bigshot said:


> Have you invested in independent scientific testing of them?



No. For small business like mine, it is too big spents.
However you can check independent tests by Archimago (it can be repeatable at home even)
http://archimago.blogspot.ru/2015/04/analysis-dsd-to-pcm-2015-foobar-sacd.html


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## bigshot

I would imagine that for file format conversion software using stock codecs there isn't much need to do testing.


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## Yuri Korzunov (Jun 4, 2017)

bigshot said:


> I would imagine that for file format conversion software using stock codecs there isn't much need to do testing.



There are "native" formats WAV, FLAC, AIFF, DSF, CDA (under Windows) that are opened straightly.
And external formats applied via coding/decoding (to/from "native" formats) by external codecs to reducing of development efforts and flexibility.

Decoded external formats processed by AuI ConverteR and may be coded to same or other external format.

AuI ConverteR provide own (by Audiophile Inventory) PCM/digma-delta modulation/demodulation/processing, safe CD ripping (now both OSX and Win), metadata management and automation.
There concentrated most development efforts.

*This modulation/demodulation/processing, number of errors of CD ripping may be subject of a tests.*
As example, you can see difference between resamplers here http://src.infinitewave.ca
AuI ConverteR tested there.

I deliberately don't limit audio abilities of DEMO, to everybody can free check and compare it by ear and/or measurements.


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## gregorio (Jun 4, 2017)

Yuri Korzunov said:


> A. For discussion of «years of experiments», currently here we haven’t stuff, besides pair of links, that were discussed above.
> B. Different SPL is accounting of psychoacoustic issues.
> However, you can:
> 1. consider or don't any details, that look for you as ridiculous;
> 2. accept or don’t any opinions/hypotheses/minds as axioms.



A. Now that's not true! When you say "we", you really mean "you", you've only seen the two links posted. I for example, have done myself or been involved with numerous ABX tests and read about and/or discussed with other pro engineers numerous other unpublished ABX tests. I also understand the basic logic/science which explains why we can't differentiate.

B. Agreed. Playback the peak level of a CD at say 140dBSPL and there's a good chance that in the really silent bits you'll hear the noise-shaped dither and therefore be able to differentiate between 24bit and 16bit.

1&2. No, it's not ridiculous to utterly mangle the gain staging and it wouldn't be ridiculous to playback your CD at 140dBSPL peak either. In fact, I think you should definitely try, and don't just try it with one track, try it with all of them. When you've ABX'ed your tracks like that for a few days, then we'll talk again about what is and what isn't ridiculous. ... What was that? I SAID, WE'LL TALK AGAIN ABOUT RIDICULOUS ... NO, TALK ... Oh never mind! 

G


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## Yuri Korzunov (Jun 4, 2017)

gregorio said:


> A. Now that's not true! When you say "we", you really mean "you", you've only seen the two links posted. I for example, have done myself or been involved with numerous ABX tests and read about and/or discussed with other pro engineers numerous other unpublished ABX tests. I also understand the basic logic/science which explains why we can't differentiate.



I can’t discuss any tests without detailed protocols.
May be you have these data. But here no the documents.

First I should see, that results:
1. Is repeatable
2. Don't depend on number of experiments, samples, participants, organizers, kinds of samples.

But we can collect data, by available results with accounting of experiment conditions.




gregorio said:


> B. Agreed. Playback the peak level of a CD at say 140dBSPL and there's a good chance that in the really silent bits you'll hear the noise-shaped dither and therefore be able to differentiate between 24bit and 16bit.



I meant ear have different frequency responce at different SPL. It can impact to results of test.




gregorio said:


> 1&2. No, it's not ridiculous to utterly mangle the gain staging and it wouldn't be ridiculous to playback your CD at 140dBSPL peak either. In fact, I think you should definitely try, and don't just try it with one track, try it with all of them. When you've ABX'ed your tracks like that for a few days, then we'll talk again about what is and what isn't ridiculous. ... What was that? I SAID, WE'LL TALK AGAIN ABOUT RIDICULOUS ... NO, TALK ... Oh never mind!



I see the point in detailed researches only, like test, that I described above in general steps.
It is only easiest draft of paper, that should be prepared before test. ABX-test is not home work for fun. It is hard work.
I suppose, methodic should be developed with hardware developers too, who know strong and weak places of equipment.
Probably, with time some person will can to do it.

But it is most important:
We *can’t *make test «*24 bit vs. 16 bit*».
Wa can test «24 bit *implementation* vs. 16 bit *implementation*».


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## Jazmanaut

Yuri Korzunov said:


> But it is most important:
> We *can’t *make test «*24 bit vs. 16 bit*».
> Wa can test «24 bit *implementation* vs. 16 bit *implementation*».



How about good ol Null test?


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## Yuri Korzunov (Jun 4, 2017)

Jazmanaut said:


> How about good ol Null test?



Null test is used to show difference.

Analog comparison give difference.
Pure digital comparison may give, may don't: depend on how to compared samples are created.

Null test can show, that there is difference, but we don't know what it mean for ears.

Null test may be applied in test preparing stage to correct methodic (instrumental learning of apparatus that be used in tests).

Updated:
Null test may be interesting, because we can see difference between equipment modes easy way without big number of estimations.
But this case we should compare each mode with original signal too.
As result we will have peak and average difference in dB. Also we can check diffference value distribution. May be it give some new ideas to further researches.


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## gregorio

Yuri Korzunov said:


> [1] I can’t discuss any tests without detailed protocols.
> [2] I meant ear have different frequency responce at different SPL. It can impact to results of test.



1. Exactly my point, thanks for confirming! It's what YOU can't discuss, what protocols and tests YOU'VE seen, not "we"!
2. Again, exactly my point. Try it at different SPLs and get back to us on what SPL you can hear noise-shaped dither.



Yuri Korzunov said:


> Null test can show, that there is difference, but we don't know what it mean for ears.



Again, typical audiophile nonsense! If for example we have a null test which results in a difference at say -96dB and we play back the original signal at say 90dBSPL, any sensible/logical person would know exactly what that "mean for ears"!

G


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## Yuri Korzunov

gregorio said:


> YOU'VE seen, not "we"!



We (except you).



gregorio said:


> Again, typical audiophile nonsense! If for example we have a null test which results in a difference at say -96dB and we play back the original signal at say 90dBSPL, any sensible/logical person would know exactly what that "mean for ears"!



Read update to my post.


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## gregorio

Yuri Korzunov said:


> [1] We (except you).
> [2] Read update to my post.



1. No, YOU! You do not know what evidence I have witnessed or seen and you do not know what evidence everyone else here has witnessed or seen!
2. Irrelevant.

G


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## Yuri Korzunov

gregorio said:


> 1. No, YOU! You do not know what evidence I have witnessed or seen and you do not know what evidence everyone else here has witnessed or seen!
> 2. Irrelevant.



Clear. You have evidences.


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## gregorio

Yuri Korzunov said:


> Clear. You have evidences.



Yes and some of it has been presented here (and specifically to you)! In addition, there's the basic facts of why we can't hear a difference, presented in the OP. Now if you have a contrary position let's hear your factual explanation and your evidence, otherwise you are just using semantics to continue tolling!

G


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## Yuri Korzunov

gregorio said:


> Yes and some of it has been presented here (and specifically to you)! In addition, there's the basic facts of why we can't hear a difference, presented in the OP. Now if you have a contrary position let's hear your factual explanation and your evidence, otherwise you are just using semantics to continue tolling!



Do you *feel* trolling?

Read Brooko's post above:



Brooko said:


> If Yuri refuses to actually engage properly, and you feel that this is trolling behaviour - *simply don't reply*.


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## gregorio

Yuri Korzunov said:


> Read Brooko's post above:



Maybe you should read Brooko's post and not troll in the first place!

G


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## Yuri Korzunov

gregorio said:


> Maybe you should read Brooko's post and not troll in the first place!



"Trolling" is good argument. I hope you show here more test protocols later.


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## csglinux

https://www.google.com/url?sa=t&rct...6FT8eXiRKdkQma4cA&sig2=Evs2jphB-Q0k5cjD5bOSjg

Thoughts?


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## csglinux

Ok, here are mine...

Forgive me in advance for playing Devil's advocate. It's just the way I roll. I'm a scientist, so I should (I hope) be able and willing to change my stance if evidence starts to stack up in the other direction. In the past, I have argued (probably more vociferously than anybody here) about the scams of hi-res audio, but I've mellowed somewhat recently. I do believe, and have previously mentioned on this thread, that (theoretically) going from 16- to 24-bit might help, although I don't think I have (or even know of) any recordings whose noise floors were low-enough that they'd really benefit from that. The vast majority of material that's currently being sold as "hi-res" music turns out mostly to be up-sampled CD masters. Even if there were a good-quality analog source, it's very unlikely the studio recording's noise floor would warrant 24 bit.

With the bit-depth issue out of the way, I recently heard another potentially compelling argument at a seminar by Bob Stuart (of MQA fame/notoriety), who suggested that recent studies by certain neurologists have shown a high sensitivity of our hearing to timing. The argument goes like this - human hearing has evolved to help us survive, i.e., when a twig snaps behind us, we need to react quickly to not get eaten by a predator. Allegedly, we can discern transients over about 10 microseconds (the above paper quotes even lower values). If you were to try and build a Fourier representation of such an impulse with sine and cosine waves up to 20 kHz, you'd have a lot of ringing and smearing, which (again, allegedly), our ears can detect. One of the arguments for MQA, is that even hi-res audio doesn't really do the best job of resolving sharp transients that aren't tonal in nature. It sounds plausible. I asked Bob after his seminar if he had any papers (other than the QM paper, which I already knew about) that demonstrated with any statistical significance that humans could discern MQA (or even hi-res PCM or DSD audio) from CD. He took my business card and promised to send me some papers, but so far, I've only received the sound of crickets and wind blowing through the tumbleweed. (I will report back if that changes.)

Now, on to the paper above. I believe that we still have a lot to learn about our ears, brains and hearing. It seems there are occasions where we can hear subtleties that are difficult or not (as yet) impossible to measure. However, I am deeply suspicious of this paper by Reiss, for several reasons. Firstly, although Queen Mary is a reasonably-well respected institution, you have to understand what motivates academics. (I know, because I used to be one.) All university research and lecturing staff in the UK are judged and promoted entirely based on research. The worst lecturer in the world can become reader/professor if he/she gets enough journal publications. There's a strong motivation to publish papers that will 1) get published and 2) get attention. Satisfying item 1 generally requires positive results. It's not impossible that an experiment could be reported as a total failure and still get published, but it's less likely to be submitted in the first place and less likely to be accepted, if submitted. That's relevant, because this paper is not a report on a set of carefully controlled SPL-matched ABX tests conducted by the author at Queen Mary, but rather a meta-analysis of existing studies, all of which would likely have suffered the same bias. Reiss is at least honest enough to acknowledge this potential pitfall in his paper.

I've personally known two professors that used to work at QM - one was a personal friend and one of the smartest people I've ever known; the other was a total charlatan, riding the system of publishing (garbage) as often as possible in order to promote himself. I don't know Joshua Reiss, but even if his motives are pure, there's no way of knowing what motivated the individual studies considered in this meta-analysis, or whether the documented tests were really carried out properly via blind SPL-matched tests with properly down-sampled/dithered(?) CD-equivalent files from the identical master and no other potential artifacts that might otherwise cue the listener as to which file was which. In the medical field, it's typical to see research papers on new drugs or devices include a disclaimer, showing the authors' funding sources. (This is supposed to restore credibility to studies that invariably show the sponsor in a favorable light.) We would need a more extensive meta-analysis of this meta-analysis to really know what the motivations were behind the original studies. At the end of his paper, Reiss acknowledges both Bob Stuart and Bob Katz. The former has a big financial incentive with MQA; the latter I know from personal experience to be disingenuous. Have no doubt that the music studios out there are desperate to sell you your entire music catalogue again (and again) in ever-better(?) formats. Authors can pick and choose content for their meta-analysis (as happened here) and when big money has been involved, past practice has been to design studies so that results have minimized the apparent harmful effects of sugar, smoking, human contribution to greenhouse gases, etc. It would be wise to be cynical, or at least careful, here.

Let's give all the researchers involved in Reiss' meta-analysis the benefit of the doubt for a minute and look at the conclusion - listeners are only going to be able to identify the "hi-res" audio file a little over 50% of the time. In other words, they're not doing much better than if they'd flipped a coin. Now, the author's claim is that even though the differences are small (and the odds of correctly identifying the hi-res file only slightly over 50%), it's still statistically significant. Here's the problem with that argument. There's a party trick that you can try at home with some friends that goes like this. You ask all your friends to write down on a piece of paper (after you leave the room) what they imagine a sequence of 20 coin flips would look like, e.g., HTTHHTTHTHT..., etc. Then, you have them actually flip a coin 20 times and write that sequence down too. When you come back into the room, you correctly identify the piece of paper that contains the actual result from the real coin tosses. This trick works because people tend to write down sequences like HTHTHTHTHT..., whereas, in reality, a coin will more often land like HTTTTTHHTHHH...  Consider a larger sample size. To get at least 55 heads from 100 coin tosses isn't long odds (if you want to do the math, it works out at about 20%, or 0.184). In other words, looking for a result that might seem to be statistically significant could be achieved relatively easily - and as the author admits, negative studies tend to be under-represented as their results are likely simply discarded and never reported.

There is, however, one point in the paper that I suspect has more than a grain of truth to it - that a trained listener is more likely to be able to appreciate differences. Perhaps future studies might concentrate only on trained listeners and who knows - maybe those transients are playing a role? I am truly trying to keep an open mind 

Does anybody have any comments/arguments/death-threats, etc.?

P.S. Does anybody have any recommendations on really good-quality hi-res recordings that genuinely make full use of the claimed sampling rate and bit depth?


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## bigshot (Jun 6, 2017)

The problem with the timing argument is that all of the studies used to claim its importance are neurological studies that are completely unrelated to the act of listening to music in the home. The timing error in the average loudspeaker and the way it interacts with the room dwarfs the amount of timing error that MQA is talking about by more than an order of magnitude. We may in certain ways be able to "perceive" timing in a more precise way, but not for listening to music in our living room.

In general, our vision is more precise than our hearing, yet a projector running film at 24 frames per second looks smooth to us. Now imagine 44,100 samples per second- the sample rate of redbook audio. According to Nyquist, two samples can *perfectly* recreate any waveform of any frequency. That means that 44,100 samples per second can *perfectly* reproduce up to 22kHz, which is beyond the range of human hearing.

It is REALLY easy to disprove timing error as an argument for improving sound quality... simply look up the thresholds of perception for timing error in listening tests. There are two kinds of them- ones involving test tones, and ones using music. In general, most forms of timing error with tones are 100 times below the threshold of perception with modest home audio equipment. And under music, it's often more than an order of magnitude below that.

As a for instance... The threshold of perception of group delay, a common form of time error is 1 to 3 ms, being more perceptible in the core upper mid frequencies and less in other frequencies. (The spec I cite is between 500Hz and 8kHz where it is most noticeable.) Assuming the worst- 1 ms... Go look at the minuscule slivers of time MQA is talking about. Then consider the fact that the sampling rate of redbook is by definition accurate enough to perfectly reproduce a waveform 22 times faster than the threshold of perception for group delay. Now realize that listening to music, the threshold is probably considerably higher.

The trick is that you need to understand what the numbers really represent. Exactly how long is a picosecond or nanosecond or millisecond? Figure that out and you'll know better which are the boloney merchants and which aren't. It doesn't take fancy equipment and a degree in physics. All it requires is horse sense and a grasp on the relative proportions of error.


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## Don Hills

bigshot said:


> ... It is REALLY easy to disprove timing error as an argument for improving sound quality... simply look up the thresholds of perception for timing error in listening tests. There are two kinds of them- ones involving test tones, and ones using music. In general, most forms of timing error with tones are 100 times below the threshold of perception with modest home audio equipment. And under music, it's often more than an order of magnitude below that. ...



A good example is LP ("vinyl") reproduction. Play a 1 KHz tone from a test LP. You'll hear subtle variations in pitch (timing) and level, especially compared with the same tone from a digital source (or an analogue signal generator.) Now listen for the same effects whle playing a music LP. They're still there, you're just not as sensitive to them on musical transients.


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## Yuri Korzunov

csglinux said:


> https://www.google.com/url?sa=t&rct...6FT8eXiRKdkQma4cA&sig2=Evs2jphB-Q0k5cjD5bOSjg
> 
> Thoughts?



Csglinux, thank you for the paper. It is very careful document.

Now I rapidly learn the paper and don't found information: there was used single apparatus for all measurements?


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## Yuri Korzunov

csglinux said:


> With the bit-depth issue out of the way, I recently heard another potentially compelling argument at a seminar by Bob Stuart (of MQA fame/notoriety), who suggested that recent studies by certain neurologists have shown a high sensitivity of our hearing to timing.





bigshot said:


> The problem with the timing argument is that all of the studies used to claim its importance are neurological studies that are completely unrelated to the act of listening to music in the home.



As I understand here meant the paper https://phys.org/news/2013-02-human-fourier-uncertainty-principle.html

If I remember correctly, there checked that humans can discriminate short time intervals and made conversion the time to frequency domain.
However, I don't sure that we exactly know how to work brains here exactly. Probably there used other analisys way than furie.

But after I read this article about brain response to ultrasound http://jn.physiology.org/content/83/6/3548

Probably we can try link these experiments.

Other side, I don't seen experimental evidences of ultrasound impact to music perception (by criteria music better/worse with/without ultrasound).

In my opinion need be careful with received information. I'd wait for further researches and keep open mind.


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## Yuri Korzunov

Don Hills said:


> A good example is LP ("vinyl") reproduction. Play a 1 KHz tone from a test LP. You'll hear subtle variations in pitch (timing) and level, especially compared with the same tone from a digital source (or an analogue signal generator.) Now listen for the same effects whle playing a music LP. They're still there, you're just not as sensitive to them on musical transients.



Music is complex stuff. Do you try listen two pieces simultaneously? I observed there some interesting pitch/time efffects.
I suppose, same things with simple sine (easier brain analysis) vs. complex music.


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## gregorio

csglinux said:


> [1] Even if there were a good-quality analog source, it's very unlikely the studio recording's noise floor would warrant 24 bit.
> [2] I recently heard another potentially compelling argument at a seminar by Bob Stuart (of MQA fame/notoriety), who suggested that recent studies by certain neurologists have shown a high sensitivity of our hearing to timing.
> [3] I believe that we still have a lot to learn about our ears, brains and hearing. It seems there are occasions where we can hear subtleties that are difficult or not (as yet) impossible to measure.
> [4] However, I am deeply suspicious of this paper by Reiss, for several reasons.
> ...



1. Analogue sources are out of the equation, the very best commercial analogue recording technology doesn't exceed about 80dB of SNR. Acoustic signals obviously can exceed a DR of 96dB (CD) but in practise that's either undesirable, not attainable, not applicable to music recording or all three!

2. I read that argument by Stuart, in an paper or interview, and it piqued my interest too. It piqued my interest because it goes against conventional recording wisdom and against my personal experience of many years recording, editing and mixing. There's two problems: Firstly, a practical problem. In practice, the timing errors when recording, from multi-mic placement, acoustic reflections, multi-tracking, etc., are all individually relatively massive compared to the figures Stuart is quoting and that's even before we start messing with mixing processes. For example, 10 micro-secs represents a difference in mic placement of about a tenth of an inch (3mm) or conversely, a change in the position of your head of 3mm when listening to a sound source. The second problem is an academic one; my interest piqued, I looked up the neurological study cited by Stuart and in fact, it does not suggest anything of the sort! It was a while ago but if I remember correctly, the study suggested a potential area for further research and suggested discrimination tests should go as low as 10 micro-secs. There was no suggestion that humans actually had any ability to discern 10 micro-secs. To mis-quote/misrepresent the study in this way smacks of intellectual dishonesty on the part of Stuart and IMHO, that's being overly charitable!

3. In some areas/respects, understanding of the perception of hearing is still at a fairly basic stage. Even in the most studied area, where we can measure hearing perception, it's only an approximation, effectively based on an average of thousands of tests spanning many decades. The problem that many audiophiles appear unable to grasp is two-fold: Firstly, much of what they take for granted when listening/hearing does not really exist, it's not a direct property of the sound waves, it's a property of their (albeit shared) perception and secondly, even if there is some perceivable property of sound which is as yet unknown, it's irrelevant to any current or past audio recording technology (including MQA!), we'd need an entirely new technology. All audio recording technology is effectively based on measuring/converting between different types of energy, so if we can't measure it, we can't record it.

4. It has numerous problems but one of the most pernicious IMO is the inclusion on the Theiss study. The Theiss study demonstrated a reasonably high level (about 73% if I remember correctly) of detection of sine waves above 20kHz when played to students at extremely high SPLs, in excess of 110dB. This does not support the title or abstract of the paper which refers to the discrimination of high resolution audio recordings, because of course there are no commercial audio recordings of only individual ultrasonic sine waves and even if there were, consumers should certainly not be playing them back at 110dBSPL! It's debatable whether 52.3% is statistically significant but remove the Theiss study from the meta-analysis and the number falls to a point which no one could sensibly argue there's any statistical significance.

5. Sorry, I don't even know of any commercial recordings which make full use of 16bit, let alone 24bit.

G


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## gregorio

Yuri Korzunov said:


> As I understand here meant the paper https://phys.org/news/2013-02-human-fourier-uncertainty-principle.html



To quote the article: "_The score with the top timing acuity (*3 milliseconds*) was achieved by an electronic musician who works in precision sound editing._" - But Bob Stuart is claiming human timing acuity of 0.01 milliseconds?!!

G


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## castleofargh

if I'm not mistaken this is the paper they refer to http://boson.physics.sc.edu/~kunchur/temporal.pdf 
but using that as a reason to ask for more than 44.1khz doesn't convince me for a few reasons:

 - the delay between 2 samples is not strictly the limit of time information a 16/44 file can contain. 
 - the signal used... I mean come on, how much square waves are we listening to in our favorite albums that will not sound as "realistic" if we get them with a 2µs delay error? am I suppose to tremble in fear about something like that? as for actual music, we know that pretty much any minimum threshold measured, becomes magnitudes less sensitive when the subject is tested with music instead of a dedicated test signal. 
 - when it comes to ringing and limits in frequency response, all the high res guys seem to forget about transducers somehow. they ring, they roll off, but nobody cares if it happens in that specific place?
 - direct practical application, if it matters and we can notice all that in music, why do I keep failing my ABX tests against 96 or 192khz files? what is so wrong in my abx test that it turns 2 allegedly audibly different sounds into something that seems to be audibly the same for me? what are the odds? 
David Blaine Street Magic: wanna see some magic?


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## Yuri Korzunov

gregorio said:


> To quote the article: "_The score with the top timing acuity (*3 milliseconds*) was achieved by an electronic musician who works in precision sound editing._" - But Bob Stuart is claiming human timing acuity of 0.01 milliseconds?!!



I still don't deep analyse the stuff. I not once read links to this. At first sight it is not same that frequency.
I don't know that Bob Stuart mean. Probably in the future discussion we get stuff, that explain it.

I glad, that @csglinux share the link with new stuff. I allow look to issue from other side. I read it in background. One thing that I see, that trained ear skill give visible advantages in the experiment (page 370, Fig.2).




castleofargh said:


> if I'm not mistaken this is the paper they refer to http://boson.physics.sc.edu/~kunchur/temporal.pdf



Thank you for primary source.


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## csglinux

bigshot said:


> The problem with the timing argument is that all of the studies used to claim its importance are neurological studies that are completely unrelated to the act of listening to music in the home.



Well, yes and no. Progressive rock (e.g., Pink Floyd) has an awful lot of ambience and sound effects. It doesn't have to be music. The argument is only that fast, sufficiently-close transients might be used to discriminate two tracks in a listening test.



gregorio said:


> 2. The second problem is an academic one; my interest piqued, I looked up the neurological study cited by Stuart and in fact, it does not suggest anything of the sort! It was a while ago but if I remember correctly, the study suggested a potential area for further research and suggested discrimination tests should go as low as 10 micro-secs. There was no suggestion that humans actually had any ability to discern 10 micro-secs. To mis-quote/misrepresent the study in this way smacks of intellectual dishonesty on the part of Stuart and IMHO, that's being overly charitable!


That's a very, very interesting observation! I guess I need to check this out further. If this is correct, I think their whole argument is dead in the water.
I'm sure the long-term aim for Meridian is simply to have some papers they can cite when asked the obvious searching question. Once published in a half-reputable journal, authors can pretty much quote these things as gospel. After all, who has the time or funding for R&D studies to _disprove_ this paper and/or benefits of hi-res audio or MQA?

BTW, @Yuri Korzunov - the paper was not one study, but a "meta-analysis", where the author selects a number of prior (separate) studies to pool data for common analysis. (That selection process has been sketchy in some significant past meta-analysis studies.)

I am still trying to keep an open mind, but I see little evidence in terms of either science or statistics, and I am not reliably able to distinguish between hi-res files and downsampled equivalents myself (but my ears may not be good enough and/or I may not be a well-trained listener). I'm always surprised at the number of people I meet at audio shows and CanJams who are convinced they can hear the difference with hi-res audio (well over 50% of people I ask). Jude (headfi's owner) recently posted a YouTube video where he described MQA as the highlight of CES. I really like Jude - he's a good guy and doesn't come across as disingenuous, but it's fascinating to me that he and so many others seem to be on board with hi-res. With no scientific evidence for its efficacy, discussions end up like religious or political debates: i.e., there's no evidence for the existence of God and no evidence Trump has ever cared about anything but enriching himself, but convincing believers otherwise is close to impossible, which makes the whole hi-res/MQA gravy train potentially extremely lucrative. Anyway, thanks @gregorio for this thread and for injecting some sanity into an otherwise crazy, messed-up world


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## bigshot (Jun 8, 2017)

Pink Floyd is just about the single most forgiving kind of music when it comes to timing error. It changes phase every 8 bars! If you want to detect timing errors look for really dry recordings of a piano. You'll have a better chance there, although I still think it would be below the threshold of audibility.


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## castleofargh

the paper I linked(sorry for not being clearer about that) was for where the MQA guys got their 5µs value of minimum audible delay. well I think it is because other papers tend to stick to around 10µs or higher.

about training to notice things better, I never doubted it for a second (not even for 5µs). well, the ear is unlikely to just start working better(as it never stops working), the shape won't change to perceive ultrasounds, the lost cells won't grow back thank to Jedi mind tricks. I ran for years and I still can't catch up to a speeding car. but the brain can improve it's own database and the way to process and recognize patterns thanks to more focused experience. so the more examples it has of something, the better it can find it again. and the more used we get to care for something specific, the better we get at noticing it "without thinking". seems fairly intuitive. just like it makes no doubt in my mind that Greg would notice plenty of things in a song and even recognize many of the effects applied because he knows and has used them over and over again in his job. while I'd be "yep, that's a guitar". ^_^ 
now, do I need to practice and try to notice something I don't? my own question was simply "will I notice a clear difference if I pay more for highres?" be it because 16/44 is transparent or because I suck at listening, I got my answer already. 
anyway, for training to work with audio, I imagine we need to practice with things we can notice in the first place. that will probably require to start lower than 16/44 for training purpose, without any certainty to ever be able to notice things at or above 16/44 even after years of training. somehow that doesn't motivate me much.


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## gregorio

csglinux said:


> That's a very, very interesting observation! I guess I need to check this out further. If this is correct, I think their whole argument is dead in the water. I'm sure the long-term aim for Meridian is simply to have some papers they can cite when asked the obvious searching question. Once published in a half-reputable journal, authors can pretty much quote these things as gospel. After all, who has the time or funding for R&D studies to _disprove_ this paper and/or benefits of hi-res audio or MQA?



The paper castleofargh linked to was not the one I was referring to and unfortunately, I can't trace back my steps and link to the right one. However, there are a number of examples of this type of ploy by Stuart. We've already covered in this thread Stuart's earlier paper "The Audibility of Typical Digital Audio Filters in a High-Fidelity Playback System" (which he cites on numerous occasions in support of MQA) and the fact that the ABX tests they ran didn't actually include any filters from Hi-Fi systems but instead simulated a highly atypical (very narrow band) filter and in addition, they applied a bit reduction (24->16) with no dither and rectangular dither, despite the fact that Stuart himself stated in an earlier paper that triangular dither should be the required minimum and that Meridian's own DACs have employed noise-shaped dither (as indeed does MQA) since the 1990's! It's nonsense but who is going to notice it's nonsense? Pro reviewers are just going to look at the title and conclusion, never realise that the actual tests/evidence presented in the paper are in fact unrelated to it's title and conclusion and just quote it as scientific proof. You've got to hand it to him, it's sophisticated marketing. The most sophisticated audiophile marketing typically has; this is what pro engineers believe, here's a few engineer/producer celebrity endorsements to back up those lies and here's a bunch of testimonials from respected audiophile publications (with whom we spend our advertising budget). Stuart does all this and goes a step further, with scientific, peer reviewed papers published in respected journals which (seemingly) provide scientific proof of his marketing claims. The only people likely to recognise the scam and speak openly about it are the few forums like this one and the famous Carver amp expose in the 1980's demonstrated that the mainstream audiophile world can simply ignore even the obvious, demonstrated facts and carry on regardless. Here we are 30 odd years later and we still have ridiculously priced audiophile amps, only now we've got ridiculously priced audiophile DACs and audio formats to add to the ever growing pile of snake-oil!

G


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## ShenqLim

In short, bit in PCM format records the degree of the sound amplitude while the khz in PCM format shows how many times of sampling in a second.


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## castleofargh

yup the basic idea is that PCM bits contain the amplitude of the signal at a given moment, and the bit depth will essentially determine the amplitude of digital noise floor. while the sample rate tells us how high in frequency we can record. 
that doesn't tell much about what a human being can perceive though ^_^;


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## pinnahertz

castleofargh said:


> yup the basic idea is that PCM bits contain the amplitude of the signal at a given moment, and the bit depth will essentially determine the amplitude of digital noise floor. while the sample rate tells us how high in frequency we can record.
> that doesn't tell much about what a human being can perceive though ^_^;


That would be theoretical, but in the real world the digital noise floor is determined by the noise in the ADC, and everything ahead of it. Most 24bit ADCs present 18-20bit noise floors.


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## castleofargh

sure, and then the rest of the analog path usually tells the digital noise who's the noise boss, and the ambient noise chilling with us will be like "that's cute". 
I was keeping things basic and local.


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## StanD

castleofargh said:


> sure, and then the rest of the analog path usually tells the digital noise who's the noise boss, and the ambient noise chilling with us will be like "that's cute".
> I was keeping things basic and local.


That's nothing compared to the ambient noise level of forum chatter about noise and DR. There are still people holding on to the value of 32 bit DACs, they must be budding seismologists hoping for a dream come true.


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## Slaphead (Jun 11, 2017)

StanD said:


> That's nothing compared to the ambient noise level of forum chatter about noise and DR. *There are still people holding on to the value of 32 bit DACs,* they must be budding seismologists hoping for a dream come true.



Well I totally understand the need for 24 or 32 bits, but only within a production environment.

Everything, and I mean everything is EQ'd and mixed on a computer these days, and that means that the sound (or signal) remains entirely within the digital domain. Having those extra bits for dynamic range allows for far more accurate EQ, summing, and effects. If you were only working with 16/44 then quantisation noise would gradually creep in with more and more operations on the source. With 32/384 this is simply not a problem - even 24/192 would suffice for the vast majority of productions.

When the project is finalised and bounced down to 16/44 it will sound identical to that which the producers have worked with. Nothing more is needed for the delivery to the end user.

People seem to jump onto the fact that studios use this technology and therefore believe that they must have it to experience the sound as it was originally. However they fail to grasp that that the reason that studios use this technology is to allow editing and adjustments with minimal impact to the sound quality. For simply playback it's totallly not necessary to be sitting on a 32/384 master, or even a 24/192 master - 16/44 is more than is required.

So with that in mind I agree with you, but 32 bit DACs are still needed.


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## bigshot

One of the best sounding albums I've ever heard, Donald Fagan's The Nightfly was recorded 16/44.1, and another of the best sounding albums I've heard is Fiedler's Gaetie Parisienne which was one of the first stereo recordings in 1952. Sound quality depends a lot more on engineering than it does on numbers. Even in a production environment we're well into the range of overkill.


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## EtherealPoetry

I have a few 24/92 albums, and while they sound great, I can't honestly say they sound better by any significant degree to 16/44.1 versions. 

The latter definitely sounds superior to even higher bitrate MP3s, but I can't tell the difference between 24-bit and 16-bit, much less 32-bit.


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## bigshot

With MP3 it depends on the codec. LAME 320 would probably sound exactly the same as the super fancy stuff. AAC 256 VBR would too.


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## pinnahertz

Slaphead said:


> Well I totally understand the need for 24 or 32 bits, but only within a production environment.
> 
> Everything, and I mean everything is EQ'd and mixed on a computer these days, and that means that the sound (or signal) remains entirely within the digital domain. Having those extra bits for dynamic range allows for far more accurate EQ, summing, and effects. If you were only working with 16/44 then quantisation noise would gradually creep in with more and more operations on the source.


All DAWs worth professional consideration are working internally with 32 or 64 bit floating point already, and have been for years.


Slaphead said:


> With 32/384 this is simply not a problem - even 24/192 would suffice for the vast majority of productions.


As I said above...but you can leave the sampling frequency out of the DR discussion.  It has nothing to do with it, until you are forced to run down-sampling to get to a releasable 44.1K or 48K, then that high sampling rate causes you to take a noise hit.


Slaphead said:


> When the project is finalised and bounced down to 16/44 it will sound identical to that which the producers have worked with. Nothing more is needed for the delivery to the end user.


There is a theoretical degradation in the down-sampling to a lower frequency.  If you want identical, work in identical.


Slaphead said:


> People seem to jump onto the fact that studios use this technology and therefore believe that they must have it to experience the sound as it was originally. However they fail to grasp that that the reason that studios use this technology is to allow editing and adjustments with minimal impact to the sound quality. For simply playback it's totallly not necessary to be sitting on a 32/384 master, or even a 24/192 master - 16/44 is more than is required.


Agreed with the basic concept, if not the specifics.  Nobody does anything at 32/384, for example.


Slaphead said:


> So with that in mind I agree with you, but 32 bit DACs are still needed.


What? So 16/44 is more than required, but we need 32 bit DACs?  What, exactly, would a 32 bit DAC....um...DAC?  Half the bits would do nothing but dither around in the noise?  Why do we need that...again?


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## pinnahertz

bigshot said:


> With MP3 it depends on the codec. LAME 320 would probably sound exactly the same as the super fancy stuff. AAC 256 VBR would too.


For me, AAC 256 CBR beats LAME 320.  But most people think I'm weird. Most of the library is in AAC 320 or uncompressed, until it gets to the evil and fearsome iDevice.


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## sonitus mirus

pinnahertz said:


> For me, AAC 256 CBR beats LAME 320.  But most people think I'm weird. Most of the library is in AAC 320 or uncompressed, until it gets to the evil and fearsome iDevice.



That makes sense. It is a newer technology and I seem to recall JJ Johnston of AT&T/Bell Labs mention some inherent limitation with MP3 that was largely resolved with AAC.  I don't recall the exact details, but it has something to do with the psychoacoustic model's FFT filterbank data block having two different lengths available, and being significantly longer.  I can't hear any difference between AAC 256 and MP3 VBR -0, but AAC is most certainly the technologically superior format.


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## pinnahertz

It's been demonstrated that people can be "trained" to hear things not easily picked up normally.  In my own unfortunate case I've trained to hear dynamic processing artifacts of compression and limiting, and also the artifacts of bit-rate reduction, lossy codecs, particularly mp3.  So now I'm sensitive to low-rate mp3 artifact, and hear it a lot, even on radio where they've used mp3 codecs for electronic distribution of commercials.  Once you know what it sounds like, you can never ignore it.  So far, I haven't had that issue with AAC, but also don't try it at low bit rates either.  Something I don't really want to train for.


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## bigshot

pinnahertz said:


> For me, AAC 256 CBR beats LAME 320.  But most people think I'm weird. Most of the library is in AAC 320 or uncompressed, until it gets to the evil and fearsome iDevice.



I did a lot of tests before I arrived at a setting for my music server. I found MP3 LAME 320 CBR was audibly transparent, but I could get the same thing with AAC at 256. My library is all AAC 256 VBR. With AAC it's best to always use VBR because it never degrades the sound, but can only improve it. One interesting difference is that VBR at 320 in AAC can actually go beyond 320. In MP3, it's capped at 320, so there's no point using VBR with MP3 320.


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## Cold Train

bigshot said:


> With MP3 it depends on the codec. LAME 320 would probably sound exactly the same as the super fancy stuff. AAC 256 VBR would too.



It actually depends on the DAC and how it has been implemented to decode M4a or Mp3. 

As long as you have idevices, M4a will always sound better than MP3, but MP3 sounds better than M4a on non-Apple products, I did some proper ABX with the same master from a WAV 24/192, converted in multiple formats including 16/44 Flac, Mp3 and M4a and there's some kind of veil on the M4a when playing on my non-idevices.


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## bigshot

On Apple products high bitrates of both MP3 and AAC sound exactly the same as lossless. I guess your non-Apple player isn't as good at AAC.


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## sonitus mirus

Cold Train said:


> It actually depends on the DAC and how it has been implemented to decode M4a or Mp3.
> 
> As long as you have idevices, M4a will always sound better than MP3, but MP3 sounds better than M4a on non-Apple products, I did some proper ABX with the same master from a WAV 24/192, converted in multiple formats including 16/44 Flac, Mp3 and M4a and there's some kind of veil on the M4a when playing on my non-idevices.



A DAC isn't responsible for decoding MP3 or any other audio file format.


----------



## Cold Train (Jun 12, 2017)

bigshot said:


> On Apple products high bitrates of both MP3 and AAC sound exactly the same as lossless. I guess your non-Apple player isn't as good at AAC.



Regardless of the player, the tuning varies from one to another.  For e.g my Colorfly C3 sounds marvelously good with MP3, but my $2.4k Astell & Kern AK240 failed to match this synergy on lossy files. All depends on how things have been implemented, Somehow, even on my android smartphones I feel that MP3 is more ''crispier'' than M4a, it may be a matter of preference afterall, but I heard the same story from another music junkies


----------



## Cold Train

sonitus mirus said:


> A DAC isn't responsible for decoding MP3 or any other audio file format.


What is responsible then?  Please tell me


----------



## bigshot (Jun 12, 2017)

Cold Train said:


> Regardless of the player, the tuning varies from one to another.



Well, I only know Apple, but going back to the original TV set iMacs, AAC 256 and MP3 LAME 320 both sound exactly like lossless. It doesn't matter if it's a computer, iPod or phone... transparent is transparent. And that's the way it should be. If the funky format is one you never use, I suppose it's fine, but if I bought a player that played one format worse than another, I'd probably return it. But that's just me. I test every piece of equipment I buy carefully to make sure it works properly so I can return it in case I get a bum one.


----------



## StanD

Slaphead said:


> Well I totally understand the need for 24 or 32 bits, but only within a production environment.
> 
> Everything, and I mean everything is EQ'd and mixed on a computer these days, and that means that the sound (or signal) remains entirely within the digital domain. Having those extra bits for dynamic range allows for far more accurate EQ, summing, and effects. If you were only working with 16/44 then quantisation noise would gradually creep in with more and more operations on the source. With 32/384 this is simply not a problem - even 24/192 would suffice for the vast majority of productions.
> 
> ...


Most of the technically oriented posters to this thread have said everything you just stated, except for the bit about 32 bit DACs. After doing all of the signal processing, etc. at 32 bits, one does not need a 32 bit DAC to produce listenable output, especially since a deliverable format of 16/44 does not require 32 bit DACs. Next most important item, what DAC actually delivers a DR or SNR that begins to approach 1 LSB of 32 bits? To get a feel for the scale, take 1 Volt and divide that by 2^32 and you get  0.00000000023 which is less than a nano-volt. If you want to fiddle with the MSB as a sign bit, it's still less than a nano-volt. Good luck finding a DAC that can deliver that kind of a signal that is not swamped by noise. What is the DR that you expect a 32 bit DAC to be able to deliver to be called 32 bit?


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## sonitus mirus (Jun 13, 2017)

Cold Train said:


> What is responsible then?  Please tell me


The music player.  The application that ultimately sends the PCM (in most cases) signal to your DAC to convert it to analog.  The DAC does not take raw AAC, MP3, OGG, Opus, ALAC, or FLAC without first being decoded to a format the chip can read.


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## castleofargh

Cold Train said:


> What is responsible then?  Please tell me


https://www.head-fi.org/f/threads/tidal-lossless-streaming.733233/page-236#post-13539596


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## Cold Train (Jun 14, 2017)

bigshot said:


> Well, I only know Apple, but going back to the original TV set iMacs, AAC 256 and MP3 LAME 320 both sound exactly like lossless. It doesn't matter if it's a computer, iPod or phone... *transparent is transparent.* And that's the way it should be. If the funky format is one you never use, I suppose it's fine, but if I bought a player that played one format worse than another, I'd probably return it. But that's just me. I test every piece of equipment I buy carefully to make sure it works properly so I can return it in case I get a bum one.





Agree that both AAC 256 and MP3 320 sounds like lossless on some devices, agree that originally they may be transparent but they are not transparent through any setup, and surely not from the output of an Apple device


----------



## sonitus mirus

Cold Train said:


> Agree that both AAC 256 and MP3 320 sounds like lossless on some devices, agree that originally they may be transparent but they are not transparent through any setup, and surely not from the output of an Apple device



You stated earlier that a "proper ABX" was performed and that you converted a 24/192 WAV into multiple formats.  Maybe start there.  How did you convert the file.  What tools and settings were used to make the conversion?  How did you perform the ABX to verify your claims?  How did you volume match the different versions?  Seems like a few steps along the way could be responsible for any noticeable differences that might be heard.  Without providing a bit more information, there is no way for any of us to know.


----------



## bigshot

Cold Train said:


> Agree that both AAC 256 and MP3 320 sounds like lossless on some devices, agree that originally they may be transparent but they are not transparent through any setup, and surely not from the output of an Apple device



I'm not sure what you're saying here. "Sounding like lossless" is the same as "audibly transparent" and the audio output of Apple devices is generally as good or better than most dedicated audio components.


----------



## castleofargh

I know nothing of Apple gears so some of what I say might not apply:
some devices mess up a codec, in my experience it has too often been typically a problem with .OGG, but it's a non null possibility. with mp3 it happened to me only twice on DAPs with the firmware still in progress. I don't know if Apple is the type to sell gears while still beta testing the firmware.
also one encoder could do stuff like lower the loudness a little, and that in turn could be audible because it's clearly quieter, or result in removing the audible intersample clipping you get on the other file. that I've experienced several times while trying random encoders or crappy files from a buddy. the song needs to have first been mastered without a brain, but loudness war is still going strong.  
and that's about it as to why I imagine one would clearly identify aac 256 or 320mp3. 

oh maybe if they are at different sample rates and the device sucks at 44.1khz with default rate at 48khz? but I wouldn't consider that a proper test of formats anyway. 


can I complain about how it's not on topic now that I've participated to the off topic? you know, "do what I say not what I do" kind of modo.


----------



## gregorio

Cold Train said:


> Agree that both AAC 256 and MP3 320 sounds like lossless on some devices, agree that originally they may be transparent but they are not transparent through any setup, and surely not from the output of an Apple device



Actually, Apple devices are generally very transparent, with a virtually flat FR throughout the audible range. This raises two observations I've noticed over the years:

1. Audiophiles often like to cite the rule/law of diminishing returns. What's interesting is that not only does this rule not apply in many areas of audiophilia but often it's even actually inverted! IE. Increasing cost often gives either effectively the same performance or in some cases worse performance. For example, audiophile cables typically provide no performance gains, at any price, USB "de-crapifiers" typically do the opposite and audiophile DACs perform equally or less transparently than bog-standard Apple devices.

2. Audiophiles typically reference everything to their personal preferences and simply misappropriate terms and invent new ones, to get around this fact. If an audiophile prefers some bit of kit they tend to describe it as: "accurate", "neutral" or "transparent", regardless of the fact that it may actually be less accurate, neutral or transparent than a far cheaper/mass produced version. In my experience, present most audiophiles with a very accurate/neutral/transparent listening experience and they'll be shocked/surprised and often critical. A typical amalgam of comment would be "Very nice but too clinical, I prefer my system because it's much more musical." - The misappropriated term "musical" apparently means coloured/inaccurate and "clinical" apparently means accurate. I don't see how a system can be too accurate though, so I suspect the term "clinical" has been invented by audiophiles to counter/defend a personal preference for a certain level of inaccuracy. I'm not saying audiophiles cannot have a personal preference, just that they should call it what it is and not lie about it! I've got no problem for example, with someone preferring vinyl to digital, I do object to them asserting that it's more accurate or transparent though.

G


----------



## bigshot

If Shakespeare was alive today, he wouldn't be writing sonnets. He'd be writing audiophile equipment reviews!

"The quality of tweeters are not strained." "To PRAT or not to PRAT, that is the question."


----------



## StanD

bigshot said:


> If Shakespeare was alive today, he wouldn't be writing sonnets. He'd be writing audiophile equipment reviews!
> 
> "The quality of tweeters are not strained." "To PRAT or not to PRAT, that is the question."


Whether 'tis nobler in the mind to make up Schitt.
The slings and arrows of outrageous forum chatter.
Or to take arms against a sea of myths
And by opposing end them.


----------



## Cold Train

StanD said:


> Whether 'tis nobler in the mind to make up Schitt.
> The slings and arrows of outrageous forum chatter.
> Or to take arms against a sea of myths
> And by opposing end them.



_A bove ante_, ab asino retro, a stulto undique caveto


----------



## Cold Train

bigshot said:


> I'm not sure what you're saying here. "Sounding like lossless" is the same as "audibly transparent" and the audio output of Apple devices is generally as good or better than most dedicated audio components.



Lossless and transparent, the same? I can read that you like Apple so much....


----------



## bigshot

If lossless is the unaltered source and a compressed version is audibly transparent (adding or subtracting nothing you can hear) then they sound exactly the same. Are you talking about theoretical sound or sound you can actually hear?


----------



## Cold Train (Jun 16, 2017)

bigshot said:


> If lossless is the unaltered source and a compressed version is audibly transparent (adding or subtracting nothing you can hear) then they sound exactly the same. Are you talking about theoretical sound or sound you can actually hear?



Only sound that a human can hear, theoretical graphs and numbers are for engineers and are somewhat useless in real world.

We shall only focus on what we can hear !


----------



## StanD

Cold Train said:


> _A bove ante_, ab asino retro, a stulto undique caveto


Donkeys are smart enough not to fall for hirez marketing hype.


----------



## Cold Train

StanD said:


> Donkeys are smart enough not to fall for hirez marketing hype.



MP3 is still the king of my collection, so I don't feel that I'm directly targeted by your statement


----------



## StanD

Cold Train said:


> MP3 is still the king of my collection, so I don't feel that I'm directly targeted by your statement


I stream 320k (more than enough) MP3 from my Google All Music subscription and m very satisfied by the result. What's wrong with donkeys?


----------



## Cold Train (Jun 16, 2017)

StanD said:


> I stream 320k (more than enough) MP3 from my Google All Music subscription and m very satisfied by the result. What's wrong with donkeys?



I personally believe that everything above 16/44 is a scam !

_'' Even with a $4000 Focal Utopia on it's head, a donkey remains a donkey,''_


----------



## StanD

Cold Train said:


> I personally believe that everything above 16/44 is a scam !
> 
> _'' Even with a $4000 Focal Utopia on it's head, a donkey remains a donkey,''_


You shouldn't pick on donkeys, they're as smart as any audiophile. They're stubborn because they are smart, ask one and they'll tell you, "You want tricks, get a dog."


----------



## Cold Train

StanD said:


> You shouldn't pick on donkeys, they're as smart as any audiophile. They're stubborn because they are smart, ask one and they'll tell you, "You want tricks, get a dog."



_a cane muto et aqua silente cave tibi_


----------



## StanD

Cold Train said:


> _a cane muto et aqua silente cave tibi_


My dog drinks bottled water. Hold it, I don't have a dog, but I do have headphones.


----------



## Cold Train

StanD said:


> My dog drinks bottled water. Hold it, I don't have a dog, but I do have headphones.


----------



## Cutestudio

gregorio said:


> Dither: Essentially during the conversion process a very small amount of white noise is added to the signal, this has the effect of completely randomising the quantisation errors. Randomisation in digital audio, once converted back to analogue is heard as pure white (un-correlated) noise. The result is that we have an absolutely perfect measurement of the waveform (2*) plus some noise. In other words, by dithering, *all* the measurement errors have been converted to noise. (3*).
> 
> Hopefully you're still with me, because we can now go on to precisely what happens with bit depth. Going back to the above, when we add a 'bit' of data we double the number of values available and therefore halve the number of quantisation errors. If we halve the number of quantisation errors, the result (after dithering) is a perfect waveform with halve the amount of noise. To phrase this using audio terminology, each extra bit of data moves the noise floor down by 6dB (half). We can turn this around and say that each bit of data provides 6dB of dynamic range (*4). Therefore 16bit x 6db = 96dB. This 96dB figure defines the dynamic range of CD. (24bit x 6dB = 144dB).
> 
> So, 24bit does add more 'resolution' compared to 16bit but this added resolution doesn't mean higher quality, it just means we can encode a larger dynamic range. This is the misunderstanding made by many. There are no extra magical properties, nothing which the science does not understand or cannot measure. The only difference between 16bit and 24bit is 48dB of dynamic range (8bits x 6dB = 48dB) and *nothing else*. This is not a question for interpretation or opinion, it is the provable, undisputed logical mathematics which underpins the very existence of digital audio.



On page 272 I can tell this is an interesting subject, I haven't had time to read all of the pages yet so I'm just responding to the original assertion with my thoughts here.

Whenever bit depth is discussed along comes _dither_ as the knight in shining armour, and the noise is always a 'tiny bit'. 
For 2 bit audio the noise would in fact be quite significant, but lets leave that aside. In order to include classical music - some of which has a fair bit of music at 48dB down we are not talking about 16bit audio but 8bit audio, so (single bit) dither in that case is 1/255 or 0.4% noise, whereas in a 24bit system 48dB down (24 - 8 = 16) it would be 0.0015%.

But it's not so much even the noise, it's the problem of transient peaks, little transient peaks. Dither is a process for continuous waves because the error gets averaged out over time, this doesn't help transient peaks, which technically dither will damage more than no dither, because you are potentially moving all of the points around by a level (1). That means for a quiet 16bit classical section (which 48db down is 8bit) that 0.4% noise becomes a 0.4 - 0.8% level error for each of the several points of the peak, which I've always considered to be _distortion_.

This tells us that 24bit will sound better for quiet, transient sounds like a soft cymbal brush, ambient sounds, subtle cues, but for 1kHz test tones there may not be any point to 24bit.
Maybe others are different, but I don't listen to 1kHz test tones, I prefer dynamic music with both loud (fine for 16bit) and quiet (bad for 16bit) parts.


----------



## gregorio

Cutestudio said:


> [1] Whenever bit depth is discussed along comes _dither_ as the knight in shining armour, and the noise is always a 'tiny bit'.
> [2] For 2 bit audio the noise would in fact be quite significant, but lets leave that aside.
> [3] In order to include classical music - some of which has a fair bit of music at 48dB down we are not talking about 16bit audio but 8bit audio, so (single bit) dither in that case is 1/255 or 0.4% noise, whereas in a 24bit system 48dB down (24 - 8 = 16) it would be 0.0015%.
> [4] Dither is a process for continuous waves because the error gets averaged out over time, [4a] this doesn't help transient peaks ...
> ...



1. Dither is not the knight in shining armour, it is intrinsic to digital audio implementation. It's like saying wheels are the knight in shining armour for a car!
2. And for 1 bit audio the noise would be twice as much as for 2 bit. Ever heard a SACD, did you notice that ridiculous amount of noise?
3. Not quite sure how you're working that out or what you're trying to say but yes, I agree, 24bit audio is generally easily distinguishable from 8bit (with music recordings and no noise-shaped dither), due to the amount of audible noise. 
4. I think you've misunderstood something at a fundamental level. Dither does NOT "average the error out over time", it effectively converts the error into uncorrelated noise (white noise). The fewer the bits, the more error and therefore more dither is required to eliminate that error and you end up with more white noise but regardless of the amount of dither (and resultant white noise), what we're left with after dithering is ZERO error. So for 16bit, the error is NOT 0.8% it's 0.0%, it's always 0.0% after dithering at any bit depth, which is why dithering is ALWAYS applied and why it's intrinsic to digital audio!
4a. Correct, it doesn't help the transient peaks. However, it doesn't hurt/harm the peaks either. In fact, it doesn't affect the peaks (or any other property/characteristic of the audio) in any way whatsoever, until we get down to the level of the white noise and even then the audio still has zero error, it's just difficult to recognise that fact because it's buried in that white noise. The only issue left then, is how much white noise do we have? For 16bit, that white noise would be at about -92dB, which is more than 100 times below the -48dB figure you're quoting!
5. No, the facts tell us the exact opposite! It tells us that 24bit CANNOT sound better for quiet transients or any other sounds, because whatever bit depth we use the result will always be zero error after dithering! Again, the only issue is how much white noise do we have in addition to our zero error signal but when the level of that white noise is well below the noise floor of the listening environment, the noise floor of the recording chain, the noise floor of the reproduction chain or the noise floor of the recording environment itself, then we can forget about it because it's inaudible and irrelevant. The level of the white noise with 16bit fulfils not just one but all of these conditions!

G


----------



## castleofargh

Cutestudio said:


> On page 272 I can tell this is an interesting subject, I haven't had time to read all of the pages yet so I'm just responding to the original assertion with my thoughts here.
> 
> Whenever bit depth is discussed along comes _dither_ as the knight in shining armour, and the noise is always a 'tiny bit'.
> For 2 bit audio the noise would in fact be quite significant, but lets leave that aside. In order to include classical music - some of which has a fair bit of music at 48dB down we are not talking about 16bit audio but 8bit audio, so (single bit) dither in that case is 1/255 or 0.4% noise, whereas in a 24bit system 48dB down (24 - 8 = 16) it would be 0.0015%.
> ...


the transient things is a mystery to me. I can only agree that 16bit has higher quantization noise than 24bit. we all agree on that.  
dither in it's most basic form is there to decorrelate that quantization noise, but noise shaping does in practice the opposite of what you're saying. it provides in principle at least, the ability to perceive signals even below -96db. I know it's not intuitive, dither applied to a picture would probably make more sense than trying to provide an example at such a low audio level, or you could always fool around with a 8bit files and different dither options to try and experience the principle yourself. 

now about taking a quiet sound at -48db(assuming full scale would be at normal to loud listening level) within the track and looking at it as if it was somehow the only sound and we have perfect hearing at that level, that doesn't work because in real life situations signal amplitude and quantization noise aren't the only variables involved. for example, what about all the other noises? noises in the studio, noises from the microphones, both most likely increased on some of the tracks to match the gain with the other instruments while mixed. what about your playback system? your listening room? they're all likely to add noises/distortions louder than the dither noise. so you can cry about your low level sounds, because yes they have noises and other stuff creepping just below or even above them. and yes as waves have a cumulative nature, all of it ends up in the final sound wave. but 16bit dither will in practice still be low enough to register most of those noises and play them back to you on a perfect playback system in an anechoic chamber where you might be able to tell how they sound. well you will if they aren't masked psycho acoustically by the -48db music. ^_^


----------



## Cutestudio

castleofargh said:


> the transient things is a mystery to me. I can only agree that 16bit has higher quantization noise than 24bit. we all agree on that.
> dither in it's most basic form is there to decorrelate that quantization noise, but noise shaping does in practice the opposite of what you're saying. it provides in principle at least, the ability to perceive signals even below -96db.
> I know it's not intuitive, dither applied to a picture would probably make more sense than trying to provide an example at such a low audio level, or you could always fool around with a 8bit files and different dither options to try and experience the principle yourself.



Jiggling a signal that’s not causing a bit transition pushes it over the transition point for an amount statistically proportional to its actual amplitude level.

My argument is that for short transient sounds in music you haven't got enough samples for those statistics to work. 
The correct fix for quantisation error in 16bit audio is therefore to switch to 24bit audio, which today is simple and almost without cost.



castleofargh said:


> now about taking a quiet sound at -48db(assuming full scale would be at normal to loud listening level) within the track and looking at it as if it was somehow the only sound and we have perfect hearing at that level, that doesn't work because in real life situations signal amplitude and quantization noise aren't the only variables involved. for example, what about all the other noises?



Classical music has a wide dynamic range with quiet stuff being played at reasonable volume. With a decent amp and speakers: or decent headphones, this is not an unreasonable scenario in the HiFi world.
You appear to be saying 24 bits is more accurate than we need...
...What's the drawback of using a format that's better than we need?


----------



## Cutestudio

gregorio said:


> Dither does NOT "average the error out over time", it effectively converts the error into uncorrelated noise (white noise). The fewer the bits, the more error and therefore more dither is required to eliminate that error and you end up with more white noise but regardless of the amount of dither (and resultant white noise), what we're left with after dithering is ZERO error. So for 16bit, the error is NOT 0.8% it's 0.0%, it's always 0.0% after dithering at any bit depth, which is why dithering is ALWAYS applied and why it's intrinsic to digital audio!


I understand how this applies to continuous tones, but not how it applies to transients. 
That "uncorrelated noise" is the error being spread (and resolved) over time - is it not?
If so, don't we need the error fixing at the time of the transient, rather then over the next few hundred samples?


----------



## bigshot (Jun 18, 2017)

What kind of transient are you going to find in music that doesn't cover hundreds of samples? I'm just guessing, but I would bet a very fast snare drum hit probably lasts no less than 20 ms from attack to decay. Even if you just worry about the attack, that's still 2 or 3 ms.


----------



## pinnahertz

bigshot said:


> What kind of transient are you going to find in music that doesn't cover hundreds of samples? I'm just guessing, but I would bet a very fast snare drum hit probably lasts no less than 20 ms from attack to decay. Even if you just worry about the attack, that's still 2 or 3 ms.


You're of by at least an order of magnitude.  What frequency would have a 20ms period?  That's  50Hz.  The attack portion of a snare hit is much faster, the attack of a cymbal, bell or glockenspiel is up higher than 20kHz (the decays are of course much longer).  The fastest single transient that can be recorded accurately would have a period of 1/2 the sampling frequency, for 44.1 that's about 45uS, and that would be two samples.  If a transient with that kind of attack decays a while, it becomes many samples quickly.


----------



## bigshot (Jun 18, 2017)

Maybe I'm misunderstanding, math isn't my best subject, but I'm not talking about frequency, I'm talking about how long the attack and decay of a fast snare drum hit would be in music- real world transients, not theoretical ones. I've looked at waveforms of drums and none of the hits were anywhere near the microsecond range.On the fastest hits, it usually took 2 or 3 milliseconds for the attack to register, then a decay that lasted at least ten or twenty times that, often much longer, Go look at the waveform of a drum hit. It isn't like a brick wall hitting all at once. It's a mountain that rises to a peak and then trails off. It seems to me that worrying about anti-aliasing affecting real world musical transients would be like worrying about a rock or two being out of place on Mount Everest.


----------



## pinnahertz

bigshot said:


> Maybe I'm misunderstanding, math isn't my best subject, but I'm not talking about frequency, I'm talking about how long the attack and decay of a fast snare drum hit would be in music- real world transients, not theoretical ones.


I'm talking real-world music also.  Time and Frequency are just a different window on the same information, different "domains" if you like.  The two are inseparable, as they are the same data.  


bigshot said:


> I've looked at waveforms of drums and none of the hits were anywhere near the microsecond range.On the fastest hits, it usually took 2 or 3 milliseconds for the attack to register, then a decay that lasted at least ten or twenty times that, often much longer, Go look at the waveform of a drum hit. It isn't like a brick wall hitting all at once. It's a mountain that rises to a peak and then trails off.


What you refer to here is actually the "envelope" of the signal, and your numbers are fine for that. The envelope is the overall amplitude vs time graph of the hit.  If you zoom in and look at that waveform a bit more carefully and note the specific rise-times involved at various points, you'll see there are components of that sound that rise much faster than 2-3ms, fast enough to produce frequency components all the way up to 20kHz and beyond.  I did an FFT of a single snare hit sample here and saw dense spectrum up to Nyquist.  You can't get that if your fastest rise time is 3ms (that inverts to 333Hz).  The math is simple, reciprocal of time is frequency, and vise-versa.   


bigshot said:


> It seems to me that worrying about anti-aliasing affecting real world musical transients would be like worrying about a rock or two being out of place on Mount Everest.


Well, perhaps on some level, but aliasing is nasty, folding products way down into the audio spectrum.  You'd hate it if you heard it.  Less aliasing is better, aliasing can be audible and not harmonically related, whereas phase distortion around filter cutoff is not as easily audible.


----------



## pinnahertz

Cutestudio said:


> I understand how this applies to continuous tones, but not how it applies to transients.


Why would it be different?


Cutestudio said:


> That "uncorrelated noise" is the error being spread (and resolved) over time - is it not?


No, that's not quite right.  One way to look at it is changing a hard quantization threshold to a smooth transition by virtue of uncorrelated noise.


Cutestudio said:


> If so, don't we need the error fixing at the time of the transient, rather then over the next few hundred samples?


No, a transient is typically a higher level signal, and wouldn't be affected (or need "error fixing") until it gets way down into the noise floor.


----------



## castleofargh

Cutestudio said:


> Jiggling a signal that’s not causing a bit transition pushes it over the transition point for an amount statistically proportional to its actual amplitude level.
> 
> My argument is that for short transient sounds in music you haven't got enough samples for those statistics to work.
> The correct fix for quantisation error in 16bit audio is therefore to switch to 24bit audio, which today is simple and almost without cost.
> ...


I do think it is unreasonable in the hifi world. maybe you should get some measurement gears and test what your system, and listening environment do to the music in real life. if you think that dither or any form of added noise is damaging the music, then you need to keep that view and agree that any other and louder noises or distortions are doing more damage and most likely making the dither noise irrelevant because of the difference in magnitude.
but I don't even see audio that way. all I see is one sound + one sound + one sound + some noises, all in one wave per channel. all the sounds are entirely contained(but band limited)in the file. some extra noise isn't changing the instrument, some extra noise is only making extra noise. unless you have so few bits that you start cutting the amplitude of the signal itself, which in this case would require a music piece of more than 90dB of dynamic, quantization noise is only noise. extra noise down at around -90dB. do you hear such noise in normal listening conditions? no. and if you did somehow in a strange and unlikely situation, after dither it would be even less noticeable. 

anyway, there is nothing wrong with using 24bit aside from making the file bigger and costing more. if you're fine with that, enjoy. I'm arguing the motivation for using 24bit in a playback system, not the use of 24bit.


----------



## gregorio

Cutestudio said:


> [1] My argument is that for short transient sounds in music you haven't got enough samples for those statistics to work....
> [1a] That "uncorrelated noise" is the error being spread (and resolved) over time - is it not?
> [1b] The correct fix for quantisation error in 16bit audio is therefore to switch to 24bit audio ....
> [2] Classical music has a wide dynamic range with quiet stuff being played at reasonable volume. With a decent amp and speakers: or decent headphones, this is not an unreasonable scenario in the HiFi world.
> [2a] ...What's the drawback of using a format that's better than we need?



1. Two points: Firstly, as long as the transient contains frequencies with a waveform duration of at least 2 samples then we can encode and dither it perfectly. With CD (16/44.1) that means transient peaks with freq content no higher than 22kHz. Secondly, it doesn't matter what amplitude values we have or how quickly the amplitude changes, providing of course we do not exceed the clipping point (0dBFS). 
1a. No, the error is not spread over time. I think you're getting confused by the fact that the quantisation error of each sample is rounded but as each sample is rounded up or down randomly with dither, you obviously need several samples for it to be apparent that the rounding is random (and therefore statistically perfect). Dither operates as well (effectively perfectly) on transient peaks as it does on any other variation of amplitude (within the Nyquist limit) and the dither not appearing random could only potentially be a problem with commercial recordings which only last a few dozen micro-secs and of course, there aren't many of those! BTW, I say "potentially" because in practice you couldn't identify any sort of quantisation error in a recording of such short duration.
1b. No, 24bit does not fix quantization error, it just reduces it. Dither on the other hand does fix quantization error, after dither there is no quantization error at all and therefore, dithered 16bit is more accurate than un-dithered 24bit.
2. Yes, classical music has a very wide dynamic range. Symphony orchestra recordings typically have the widest dynamic range of any music genre, up to a maximum of about 60dB, which is the equivalent of about 10bits ...
2a. Absolutely true and that's what we already have with 16bit! A typical commercial music recording has a dynamic range equivalent to about 8 or fewer bits but it can go as high as about 10bits, as mentioned in #2. 16bits therefore provides over 100 times more dynamic range than is typically used and about 40 times more than is required even for the more extreme symphony orchestra recordings. How much more "better" do you want? Apart from another 8bits of inaudible noise and audio files which are a third larger, what do you think you actually gain from 24bit?

G


----------



## Cutestudio

bigshot said:


> What kind of transient are you going to find in music that doesn't cover hundreds of samples?



It's a good question, here's a clip that has 93 samples in:







Some of those shapes in the selected area are formed by a very small number of samples.

When I look at that waveform I find that each vertical point is quantized to a particular level, so for instance a distant (quiet) 'click' sound at an average level of -60dB will have a quantisation error of around 1/255 of fsd. This error will be there regardless of dither:  imagine you can print 50 samples out onto clear sheet of an 8 bit non dithered, 8 bit dithered, and the analog, and you placed them all on a lightbox to they overlapped perfectly.

Then as a test you could print out the analog waveform with some pure analog white noise of equal amplitude to the dither signal, and lay that on top for a 4th version of that waveform.

My objection to settling for the 'good enough' 16bit over the easily accessible 'better than we need' 24bit is therefore demonstrated by the errors in these waveform shapes.

I am aware that the maths in the continuous case says that it's all just a type of noise, but isn't any distortion a type of noise? 
The fact remains that if the aim is to preserve the waveform shape (surely the very definition of 'High Fidelity'?) why are we arguing that 16bit is an acceptable solution when the waveform snapshots clearly show errors for transients in quiet passages?

I'd understand the defence of 16bit more if we'd hit a real issue with purity of iron cores, resistance of silver wire etc but moving from 16 to 24bits? In todays age of 1920x1080 Netflix streaming, huge storage,quad core 1.4GHz phones and smart fridges: 24bit  is trivial. In fact we did it two decades ago back when everyone thought Windows 98 was good, with SACD and DVD-A.
https://en.wikipedia.org/wiki/DVD-Audio


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## gregorio

Cutestudio said:


> [1] When I look at that waveform
> [2] so for instance a distant (quiet) 'click' sound at an average level of -60dB will have a quantisation error of around 1/255 of fsd. This error will be there regardless of dither ...
> [3] My objection to settling for the 'good enough' 16bit over the easily accessible 'better than we need' 24bit is therefore demonstrated by the errors in these waveform shapes.
> [4] I am aware that the maths in the continuous case says that it's all just a type of noise ...
> [5] The fact remains that if the aim is to preserve the waveform shape (surely the very definition of 'High Fidelity'?) why are we arguing that 16bit is an acceptable solution when the waveform snapshots clearly show errors for transients in quiet passages?



1. You are not looking at a waveform!!! You are looking at a graphical representation of the data point on your disk, NOT the waveform once it has been converted (via a dithering quantiser in the DAC)!
2. Again, NO! It will have zero error after dither.
3. No one is arguing for "good enough" (10bit audio), we are arguing for at least 40 times "better than we need" which is what we get with 16bit. What do you not understand?
4. Yes, white noise that's 40 times lower than the noise floor of any commercial recording! What is there here which you do not understand?
5. The only fact which remains is that you're refusing to understand how digital audio works and are using any audiophile myth you can find to prove a point about 24bit which is false! The fact is, that with dither the waveform shape is perfect, down to about -92dBFS with dither and then it's still perfect but covered by white noise.

Answer the question please, how much more than 40 to 100+ times more dynamic range than is ever used do you want, and why?

G


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## bigshot (Jun 19, 2017)

pinnahertz said:


> What you refer to here is actually the "envelope" of the signal, and your numbers are fine for that. The envelope is the overall amplitude vs time graph of the hit.  If you zoom in and look at that waveform a bit more carefully and note the specific rise-times involved at various points, you'll see there are components of that sound that rise much faster than 2-3ms, fast enough to produce frequency components all the way up to 20kHz and beyond. Less aliasing is better, aliasing can be audible and not harmonically related, whereas phase distortion around filter cutoff is not as easily audible.




The envelope of the sound is the sound that we recognize as a drum hit when we hear it. Individual frequency waveforms within that are part of it, but on a very small scale. If a recording medium is able to capture a frequency up to 20kHz even with anti-aliasing, I can't for the life of me see how anti-aliasing could affect transient time in a drum hit to the degree it isn't audibly the same drum hit any more. That's like saying one pebble out of place on Mount Everest changes what Mount Everest is. I think we're saying the same thing, but I'm standing back a bit and looking at the overall sound of a musical transient and you're pulling out a microscope and looking at one sliver of it. I always find that it's easier for me to keep a realistic perspective if I keep focused on the overall stuff that really matters.

My general point here is that the way modern DACs handle the top end, whether it uses a brick wall above 20kHz or a roll off above 20kHz, it doesn't matter because the difference between those two isn't audible. I see you saying basically the same thing. You're just assuming that because there is some theoretical difference in phase distortion, you add that qualifier "not as easily audible". I don't need to do that because the difference is so small it's clearly below the threshold of perception. Our ability to hear phase error is in the millisecond range. The phase error of cutoff filters is well below the threshold of perception. The transients of drum hits in recorded music exist on an even larger scale than that. Therefore, worrying that transients in music might be affected by phase distortion from cutoff filters is essentially running down a mental rabbit hole. Audiophiles of all kinds tend to do that too often because they can't see the forest for the trees.


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## bigshot (Jun 19, 2017)

I just realized I conflated this discussion with the one about slow and fast rolloff. Sorry about that!

It's OK though, because I had the audacity to ask a fella over there if what he was talking about was audible to human ears. He said that question was "derailing the thread". There ya go!


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## Cutestudio (Jun 21, 2017)

gregorio said:


> 1. You are not looking at a waveform!!! You are looking at a graphical representation of the data point on your disk, NOT the waveform once it has been converted (via a dithering quantiser in the DAC)!
> 2. Again, NO! It will have zero error after dither.
> 3. No one is arguing for "good enough" (10bit audio), we are arguing for at least 40 times "better than we need" which is what we get with 16bit. What do you not understand?
> 4. Yes, white noise that's 40 times lower than the noise floor of any commercial recording! What is there here which you do not understand?
> ...



I'm not sure why you are saying I'm not looking at the waveform, do you suggest I get inside the wires and watch the electrons drifting back and forth?

I don't think you realise what dither actually is, it's just noise added before quantisation to even out the quantisation errors, it doesn't actually make each little wavelet have the right shape, each point will still be quantized to the wrong value.

There's a Wikipedia article that's fairly good at explaining:
https://en.wikipedia.org/wiki/Dither



			
				Wikipedia said:
			
		

> If a series of random numbers between 0.0 and 0.9 (ex: 0.6, 0.1, 0.3, 0.6, 0.9, etc.) are calculated and added to the results of the equation, two times out of ten the result will truncate back to 4 (if 0.0 or 0.1 are added to 4.8) and the rest of the times it will truncate to 5, but each given situation has a random 20% chance of rounding to 4 or 80% chance of rounding to 5. Over the long haul this will result in results that average to 4.8 and a quantization error that is random — or noise.



Note the phrase 'Over the long haul' as a clue to the temporal difficulties: the quantisation is still there, dither is not magic, it just trades the repeating error into noise over time.
This noise is also itself quantised of course and there are various home brew dithers that sound better than others, but it's not analogue noise, it's still stuck to a number of levels like a randomised PWM.
For instance a brief level of 0.5 bit is transformed from a truncated 0 with dither to a random 50/50 split between 0 and 1 which one hears as noise. Not a nice noise as in analog, but a rather coarse noise: try it on an 8 bit waveform and listen.

Why exactly are you against 24bit, do you need a bigger disk or is it metering charges on the internet?

Here's a study that found 24bit was more perfect than 16bit BTW:
http://www.aes.org/tmpFiles/elib/20170620/18296.pdf

But even disregarding that, I'm puzzled by the fight for 'good enough' when clearly 24/96 is not only achievable with ease but it seems used by everyone outside of audio: why have the HiFi crowd got the worst format? Is the obsolete CD format really that important anymore? I can't recall the last time I listened to a CD on a CD player, they arrive in the post and get ripped that day.
Perhaps the lack of easily available iTunes downloads at 96/24 isn't the record companies fault after all, but out fault for fighting and insisting we get sold an inferior format?
Why are we demanding mediocrity in audio? Do we turn down 100W amps because we may only want 2W to use? Is that speaker too good for us so we get a lower grade one? Is that steel beam holding up the house too good for the job and we demand a smaller one? Format wars appear to be the only branch of HiFi where people demand less, rather than more.


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## pinnahertz

Cutestudio said:


> But even disregarding that, I'm puzzled by the fight for 'good enough' when clearly 24/96 is not only achievable with ease but it seems used by everyone outside of audio: why have the HiFi crowd got the worst format?


Where is 24/96 used outside of audio?


Cutestudio said:


> Is the obsolete CD format really that important anymore?


Obsolete means "no longer produced or used, out of date".  The vast majority of music content is released and in 16/44.1...today....but it's also in a compressed, bit-rate reduced format, either mp3 or AAC.  The vast majority of 24 bit content released and sold today is nothing more than up-sampled from 16/44.1, and therefore no better. 


Cutestudio said:


> I can't recall the last time I listened to a CD on a CD player, they arrive in the post and get ripped that day.
> Perhaps the lack of easily available iTunes downloads at 96/24 isn't the record companies fault after all, but out fault for fighting and insisting we get sold an inferior format?


That's nonsense.  Music is purchased for the music, not the level of quantization, except for a very tiny splinter market that buys it only for the level of quantization, not realizing or caring they're still buying the original quantization level anyway.  Then there's the true hi-res market that actively seeks out real 24 bit content with provenance all the way back through the production chain.  That market is so small as to be static in the statistics. 


Cutestudio said:


> Why are we demanding mediocrity in audio?


You're being ridiculous now.  Nobody does that.  They're buying what's available.  And frankly, what's available in the most popular music today could be resampled to 8 bits without noticeable degradation.  And the rest of it is represented perfectly in 16 bits.  


Cutestudio said:


> Do we turn down 100W amps because we may only want 2W to use?


No, but some people buy low power amps because the think the active devices and topology used is somehow better.  5W tube amps are actually fairly popular within the tube amp category. 


Cutestudio said:


> Is that speaker too good for us so we get a lower grade one? Is that steel beam holding up the house too good for the job and we demand a smaller one? Format wars appear to be the only branch of HiFi where people demand less, rather than more.


Nobody demands lower quality unless it's cheaper.  And then, cheaper almost always wins the toss.  Welcome to the world market.  The better more expensive mouse trap doesn't sell, the cheaper ones that still do the job do.  Don't be mislead into thinking that the audiophile market is a perfect model of the market in general.


----------



## Cutestudio

pinnahertz said:


> Where is 24/96 used outside of audio?



Lots of places, even movie sound tracks have better formats now.
https://en.wikipedia.org/wiki/DVD-Video



pinnahertz said:


> Obsolete means "no longer produced or used, out of date".


Yes. 30 years out of date.



pinnahertz said:


> The vast majority of music content is released and in 16/44.1...today....but it's also in a compressed, bit-rate reduced format, either mp3 or AAC.  The vast majority of 24 bit content released and sold today is nothing more than up-sampled from 16/44.1, and therefore no better.


Sad isn't it. 



pinnahertz said:


> That's nonsense.  Music is purchased for the music, not the level of quantization, except for a very tiny splinter market that buys it only for the level of quantization, not realizing or caring they're still buying the original quantization level anyway.  Then there's the true hi-res market that actively seeks out real 24 bit content with provenance all the way back through the production chain.  That market is so small as to be static in the statistics.





pinnahertz said:


> You're being ridiculous now.  Nobody does that.  They're buying what's available.



No, not nonsense or being ridiculous, my opinions and view are mine and you are free to disagree, but please don't belittle people you disagree with.
You also appear to contradict yourself here - saying there is no demand for a decent format but then saying people buy what is available.

Threads like this however make me realise that for every audiophile who would want a better format there is a crowd of people just waiting to rush up to explain why they don't need it, and in this way we stay with these pointless discussions and no progress is made. Any time a music industry product manager looks at these threads they see the vocal defence of 16/44.1 which is frankly rather sad.

It looks like for the best formats we can only look to the video industry and hope one day that the format of dedicated audio will ever be as good, but if this thread is any indicator, I think we'll always be using the inferior, obsolete 16/44.1 format and playing with dithers (not a good idea if you are feeding the output into a digital room EQ or crossovers) and upsampling.


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## Niouke

Cutestudio said:


> I'm not sure why you are saying I'm not looking at the waveform, do you suggest I get inside the wires and watch the electrons drifting back and forth?
> Why exactly are you against 24bit, do you need a bigger disk or is it metering charges on the internet?
> 
> Here's a study that found 24bit was more perfect than 16bit BTW:



24bits implications: 
Bigger storage space
Metered internet charge/Bandwith limitation
Bigger decoding power required -> Battery usage, power demands

I don't think anyone here is "against" 24bits, it is just as you stated "more perfect" and unneeded to achieve the best listening experience possible (for humans at least). Also the marketing hype surrounding it and the attempt to sell unneeded overpriced equipment is disgusting.


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## Niouke

regarding DVD video, it has been discussed around here. I'll let the wikipedia page speak for itself:

AC-3: 48 kHz sampling rate, 1 to 5.1 (6) channels, up to 448 kbit/s;
DTS: 48 kHz or 96 kHz sampling rate; channel layouts = 2.0, 2.1, 5.0, 5.1, 6.1; bitrates for 2.0 and 2.1 = 377.25 and 503.25 kbit/s, bitrates for 5.x and 6.1 = 754.5 and 1509.75 kbit/s;[12]
MP2: 48 kHz sampling rate, 1 to 7.1 channels, up to 912 kbit/s.
these are lossy codecs, note that the bitrates given are the total bitrates, all channels added, and are nowhere near PCM 44.1/16


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## bigshot (Jun 21, 2017)

I suppose I could wrap my sandwich in a 40 gallon lawn and leaf trash bag and seal my letters with 3 inch fiber reinforced duct tape, but a sandwich bag and licking the envelope work just as well. File size does matter. I have a large music library with over a year and a half worth of music on it. It all fits on a 2TB hard drive with room to spare. That makes it easy for me to back up, stream over my home wifi network, and load onto my mobile devices. I could store everything at 24/192 but it would be much less convenient and wouldn't sound a bit better. Super high bitrates in blu-ray movies are just as much of an advertising gimmick as "HD Audio" in music. There are people willing to pay extra for sound they can't hear. The nice big numbers give them peace of mind, so the format allows for that. Regular plain vanilla CD sound quality is already into the range of overkill. Lossy audio can be audibly transparent and sound just as good to human ears.


----------



## pinnahertz

Cutestudio said:


> Lots of places, even movie sound tracks have better formats now.
> https://en.wikipedia.org/wiki/DVD-Video


You said "outside of audio". That is all audio.


Cutestudio said:


> Yes. 30 years out of date.


You said "obsolete".  Last I looked (like a few seconds ago) CDs are still produced by the millions, and as I said, the vast majority of all distributed music is distributed at 16/44.1.  That makes it not "obsolete" or "out of date" by any definition.


Cutestudio said:


> Sad isn't it.


I'm not clear on what you mean.  What about it is sad?



Cutestudio said:


> No, not nonsense or being ridiculous, my opinions and view are mine and you are free to disagree, but please don't belittle people you disagree with.


You were making statements as fact.  They didn't sound like opinions, and were not based in reality.  You can desire a (theoretically) higher resolution format, that doesn't mean because you do, and because a tiny bit of the total produced music is available that way, doesn't mean 16/44.1 is obsolete, outdated, or inadequate.


Cutestudio said:


> You also appear to contradict yourself here - saying there is no demand for a decent format but then saying people buy what is available.


That is not a contradiction, that is simply stating fact.


Cutestudio said:


> Threads like this however make me realise that for every audiophile who would want a better format there is a crowd of people just waiting to rush up to explain why they don't need it, and in this way we stay with these pointless discussions and no progress is made.


Progress is driven by demand for advance, or the solving of a fundamental inhibiting problem.  These discussion don't inhibit or promote anything.


Cutestudio said:


> Any time a music industry product manager looks at these threads they see the vocal defence of 16/44.1 which is frankly rather sad.


They're not even looking at this at all, and even if they did, a half-dozen bickering old guys does not constitute responsible market research.  

It doesn't matter to the industry unless you can actually sell it.  To sell it, the product has to fill a significant (not a tiny splinter) demand.  The bulk of the demand for music is today satiated by YouTube...free music, on demand, and you don't even have to buy anything.  The rest is 16/44.1 downloads.  The quality limiter in those files is all in creative choices and mastering choices driven by the industry's percieved competitive evaluation of the market.


Cutestudio said:


> It looks like for the best formats we can only look to the video industry and hope one day that the format of dedicated audio will ever be as good,


You mean the film industry, not the video industry.  Film audio in consumer distribution has the capability of 24 bit at various rates, but not all released films are released that way.  The reason that any of it is available in 24 bit is because the film industry has a release chain that can handle it without incompatibility, and the production chain is all 24/48.  Hence the standard rate for film distribution (and video only productions) is 48kHz, but 24bits is not as standard.   

The music industry does not have that cohesive and universal release chain, they must always release in 16/44.1, and can do other formats if the feel there's a financial advantage, which mostly there is not, and so they don't.


Cutestudio said:


> but if this thread is any indicator, I think we'll always be using the inferior, obsolete 16/44.1 format


The factual definition of "obsolete" does not apply to 16/44.1, regardless of your opinion.  Inferior, well, you can have that opinion if you want, there's no actual proof.


Cutestudio said:


> and playing with dithers (not a good idea if you are feeding the output into a digital room EQ or crossovers) and upsampling.


Digital room EQ and crossovers all operate at at least 24 bits, and are downstream of the volume control.  The fact that all that horrible and unlistenable 16/44.1 material is dithered has absolutely no impact on what room EQ and crossovers have to do.  None at all!  Just to be clear, this is NOT an opinion, this is FACT.  A volume control in a typical system spends most of its time at -20 or -10, that puts any dither signal at -113 to -103, which is below the residual noise floor of any 24 bit EQ or crossover.  With the volume control at "0", the noise floor in the recording will be the limiting factor (or the room), and not dither.  Dither signals, the good ones, are shaped to minimize audible noise contribution and maximize their intended purpose.


----------



## bigshot

My sub major in college was English. Maybe that's why line by line commenting makes me just want to skip over the whole post. I blame the internet for chopping up thoughts into tiny bits and forcing the reader to reassemble them like a jigsaw puzzle. I like nice paragraphs that start out with a clear premise, followup proofs and examples, and finish off with a neat summation. Assemble a few of those together with logical flow and you end up with a convincing argument.


----------



## pinnahertz

bigshot said:


> My sub major in college was English. Maybe that's why line by line commenting makes me just want to skip over the whole post. I blame the internet for chopping up thoughts into tiny bits and forcing the reader to reassemble them like a jigsaw puzzle. I like nice paragraphs that start out with a clear premise, followup proofs and examples, and finish off with a neat summation. Assemble a few of those together with logical flow and you end up with a convincing argument.


Tell me now, honestly, would it work better for you for me to number my responses?  I find that just a tad more confusing for me, but if reads better...  

I do prefer to respond to individual points, writing a long paragraph makes people have to extract and apply the responses to the original post.  I agree, it chops things up, but it kind of dumbs it down to the specific.  I'm willing to change, though.


----------



## pinnahertz

bigshot said:


> I suppose I could wrap my sandwich in a 40 gallon lawn and leaf trash bag and seal my letters with 3 inch fiber reinforced duct tape, but a sandwich bag and licking the envelope work just as well. File size does matter. I have a large music library with over a year and a half worth of music on it. It all fits on a 2TB hard drive with room to spare. That makes it easy for me to back up, stream over my home wifi network, and load onto my mobile devices. I could store everything at 24/192 but it would be much less convenient and wouldn't sound a bit better. Super high bitrates in blu-ray movies are just as much of an advertising gimmick as "HD Audio" in music. There are people willing to pay extra for sound they can't hear. The nice big numbers give them peace of mind, so the format allows for that. Regular plain vanilla CD sound quality is already into the range of overkill. Lossy audio can be audibly transparent and sound just as good to human ears.


The lawn leaf trash bag and reinforced duct tape...  funny!

Did you ever calculate how big your storage would have to be if your entire library were 24/96? Or uncompressed 16/44.1?

Just an observation, the Blue-ray disc is not a very successful format. Download/streaming/buying content has won that war, and none of that is uncompressed 24/96, not even close.  And sounds just fine.

In the big picture, the quality of the content hasn't been the limiting factor for some time, it's the reproducing system and the environment its in.  The 16/44.1 container has exceeded general reproducing capabilities since inception.


----------



## WoodyLuvr

bigshot said:


> I suppose I could wrap my sandwich in a 40 gallon lawn and leaf trash bag and seal my letters with 3 inch fiber reinforced duct tape, but a sandwich bag and licking the envelope work just as well. File size does matter. I have a large music library with over a year and a half worth of music on it. It all fits on a 2TB hard drive with room to spare. That makes it easy for me to back up, stream over my home wifi network, and load onto my mobile devices. I could store everything at 24/192 but it would be much less convenient and wouldn't sound a bit better. Super high bitrates in blu-ray movies are just as much of an advertising gimmick as "HD Audio" in music. There are people willing to pay extra for sound they can't hear. The nice big numbers give them peace of mind, so the format allows for that. Regular plain vanilla CD sound quality is already into the range of overkill. Lossy audio can be audibly transparent and sound just as good to human ears.


Can't help but remember the days when we are all happy with vinyl at 30-35 kHz if we were extremely lucky... which btw equates to less than 11 bits of resolution!

A long time ago there was an interesting listening test that took a 44 kHz/16-bit clip and converted it to 11 to 15 bits with dither... the test showed that most people could distinguish 11 bits, some 12 bits, but very, very very few could tell 13-14 bits or more.


----------



## pinnahertz

WoodyLuvr said:


> Can't help but remember the days when we are all happy with vinyl at 30-35 kHz if we were extremely lucky... which btw equates to less than 11 bits of resolution!


I don't recall ever being happy with vinyl.  I pushed it as far as it could possibly go, it always fell short.  Always, and in several aspects.  It has several unsolvable problems, and those of us working with it professionally were never unaware of those limitations.  They weren't small, and they were audible.  And even the earliest digital systems nuked pretty much all of those problems. 

30-35kHz isn't a real figure on vinyl apart from distortion products and CD-4 carriers which required a special stylus and cart to recover.  Getting good equalized flat response to 20kHz was challenging and limited to fairly low level signals.  

And, apart from all that nasty "phase distortion" in digital audio...you should see what a square wave looks like coming off a phono preamp and cart!  Blech. 


WoodyLuvr said:


> A long time ago there was an interesting listening test that took a 44 kHz/16-bit clip and converted it to 11 to 15 bits with dither... the test showed that most people could distinguish 11 bits, some 12 bits, but very, very very few could tell 13-14 bits or more.


I recall that, can't remember where I saw it.  AES paper? I also recall the results being highly condition specific.


----------



## WoodyLuvr

pinnahertz said:


> I don't recall ever being happy with vinyl.


Not even a happy memory of Christmas LPs playing in the background as a kid at your grandparent's house?


----------



## StanD

WoodyLuvr said:


> Not even a happy memory of Christmas LPs playing in the background as a kid at your grandparent's house?


Perhaps this is in the context of listening to music in a more pure sense rather than holiday nostalgia. He was probably too busy trying to score some spiked egg nog.


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## bigshot (Jun 21, 2017)

I have great sounding LPs. They're perfectly serviceable. Personally, I think that the last real achievement in sound quality (aside from multichannel sound) was stereo in 1952. Everything since then has been serviceable... even cassette tapes and 8 tracks. CD cleaned up the niggling details and made it a skosh beyond perfect... which is very nice.

But the odd thing is, for some reason it's VERY difficult to transfer an acoustic recording made in the pre-electric era to digital and have it sound remotely like it was intended to sound. The dynamic range and volume of an acoustic phonograph playing a Caruso record is awe inspiring, but on CD it sounds thin and distant. I did some experiments transferring old records and was able to match the sound well, and even get it to sound a little bit better than on a good acoustic phonograph, but it took a lot of work and processing. It also took throwing out any attempt at reducing horn resonances or squeezing a natural response curve out of it. I had to go for recreating the distortions the phonograph was adding in playback. The sound of the phonograph was a big part of the sound of the music.



pinnahertz said:


> I do prefer to respond to individual points, writing a long paragraph makes people have to extract and apply the responses to the original post.  I agree, it chops things up, but it kind of dumbs it down to the specific.  I'm willing to change, though.



I find it is clearer and easier to read if  I quote as little as possible and always restate the context in my answer at the beginning and end of each paragraph. That way people's eyes can read the flow of my comments straight ahead without having to apply a numerical key or jump back and forth between two completely different voices. On the internet, people generally blow through writing pretty fast, so it helps get points across if I organize my comments into easy to read chunks. It also makes it easier for people to reply to my comments, because they aren't dependent on a second level quote for context. Brevity is the soul of wit and concise is nice!


----------



## pinnahertz

bigshot said:


> I have great sounding LPs. They're perfectly serviceable. Personally, I think that the last real achievement in sound quality (aside from multichannel sound) was stereo in 1952. Everything since then has been serviceable... even cassette tapes and 8 tracks. CD cleaned up the niggling details and made it a skosh beyond perfect... which is very nice.
> 
> But the odd thing is, for some reason it's VERY difficult to transfer an acoustic recording made in the pre-electric era to digital and have it sound remotely like it was intended to sound. The dynamic range and volume of an acoustic phonograph playing a Caruso record is awe inspiring, but on CD it sounds thin and distant. I did some experiments transferring old records and was able to match the sound well, and even get it to sound a little bit better than on a good acoustic phonograph, but it took a lot of work and processing. It also took throwing out any attempt at reducing horn resonances or squeezing a natural response curve out of it. I had to go for recreating the distortions the phonograph was adding in playback. The sound of the phonograph was a big part of the sound of the music.
> 
> ...


Clearly you needed 24 bits.


----------



## bigshot (Jun 21, 2017)

If I was recording a phonograph in a recording studio, I would definitely want to record in 24 bit. The peaks on Caruso records make your ears ring and can be heard a block away, and the sotto parts are in the same volume range as a natural whisper. It goes beyond just what's cut into the record. Specific brands of acoustic records were designed to be played back on specific brands of phonographs. They tuned the recordings to the acoustics of the machines. It's almost like a Dolby pre-emphasis thing. The mica diaphragm that vibrates to make the sound has its own effect, and it tends to emphasize some frequencies and expand the dynamic range by ringing with loud clear tones in a particular frequency range. (Kind of like a kazoo, except more controlled.) Horn resonances do things to the sound as well, and the placement of the phonograph in the room can make a big difference too. The sweet spot lies smack dab on the male tenor voice. There's more to the sound than just what's in the grooves. I learned a lot about sound reproduction from experimenting with my antique phonographs.

Early on in the days of multichannel there was a company that put out four channel recordings of a really good phonograph with a huge horn playing acoustic recordings in a concert hall. I never heard any of them because it was made in a strange obsolete format, but I regret not being able to hear the sound of those recordings. In 2 channel, they sucked.


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## pinnahertz

bigshot said:


> If I was recording a phonograph in a recording studio, I would definitely want to record in 24 bit. The peaks on Caruso records make your ears ring and can be heard a block away, and the sotto parts are in the same volume range as a natural whisper. It goes beyond just what's cut into the record. Specific brands of acoustic records were designed to be played back on specific brands of phonographs. They tuned the recordings to the acoustics of the machines. It's almost like a Dolby pre-emphasis thing. The mica diaphragm that vibrates to make the sound has its own effect, and it tends to emphasize some frequencies and expand the dynamic range by ringing with loud clear tones in a particular frequency range. (Kind of like a kazoo, except more controlled.) Horn resonances do things to the sound as well, and the placement of the phonograph in the room can make a big difference too. The sweet spot lies smack dab on the male tenor voice. There's more to the sound than just what's in the grooves. I learned a lot about sound reproduction from experimenting with my antique phonographs.
> 
> Early on in the days of multichannel there was a company that put out four channel recordings of a really good phonograph with a huge horn playing acoustic recordings in a concert hall. I never heard any of them because it was made in a strange obsolete format, but I regret not being able to hear the sound of those recordings. In 2 channel, they sucked.


I understand all of that, and have experience with acoustic recordings and acoustic phonographs.  

But, even though you would probably use it, 24bits would be a complete waste to record acoustic records and machines.  The total dynamic range is very small, strongly limited by a high noise floor..


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## old tech

bigshot said:


> If I was recording a phonograph in a recording studio, I would definitely want to record in 24 bit. The peaks on Caruso records make your ears ring and can be heard a block away, and the sotto parts are in the same volume range as a natural whisper. It goes beyond just what's cut into the record. *Specific brands of acoustic records were designed to be played back on specific brands of phonographs.* They tuned the recordings to the acoustics of the machines. It's almost like a Dolby pre-emphasis thing. The mica diaphragm that vibrates to make the sound has its own effect, and it tends to emphasize some frequencies and expand the dynamic range by ringing with loud clear tones in a particular frequency range. (Kind of like a kazoo, except more controlled.) Horn resonances do things to the sound as well, and the placement of the phonograph in the room can make a big difference too. The sweet spot lies smack dab on the male tenor voice. There's more to the sound than just what's in the grooves. I learned a lot about sound reproduction from experimenting with my antique phonographs.
> 
> Early on in the days of multichannel there was a company that put out four channel recordings of a really good phonograph with a huge horn playing acoustic recordings in a concert hall. I never heard any of them because it was made in a strange obsolete format, but I regret not being able to hear the sound of those recordings. In 2 channel, they sucked.


Is that the different pre-emphasis applied to early records before the RIAA standard was adopted?

I remember some of those early 50s/60s record players which had a knob to select various labels, eg RCA, Columbia etc.

Does that mean that to get the optimal sound from those early (pre RIAA) records today, one would need to know the de-emphasis curve and somehow implement it in the pre-amp chain?


----------



## gregorio

Cutestudio said:


> [1] No, not nonsense or being ridiculous, my opinions and view are mine and you are free to disagree, but please don't belittle people you disagree with.
> [2] Threads like this however make me realise that for every audiophile who would want a better format there is a crowd of people just waiting to rush up to explain why they don't need it, and in this way we stay with these pointless discussions and no progress is made.



1. You are free to have any opinion that you want. However, if you want the opinion that 1 + 1 = 3, then yes, you are going to get somewhat belittled in a science forum because your opinion goes against basic fact. So, either you should make an effort to understand the basic facts or if you want to stick with your 1 + 1 = 3 opinion, then maybe a science forum isn't the right place to post it.
2. The discussion is only pointless when an audiophile insists on arguing that 1 + 1 = 3. You can argue for a better format all you want but within the audible range there is no better format than 16bit, 1 + 1 = 2! 



Cutestudio said:


> [1] I'm not sure why you are saying I'm not looking at the waveform ...
> [2] For instance a brief level of 0.5 bit is transformed from a truncated 0 with dither to a random 50/50 split between 0 and 1 which one hears as noise. Not a nice noise as in analog, but a rather coarse noise: try it on an 8 bit waveform and listen.
> [2a] I don't think you realise what dither actually is, it's just noise added before quantisation to even out the quantisation errors, it doesn't actually make each little wavelet have the right shape, each point will still be quantized to the wrong value.
> [3] Why exactly are you against 24bit, do you need a bigger disk or is it metering charges on the internet?
> [4] But even disregarding that, I'm puzzled by the fight for 'good enough' .... [4a] Why are we demanding mediocrity in audio? [4b] Do we turn down 100W amps because we may only want 2W to use? ... [4c] Format wars appear to be the only branch of HiFi where people demand less, rather than more.



1. Because you are NOT looking at a waveform, you are looking at a graphical representation of the digital data. Zoom in to that representation and you'll see the discrete quantisation values, the "stair step" image BUT that's just the encoded digital data not the decoded data that come out of the DAC. Once decoded, there is no "stair step", a continuous analogue waveform!
2. It's not a nice or nasty noise, it's white noise and, it's way below the noise floor of the recording/reproduction with 16bit.
2a. I don't think you know what dither is, you need to learn that 1 + 1 = 2, not 1.5 or 3! The result of dithering is a perfect, error free signal plus some benign white noise!
3. And why are you just making up nonsense to defend your ridiculous "opinion"? I'm not in the least bit against 24bit, I've been using it every day since long before you even knew it existed! 24bit is very useful, just not for a consumer music distribution format.
4. And I'm puzzled as to why you keep repeating that nonsense when NO ONE is arguing for "good enough". We want many times better than good enough, which is why we have 16bit.
4a. I don't know, why are you demanding mediocrity in music? Why are you demanding higher data rate formats which make no audible difference whatsoever, instead of demanding better quality (not mediocre) music?
4b. OK, you want to avoid the question asked, so let's use your analogy. Let's say the vast majority of the time you only use about 2W but sometimes you use as much as 25W, although never more than 25W: Why would a 1000W amp not be enough? What benefit would you get from a 100,000W amp that you wouldn't already get from using a 1000W amp?
4c. Who's demanding less? I'm demanding perfect fidelity from digital audio, what more is there and what are you arguing for? If you're going to attempt a response to this question, just a reminder; this is the science forum and your opinion that 1 + 1 = 3 is not valid here!

G


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## pinnahertz

old tech said:


> Is that the different pre-emphasis applied to early records before the RIAA standard was adopted?


No, he's referring to acoustic recordings where the actual characteristics were unknown, but different as a result of the materials and design of the recording device.


old tech said:


> I remember some of those early 50s/60s record players which had a knob to select various labels, eg RCA, Columbia etc.


Those were all 78rpm "electrical" recordings where the actual recording curve was known, the switch matched that curve.


old tech said:


> Does that mean that to get the optimal sound from those early (pre RIAA) records today, one would need to know the de-emphasis curve and somehow implement it in the pre-amp chain?


Yes.  There have been many preamps with the ability to match the various recording characteristics.  *Here's a current one.*


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## bigshot (Jun 22, 2017)

pinnahertz said:


> But, even though you would probably use it, 24bits would be a complete waste to record acoustic records and machines.  The total dynamic range is very small, strongly limited by a high noise floor..




There would be no point to 24 bits if you used electrical transcription. But if you were miking an acoustic phonograph, you might actually need it. As I mentioned before, the machine itself expands the dynamics in the recording. My Victrola produces ear splitting volumes with Caruso records with a very low noise floor. I think that has to do with the way the sound box and mica diaphragm react to energy in certain frequency ranges. In Caruso's high Cs it amplifies it by ringing in a very loud and natural sounding way. In the higher frequencies where surface noise occurs, it attenuates it. The horn also amplifies and has particular resonances. These things effectively expand the dynamic range beyond the level encoded in the grooves of the record.



old tech said:


> Is that the different pre-emphasis applied to early records before the RIAA standard was adopted?



This is even before electricity. The same companies that made phonographs also sold records... Victor made Victrollas and sold Victor records, Columbia made machines and records, the same with Brunswick and Edison. Each company had a sound lab whose job it was to design the recording equipment to suit the machines they made, and vice versa. Some companies' records could be only played on their own brand of machine, like Edison and Pathe. You couldn't even play an Edison or Pathe disk on a Victor phonograph.

The music was recorded by performing into a large horn which funneled into a cutting lathe that cut the grooves in a wax master. A clockwork motor ran the lathe. The records were played back with the same process turned around backwards, a clockwork turntable, the needle and soundbox tracked the grooves and funneled the vibrations out through a horn which amplified the sound. The composition and configuration of the soundbox and diaphragm and the shape of the horn and the cutting lathe were all designed to complement each other. Very primitive technology, but very direct- vibrations being made into a physical record then played back the same way. The effect is startlingly present when you hear a really good recording and phonograph.

Here is what a recording session looked like. The horn had a limited range. Everything over ten feet away would fade into nothing. So the band had to crowd around the horn. With an orchestra, they would put musicians on swings suspended from the ceiling so they could get everyone in close enough.







To the left of this photo was the booth with the cutting lathe in it. Note the violin on the right with the horn attached to it. That's called a Stroh Violin.

I used to take a suitcase phono out to the patio of a local Starbucks and play records on weekends. High school kids would come up to me amazed. They couldn't believe there was no power plug or batteries. A lot of them had never even seen a phonograph record before, much less hear one played.

There are aspects to acoustic reproduction that could inform loudspeaker systems. They really understood acoustics and how a room affects the sound of music. Phonograph dealers would suggest putting a phonograph in the corner of a room so the walls and floors would act as extensions to the horn. This lowered bass response and increased volume. The horn made sound extremely directional which projected an aural image of the musician a few feet in front of the machine. It's a really creepy effect. They also recorded dry with no room ambience because they expected the room you were playing the record in to add its own reflections to the sound. Quite different from the way we design sound equipment now.

Once electrical recording with microphones was developed in the early 20s, manufacturers continued to optimize their records for their own brand of machine. Each company had its own compensation curve. A Victor might require a different playback curve than a Columbia. However, it wasn't consistently applied. Something recorded in New York would sound different than something in Chicago. And curves might change from session to session even. Phonograph folks get really good at EQing by ear. There are preamps with presets, but they don't come close to hitting all the variations in playback curves.


----------



## pinnahertz

bigshot said:


> There would be no point to 24 bits if you used electrical transcription. But if you were miking an acoustic phonograph, you might actually need it. As I mentioned before, the machine itself expands the dynamics in the recording. My Victrola produces ear splitting volumes with Caruso records with a very low noise floor. I think that has to do with the way the sound box and mica diaphragm react to energy in certain frequency ranges. In Caruso's high Cs it amplifies it by ringing in a very loud and natural sounding way. In the higher frequencies where surface noise occurs, it attenuates it. The horn also amplifies and has particular resonances. These things effectively expand the dynamic range beyond the level encoded in the grooves of the record.


There is no possible way you need 144dB theoretical, 120dB practical, to capture a acoustic recording no matter how its played.  Do we now have to circle back around and talk about room noise, HVAC, mic self noise, none of which comes even close to requiring 24bits,  and then go 40-50dB above all of that to look at shellac surface noise?  Really? 

Got your trusty SPL meter handy?  Set it to "fast", and check how loud Caruso's C really is, then measure the surface noise in the lead-in/lead-out, and let's find out if you even need 16 bits.


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## bigshot (Jun 22, 2017)

I suppose if you set your levels carefully before recording, you could easily fit it into 16 bit, but the loud peaks in phonograph records sometimes come out of nowhere. Since it was all acoustic, it was a lot more tolerant of overdriving peaks than electrical recordings. I've recorded my own phonograph several times where a peak would suddenly overdrive and turn to digital mush. I've learned to set my levels quite low to allow for it. Different records are cut at different volume levels too. Deeper wider grooves are louder than shallower narrower grooves. There was no standard. It's not like setting a level for a live musician in a studio. The phonograph can be very very loud and very very quiet depending on the record you play. There's no volume control on a phonograph. You swap different gauges of needles for different volume levels... loud tone needles, soft tone needles, spearpoints, medium tone... It all depends. The records themselves have a very narrow dynamic range, but played back acoustically, there can be a huge variation in level.

If I had to guess, I would bet that Caruso's high C would be somewhere above 120 dB with a loud tone needle. It makes my ears ring and my eyes wince sitting on the couch on the other side of the room. Close miked it would be very big. A very quiet surface with a soft tone needle would probably be around 40dB. Surface noise isn't really a big issue with acoustic playback, just with electrical transcription.


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## pinnahertz

bigshot said:


> I suppose if you set your levels carefully before recording, you could easily fit it into 16 bit, but the loud peaks in phonograph records sometimes come out of nowhere. Since it was all acoustic, it was a lot more tolerant of overdriving peaks than electrical recordings. I've recorded my own phonograph several times where a peak would suddenly overdrive and turn to digital mush. I've learned to set my levels quite low to allow for it. Different records are cut at different volume levels too. Deeper wider grooves are louder than shallower narrower grooves. There was no standard. It's not like setting a level for a live musician in a studio. The phonograph can be very very loud and very very quiet depending on the record you play. There's no volume control on a phonograph. You swap different gauges of needles for different volume levels... loud tone needles, soft tone needles, spearpoints, medium tone... It all depends. The records themselves have a very narrow dynamic range, but played back acoustically, there can be a huge variation in level.
> 
> If I had to guess, I would bet that Caruso's high C would be somewhere above 120 dB with a loud tone needle. It makes my ears ring and my eyes wince sitting on the couch on the other side of the room. Close miked it would be very big.


So, it's the resonances that get you?  

And I don't mean to misinterpret, but it sounds like you're saying that it's harder to set a recording level to record an acoustic recording (where you can adjust and do an identical Take 2) than it is to set a level for a live performance where the highest peak will not be known until it's gone forever?.


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## bigshot (Jun 22, 2017)

I think the thing that makes the Caruso peaks so big is the sheer power of his voice, the fact that the engineer cutting the wax master was expecting it and allowed for a very large groove, and the effect of the diaphragm ringing at a resonant frequency. The horn amplification would be pretty consistent across all frequencies, but there are certain frequencies that really blast. Record collectors who do electrical transcriptions call these "wolf tones" and they try to filter them out. But they're a big part of how the records were supposed to sound, and the acoustic playback system makes them sound very natural. It's not accurate, but it makes the limited abilities of the acoustic recording process sound better than without them. I wish I knew more about their theories, but all these people are dead now and a lot of their research was proprietary. It might not exist any more. Electrical recording is better documented because that was developed at Bell Labs.

Usually when I pull out my phonograph to record it, I set up the mikes, start rolling and then play a stack of records one after another. I could set levels individually for each record by monitoring it through one full playback, but that would take three times as long. Since the phonograph requires so much attention, winding and changing needles with each record side played, it's a lot easier to record with a one size fits all setting.


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## Cutestudio

gregorio said:


> 1. Because you are NOT looking at a waveform, you are looking at a graphical representation of the digital data. Zoom in to that representation and you'll see the discrete quantisation values, the "stair step" image BUT that's just the encoded digital data not the decoded data that come out of the DAC. Once decoded, there is no "stair step", a continuous analogue waveform!


You're not telling me anything new here, I'm a bit puzzled at the point you are trying to make. All you are saying is digital is digital and analog is analog, didn't we realise that 30 years ago? Welcome to 2017 BTW.



gregorio said:


> 2. It's not a nice or nasty noise, it's white noise and, it's way below the noise floor of the recording/reproduction with 16bit.


It must be a PWM noise because the points between don't exist, whereas they do exist in analog white noise.
Have a listen to some quiet 8bit dithered music, it's a much rougher sounding hiss than analog white noise.
As for 16bits being 'way below the noise floor', can you perhaps explain why people can hear the difference between different dither methods? If 16 bits is good enough, shouldn't the shape of the dither noise be irrelevant?



gregorio said:


> The result of dithering is a perfect, error free signal plus some benign white noise!


Oh dear, a triumph of belief over reality.
Dither is simply a way to disperse quantisation errors into the digital stream in the long term. You seem to forget that the quantisation _still exists_ and is _still distortion: _
For low/mid frequencies and continuous tones of course dither works, we all know that, but it simply _can't correct the quantisation errors of short transient events_. 
Go look at a real waveform of a click or something quiet with some bite. Zoom in and have a look at your quantisation steps for the quiet sharp transients, note that compared to the analog some of these vertical points are - by definition - _in the wrong place, creating the wrong shape_.
Your dither doesn't correct them because there is no way it possibly can.

Remember: low/mid frequencies and continuous (periodic) tones. Have you got it yet?
No one has ever claimed it works for short transients except perhaps you, and you're simply wrong, sorry.

As you seem completely oblivious to what dither is perhaps some more study for you is required?
https://en.wikipedia.org/wiki/Dither#Usage


			
				Everyone Else said:
			
		

> Dither should be added to any low-amplitude or highly periodic signal before any quantization or re-quantization process, in order to de-correlate the quantization noise from the input signal and to prevent non-linear behavior (distortion).


There are many other such references, try Google, knock yourself out. They all say the same.



gregorio said:


> 24bit is very useful, just not for a consumer music distribution format.


There is no reason why it should not be a consumer format, in fact it is already common on many DVD videos, perhaps you didn't know?
Additionally 16 bit with a shaped dither is technically unsuitable for further digital processing, so that counts out digital room EQ and full digital systems with digital level and digital crossover filtering, which would be far better done with 24bit or at least TPDF dither.

Additionally there is more than one study that shows people can tell the difference between 16 and 24bit on a consistent basis which again disproves your invalid assertion that 16bit is perfect.
http://journal.frontiersin.org/article/10.3389/fpsyg.2017.00093/full
http://www.aes.org/tmpFiles/elib/20170620/18296.pdf


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## bigshot

If you understand how human hearing works, and you understand the difference between 16 and 24 bit, you know exactly what difference it makes. For a recording studio, it's useful. For listening to music in your home, it's not needed. But it does serve as great sales pitch to people who falsely believe it makes a difference, so they tout it in ad copy. Because they're selling it to you, it doesn't mean you really need it.


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## WoodyLuvr

Cutestudio said:


> Additionally there is more than one study that shows people can tell the difference between 16 and 24bit on a consistent basis which again disproves your invalid assertion that 16bit is perfect.
> http://journal.frontiersin.org/article/10.3389/fpsyg.2017.00093/full
> http://www.aes.org/tmpFiles/elib/20170620/18296.pdf


How is it physically possible that any human ear could differentiate between 16-bits (SNR 96.33 db) and 24-bits (SNR 144.49 dB)... both of these bit depths offer a dynamic range well beyond human hearing detection capability?  The frequency response is exactly the same.  Really isn't the increase in bit depth *only lowering the noise level (noise floor)* but from a point that is already well beyond human hearing/detection?  Even though 24-bit is providing a theoretical ~40-50% lower noise floor can human ears actually detect that, if so how?

By the way good sir, the first test you linked doesn't exactly state nor support your argument (please help me understand how this study relates???) and the second link you provided is not working.

Here is an interesting study directly related to 16-bit vs 24-bit:  http://archimago.blogspot.com/2014/06/24-bit-vs-16-bit-audio-test-part-ii.html


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## castleofargh

@Cutestudio seems to only see his own idea of audio signals where sound1 + sound2 = sound1 ruined with errors from sound2's amplitude added to the original signal. sound1 is the album in 24bit, sound2 is 16bit dither. oh look how horrible it is, all the amplitudes have been changed a little, those are errors as the samples aren't exactly where they were before, so the music is ruined. 
and that's of course one way to look at things. not even a very wrong one IMO. but then let's take that same way to look at sound, what happens when 2 tracks are mixed? if an extra noise down at 16bit is bad enough to justify using 24bit, what a nightmare it must be to have 2 or 3 tracks mixed together, what about 5 or 10? look at all those errors and they're not down at -90dB, they're super high in amplitude. it's like completely rewriting the signal, OMG the instrument on the first track must sound horrible now. ^_^

some representations of signal are convenient and can of course tell us a lot of things, but we mustn't mistake a representation for the actual signal. we're dealing with waves, not with dots on a graph.


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## StanD

WoodyLuvr said:


> Here is an interesting study directly related to 16-bit vs 24-bit:  http://archimago.blogspot.com/2014/06/24-bit-vs-16-bit-audio-test-part-ii.html


Good read, thanks.


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## sonitus mirus

WoodyLuvr said:


> How is it physically possible that any human ear could differentiate between 16-bits (SNR 96.33 db) and 24-bits (SNR 144.49 dB)... both of these bit depths offer a dynamic range well beyond human hearing detection capability?  The frequency response is exactly the same.  Really isn't the increase in bit depth *only lowering the noise level (noise floor)* but from a point that is already well beyond human hearing/detection?  Even though 24-bit is providing a theoretical ~40-50% lower noise floor can human ears actually detect that, if so how?
> 
> By the way good sir, the first test you linked doesn't exactly state nor support your argument (please help me understand how this study relates???) and the second link you provided is not working.
> 
> Here is an interesting study directly related to 16-bit vs 24-bit:  http://archimago.blogspot.com/2014/06/24-bit-vs-16-bit-audio-test-part-ii.html



The first link does mention the word "inaudible" in the title and several more times throughout the study.  That indicates to me that the tester is not expected to hear anything that would cause this relaxed state.  Can any of this "hypersonic" nonsense be consistently repeated?  It seems like we are a long, long away from being able to confidently claim that ultrasounds are having any real affect on the listener, positively or negatively.


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## WoodyLuvr

sonitus mirus said:


> The first link does mention the word "inaudible" in the title and several more times throughout the study.  That indicates to me that the tester is not expected to hear anything that would cause this relaxed state.  Can any of this "hypersonic" nonsense be consistently repeated?  It seems like we are a long, long away from being able to confidently claim that ultrasounds are having any real affect on the listener, positively or negatively.


Concur, this along with this:

"... Our findings have some limitations. 
      First, because we used only a visual vigilance task, it is unclear whether high-resolution audio can improve performance on tasks that involve working memory and long-term memory... "
"... Second, the underlying mechanism of how inaudible high-frequency components affect EEG activities cannot be revealed by the current data... "
"... Fourth, the present study did not manipulate the sampling frequency and the bit depth of digital audio... " High-resolution audio is characterized not only by the capability of reproducing inaudible high-frequency components but also by more accurate sampling and quantization (i.e., a higher sampling frequency and a greater bit depth) as compared with low-resolution audio... "​


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## briskly

WoodyLuvr said:


> By the way good sir, the first test you linked doesn't exactly state nor support your argument (please help me understand how this study relates???) and the second link you provided is not working.


Second link is of the hi-res meta-analysis study that made the rounds in the hi-fi press last year. In case this sub-forum hasn't seen it enough already: http://www.aes.org/e-lib/browse.cfm?elib=18296
Not particularly familiar with the methods and limitations of meta-analysis. I do know that some of the sourced studies have particular caveats that make them poorly applicable to more real-world systems, and some have gross errors.


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## WoodyLuvr

briskly said:


> Second link is of the hi-res meta-analysis study that made the rounds in the hi-fi press last year. In case this sub-forum hasn't seen it enough already: http://www.aes.org/e-lib/browse.cfm?elib=18296
> Not particularly familiar with the methods and limitations of meta-analysis. I do know that some of the sourced studies have particular caveats that make them poorly applicable to more real-world systems, and some have gross errors.


Thank you for the link (that worked)... wouldn't this "extensive training" be interpreted by some as a "forced bias"?

"... Results showed a small but statistically significant ability of test subjects to discriminate high resolution content, and this effect increased dramatically when test subjects received extensive training... "​


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## Arpiben (Jun 23, 2017)

@Cutestudio
I will suggest you to read "The Theory of Dithered Quantization" from R.A.Wannamaker (2003).
Since involved maths may be unfamiliar &/or complex you may read the conclusions of the main chapters.
Basically for audio applications:

NSD Non Substractive Dither is prefered
TPDF Triangle Probability Density Function ( Triangle shape dither) is prefered since better fullfilling error independency vs input conditions
Dither results in analog domain can be extended to digital domain: digital dither can be
used to feed a digital-to-analogue converter for analogue dithering applications
This should bring answers at least for your wonders regarding 'digital white noise vs analog' as well as differences between dither shapes.

Like others I did not understand your concerns regarding transients ( Nyquist ones), dither smoothing or averaging issues.
Hope you are not confusing with other dither applications, in non linear systems,for example,where white noise is used for stabilizing purposes....

Anyhow I much prefer @gregorio or @pinnahertz explanations vs some Wannamaker or Lip shitz papers 
I also admire their patience/fight against misconceptions. Respect.


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## bigshot

sonitus mirus said:


> It seems like we are a long, long away from being able to confidently claim that ultrasounds are having any real affect on the listener, positively or negatively.



Much less improve the perceived sound quality of music! If you can't even perceive it, how is it going to make Mozart sound better in your living room?


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## StanD

sonitus mirus said:


> It seems like we are a long, long away from being able to confidently claim that ultrasounds are having any real affect on the listener, positively or negatively.


Too much of an ultrasonic power level and you have a weapon. I guess that's a negative.


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## Don Hills

StanD said:


> Too much of an ultrasonic power level and you have a weapon. I guess that's a negative.



How about 145 to 155 dB SPL at 60 KHz, as proposed for a wireless cellphone charging system?


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## StanD

Don Hills said:


> How about 145 to 155 dB SPL at 60 KHz, as proposed for a wireless cellphone charging system?



You could kill every bat for miles with that.


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## castleofargh

Don Hills said:


> How about 145 to 155 dB SPL at 60 KHz, as proposed for a wireless cellphone charging system?



I instantly went to buy 2 "I only give negative feedback" shirts online. it's like finding my spirit animal.


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## gregorio

Cutestudio said:


> [1] Oh dear, a triumph of belief over reality.
> [2] There are many other such references, try Google, knock yourself out. They all say the same.



1. That's just funny. Such a typical audiophile response, accuse others of exactly what you're guilty of!

2. Yes, they do all say the same. How on earth can you quote something you've obviously failed to read or understand? Let's look at that quote and read to the end of it:
"_Dither should be added to any low-amplitude or highly periodic signal before any quantization or re-quantization process, in order to de-correlate the quantization noise from the input signal and *to prevent non-linear behavior* (distortion)._" - It prevents non-linear behavior, which means the system is linear with dither and there is *NO* error/distortion. Your transients and everything else is L-I-N-E-A-R with dither! Instead of telling everyone else they don't know what dither is and to go and look at graphical representation of digital data, take your own advice and go and learn something. Instead of arguing from a position of ignorance and being rude to those who aren't so ignorant, why don't you ask if there's something you don't understand? That approach will serve you much better in this sub-forum.

G


----------



## Cutestudio

gregorio said:


> 2. Yes, they do all say the same. How on earth can you quote something you've obviously failed to read or understand? Let's look at that quote and read to the end of it:
> "_Dither should be added to any low-amplitude or highly periodic signal before any quantization or re-quantization process, in order to de-correlate the quantization noise from the input signal and *to prevent non-linear behavior* (distortion)._" - It prevents non-linear behavior, which means the system is linear with dither and there is *NO* error/distortion.
> G



Oh dear, you really missed the point there, again.
You merely highlighted what dither does for long and periodic signals, which we all know already. Duh.

What you keep failing to acknowledge, either deliberately or via some form of denial or mental block is that dither doesn't fix short transient events, it merely reduces correlated quantisation artifacts over time in a statistical manner.  You have a basic but only partially formed knowledge of dither: a little knowledge can be a dangerous thing'!

You are assuming it magically converts a low bit depth waveform into a higher bit depth waveform in all cases, but it's a statistical method and can therefore only do that over time. What do you think dither actually is? Think about the mechanism!! It's adding a sub-bit noise to a waveform before quantisation to a lower bit depth to statistically knock some values up to the next integer level, instead of the zero dither option that truncates them all flat. 

This statistical method can only work over time and that is why it can't fix the shape of individual wavelets, it merely de-correlates the quantization artifacts. Of course the artifacts are non-linear, what did you think was linear about truncating certain values??

I suggest you learn a little about digital signal processing, the dither technique is not magic, it's a trade off, like a PWM signal it's making sure enough hops between the quantised levels occur over time so the errors appear as random noise rather than correlated noise: it's a trade, not a cure, and in no way makes the waveform more 'accurate', it just hides the error in a pleasant noise.

The very facts that:

1) People can hear the effect of different dithers
2) Many people post process digital sound before it's played

tells us that 16bit is inadequate for many, even it you think it's perfect for your car. Even the dither is wrong for post processing, mathematically TPDF dither is best for accuracy of long and continuos waveforms but many CDs and downloads are mastered with a final shaped dither (because they sound better), which is not ideal for any digital post processing like room EQ etc.

I'm still not sure why you keep pushing 16bit, 24bit is clearly a better format - even if you can't tell the difference - and is widely used outside of audio (DVD videos for instance often use 24 bits), your insistence that we all remain listening to a format that was barely adequate 30 years ago shows that you are living in the past and need to embrace the technology of today, not the 'good enough for me' 16bit format. 

We don't all listen to church organ recitals.


----------



## WoodyLuvr (Jun 25, 2017)

Cutestudio said:


> I'm still not sure why you keep pushing 16bit, 24bit is clearly a better format - even if you can't tell the difference - and is widely used outside of audio (DVD videos for instance often use 24 bits), your insistence that we all remain listening to a format that was barely adequate 30 years ago shows that you are living in the past and need to embrace the technology of today, not the 'good enough for me' 16bit format.


Barely adequate thirty (30) years ago?!  So does that mean you believe CDs were already inferior when consumers were just beginning to afford them in the late-80s?



Cutestudio said:


> We don't all listen to church organ recitals.


We also don't need flippant remarks that have absolutely no bearing on the conversation at hand.  It's uncalled for and unneeded and comes across childish and pity.  Why not just simply argue your point well and politely?  Let's all be nice.


----------



## gregorio

Cutestudio said:


> [1] Oh dear, you really missed the point there, again. You merely highlighted what dither does for long and periodic signals, which we all know already. Duh.
> [2] You have a basic but only partially formed knowledge of dither: a little knowledge can be a dangerous thing'!
> [3] I suggest you learn a little about digital signal processing ... [3a] it's a trade, not a cure, and in no way makes the waveform more 'accurate' ...
> [4] The very facts that:
> ...



1. What point, the point you've just invented to support your ridiculous argument? YOU are the one who quoted that dither "_prevents non-linear behavior_", where's the addendum or caveat to that quote which states: "except in the case of transients"? There isn't one, you've just made it up! Dither prevents non-linear distortion period, nothing excluded, Duh!

2. Not nearly as dangerous as a delusion instead of ANY knowledge!

3. It's way past time to take your own advice. Some 101 basics will do you for the time being! You really, really need to learn what dither is and to give you a starting clue, it's NOT just masking quantisation error in noise! 
3a. You do know what "linear" means? As dither prevents non-linear behavior, then with dither we have linear behavior and therefore the waveform is perfectly accurate, so duh, dither obviously makes the waveform more accurate. Unless you're saying the quote you posted and all the countless similar ones you mentioned are all wrong? Also, you do realise that a transient is also a waveform that occurs over time?

4.1. Of course you can hear it, just apply dither to a low level signal, increase the gain by 70dB or so and it's easy to hear.
4.2. So? ... What are you doing with your EQ'ing, boosting 17kHz by 50dB? And yes, TDPF is best for audio waveforms, hurah, I think you're getting it. Now all you need to get is that a transient is also an audio waveform, duh! You do realise that all sound occurs over time, duh?

5. Right, you can't tell a difference but it's better anyway? Anyone else for a big dose of audiophile Koolaid? Secondly, as you've been told, DVD videos typically have a per channel bit rate which is about a tenth of the bit rate for 16/44.1 (as indeed does HDTV)! Thirdly, how on earth can you describe having 40 - 100 times more dynamic range than ever required as "barely adequate"? You keep avoiding this question, which leads to the conclusion that you recognise this huge logical hole in your argument and therefore refuse to address it, presumably to suit your agenda which, I can only assume, must be either; trolling, shill'ing or a fierce determination to maintain your audiophile myth/delusion? 

G


----------



## pinnahertz

Cutestudio said:


> I'm still not sure why you keep pushing 16bit, 24bit is clearly a better format - even if you can't tell the difference - and is widely used outside of audio (DVD videos for instance often use 24 bits), your insistence that we all remain listening to a format that was barely adequate 30 years ago shows that you are living in the past and need to embrace the technology of today, not the 'good enough for me' 16bit format.


Sooo many problems here.  I'm just picking this one, because...well, frankly, I'm exhausted. 

DVD videos and 24 bit audio??  Nope.  Yes, the capability is there, but the disk only holds 8 gig, and if you did a 2 hr movie in 24 bit 6 channel PCM, that's over 6 gig.  So, no, they don't. 

Blu-ray did someone say? Yes, there is 24 bit audio on BD.  Considering lossless PCM only, it's about 10% of all releases.  Considering all codecs, even the lossy ones, we bump up to 22%.  So, yes, it's there, not it's not all, or most.  All and most is 16 bits. (statistics taken from blu-raystats.com)

Why do you think that is?  Do you think it's because it's a lousy outdated audio format "barely adequate 30 years ago"?  Or is it fully adequate to convey the creative intentions, and they're using the bandwidth for other content (like multi-language/multi-channel tracks, extras, etc.), especially so in multichannel audio? 

One more thing, I take exception to your "barely adequate 30 years ago" statement.  Some of us were actually around then, working in pro audio, and even for a decade or so before that.  Adequate?  Lemme tell you.  Analog...anything...was not really adequate.  We had to jump it through hoops to get it to be acceptable.  Tape running fast, add-on noise reduction, re-alignment of the recorder with every tape batch....no, it really kinda sucked, but that's all we had so it was "adequate"...in really big quotes.  Then along comes 16 bits.  You cannot even imagine the improvement in every single aspect of audio storage and reproduction.  It's not a small difference, it' was huge.  We went from adequate (because it's all we had) analog, to blowing it's doors off with humble 16 bit digital. 

I don't call that "barely adequate 30 years ago".  And it's not "barely adequate" now either.  

Your world is very different, clearly.


----------



## Cutestudio

WoodyLuvr said:


> We also don't need flippant remarks that have absolutely no bearing on the conversation at hand.  It's uncalled for and unneeded and comes across childish and pity.  Why not just simply argue your point well and politely?  Let's all be nice.



Now now, that was an example of music that responds well to dither, did you actually read my reply?



gregorio said:


> 1. What point, the point you've just invented to support your ridiculous argument? YOU are the one who quoted that dither "_prevents non-linear behavior_", where's the addendum or caveat to that quote which states: "except in the case of transients"? There isn't one, you've just made it up! Dither prevents non-linear distortion period, nothing excluded, Duh!



I'm sensing you're all riled up about this for some reason and didn't actually read my post properly.
Perhaps you should have a think about the mechanism, dither can only span those quantisation gaps over time, time a transient may not have.
You appear to be disputing a mathematical fact.



gregorio said:


> 3a. As dither prevents non-linear behavior, then with dither we have linear behavior and therefore the waveform is perfectly accurate.


You see this is where I'm disagreeing with you, dither can only work over a number of samples because it's a statistical method.

I suspect the problem is that you haven't reviewed many digital waveforms close in and seen how much each point counts. Take the old Genesis 'Lamb Lies Down' album, very dynamic, plenty of very quiet parts where dither is really important for 16bit due to the inherent lack of resolution. The 'perfectly accurate' you speak of is a statistical measure than cannot apply to the exact shape of one-off transient events, only in general will the waveform be considered accurate. E.g. in a periodic waveform you can look at a dithered digital cycles, superimpose all the cycles and averaged together they are indeed accurate, but individually they still have to obey the quantising levels. If therefore you only have one of them you have no idea of knowing the true shape.



gregorio said:


> And yes, TDPF is best for audio waveforms, hurah,


Well no, again you simplify, mastering doesn't just use TPDF, shaped dither is a popular way to get a better sound from the CD format: your declaration that is 'is best' disagrees with all those mastering engineers who use a shaped dither.
He's some useful posts discussing that.
https://www.gearslutz.com/board/mastering-forum/434950-noise-shaping-dither.html



gregorio said:


> 5. Right, you can't tell a difference but it's better anyway?


Who can't tell the difference?

http://journal.frontiersin.org/article/10.3389/fpsyg.2017.00093/full
http://www.aes.org/e-lib/browse.cfm?elib=18296




gregorio said:


> how on earth can you describe having 40 - 100 times more dynamic range than ever required as "barely adequate"? G


How on earth? Less hysteria, more thought please.
What makes you think 96dB covers all possible cases? I know you think it's more with dither, and for many music styles it is, especially church organ music, but as explained above it's rather debatable if dither helps short transient events.

How can you re-create a live concert sound in your house or headphones with 96dB? Or even the sound of a well hit snare drum? It seems to me you are arguing for Mid-Fi, not Hi-Fi. There are many obstacles to Hi-Fi, the ability to go to 24bit is not one of them, except perhaps here in this thread for a reason I have yet to fathom.

You also keep ignoring the fact that some people post process their digital music as which point 24bit would be rather more useful: mastering sometimes doesn't stop on release of the digital file.
And you keep refusing to explain your reasons for not liking 24bit as a 'consumer' medium, what's the deal with that? 'Because you don't think other people need it' wasn't registering here as a reason BTW.



pinnahertz said:


> Sooo many problems here.  I'm just picking this one, because...well, frankly, I'm exhausted.
> 
> DVD videos and 24 bit audio??  Nope.  Yes, the capability is there, but the disk only holds 8 gig, and if you did a 2 hr movie in 24 bit 6 channel PCM, that's over 6 gig.  So, no, they don't.


Your generalisation from a single example is noted.
Yes, the capability is there would have been sufficient.
Outside the HiFi box 24bit and high sample rates are common place, as they are in any pro-audio gear (obviously), which is often IME cheaper and better than the niche 'audio' gear.

I bought a cheap USB to S/PDIF converter the other day with coax and optical out for a project, you know what, even that goes to 24bit / 96kHz.
Yet here we are, double teaming the heretic who suggests 24bit has benefit over 16bit. Doh!


----------



## Whazzzup

I can tell the difference, matter of fact, I'm a 32 bit man  384 or 786 fr to boot.


----------



## WoodyLuvr

@Cutestudio your antics and tone are too silly and childish for my tastes and time; too bad you couldn't behave and be civil as you did have some interesting points and arguments... but alas you are now set on "IGNORE".  Goodbye.


----------



## Guidostrunk

Never run from a debate! Regardless!

Anyway....... I've never really paid attention to bit rates and whatnot. FWIW, I've downloaded stuff from HDtracks("lossless") , in the past, and have been a Spotify premium account holder for 3 years now. Listening to the same song , with my back turned to my laptop , and my wife randomly switching between foobar, and Spotify. I could not accurately tell which one was lossless , or Spotify. Almost every time I guessed , my wife said nope, you're wrong. 
And yes , I was literally guessing because I could hear no discernible difference whatsoever. 
Thank God,  because I haven't downloaded anything from there since, which in turn, saved me some cash. 

Who knows. Maybe I'm not a "Critical listener", or have "Golden ears". 
I'm content with that though.


----------



## headdict

Did you by any chance use tracks consisting of more than one sample for your comparisons? That would perfectly explain why you were not able to discern HD from the lossy format. Sorry to tell you you did it all wrong.


----------



## WoodyLuvr

Guidostrunk said:


> Never run from a debate! Regardless!
> 
> Anyway....... I've never really paid attention to bit rates and whatnot. FWIW, I've downloaded stuff from HDtracks("lossless") , in the past, and have been a Spotify premium account holder for 3 years now. Listening to the same song , with my back turned to my laptop , and my wife randomly switching between foobar, and Spotify. I could not accurately tell which one was lossless , or Spotify. Almost every time I guessed , my wife said nope, you're wrong.
> And yes , I was literally guessing because I could hear no discernible difference whatsoever.
> ...


    The silliness reached a level that simply was just giving me a big headache and nothing else.  I hear you on saving money.  I too went through a FLAC/ALAC phase a few years back... what a waste of time, money, and memory!  Cheers.


----------



## Guidostrunk

So, how exactly is the right way? Lol. And to answer your question. Yes it was multiple songs. 

Tell me what the right way is, and I'll give it a whirl.


headdict said:


> Did you by any chance use tracks consisting of more than one sample for your comparisons? That would perfectly explain why you were not able to discern HD from the lossy format. Sorry to tell you you did it all wrong.


----------



## Whazzzup

hey lay off my lossless


----------



## Guidostrunk

Lol. I laid off my lossless. Yours is fine.


----------



## headdict

Guidostrunk said:


> So, how exactly is the right way? Lol. And to answer your question. Yes it was multiple songs.
> 
> Tell me what the right way is, and I'll give it a whirl.


According to what we just learned on this thread you have to listen to single samples in order to hear the dither's ineffectiveness with lossy formats (or any 16 bit format actually). With music that lasts longer than a single sample, you won't hear it.
Lossless is for losers, by the way.


----------



## castleofargh

Cutestudio said:


> Oh dear, you really missed the point there, again.
> You merely highlighted what dither does for long and periodic signals, which we all know already. Duh.
> 
> What you keep failing to acknowledge, either deliberately or via some form of denial or mental block is that dither doesn't fix short transient events, it merely reduces correlated quantisation artifacts over time in a statistical manner.  You have a basic but only partially formed knowledge of dither: a little knowledge can be a dangerous thing'!
> ...


nobody is forcing you to use 16bit. the very guys telling you how 16bit is audibly enough for playback are or have been using way more bits on a daily basis for professional reasons. hell, I output my 16bit files to 24bit because my DAC is slightly better that way and I feel less guilty for using the digital volume in foobar. you need 24bit, you want 24bit, use 24bit. who cares? we're not from the bit police. 
you think albums need 24bit encoding? we have too many failed blind tests on different scales that suggest the opposite for playback. unlike the links you provide that consistently focus on sample rate and ultrasounds yet somehow seem relevant to you. why? or you could simply try like I suggested you do some times back, to use dither of different types at different bit depth and abx the files or at least try to notice the noise floor. you notice it, you go down a little. you notice it less, you keep going down. and then you don't notice it anymore, you look at the bit depth and it's something like 12 or 13bit maybe 14bit with quiet tracks and the volume level real loud. if you reach 18 or 20bits as your threshold, please let us know how much you had to cheat to get that result.  

but whatever your problem is, it's not like anybody claimed that 16bit has more resolution than 24. so what's the big deal? 




reading this threat the only thing I notice is that you fail to integrate any new information provided to you.
-the transient thing, nobody has any idea what the hell you're talking about. just in case, the time axis is the other one on your graph. 
-the DVD movies... someone pointed it out last time already, you don't care, you're happy to repeat your mistakes using the one example that doesn't work. 
-your understanding of dither... I don't know where to start so I wont. I'm scared of the moment when you will realize that band limiting also changes the shape of the signal in audacity. will you then say that Nyquist is wrong and buy a NOS DAC with no filter? 
all in all it's a little discouraging.




as to why we're ok with 16 when we could have 24bit? I can only answer for myself. I'm happy that way for the same reason I have "only" a 300dpi printer and "only" a 22 megapixel camera. because those are numbers which already reach the threshold of my senses for the typical use I make of those tools. maybe I lack that little alpha male thing where everything must turn into a competition? maybe I just know that better does exist but what really make something better aren't those specific variables?  probably both.


----------



## Guidostrunk

First. Who ever said anything about samples of songs? Certainly not me. My comparisons were of full songs, and multiple songs. 
Secondly. I wouldn't call anyone a loser, based on their preference in audio format. 

Definitely someone who is down right ignorant , deserves loser status 


headdict said:


> According to what we just learned on this thread you have to listen to single samples in order to hear the dither's ineffectiveness with lossy formats (or any 16 bit format actually). With music that lasts longer than a single sample, you won't hear it.
> Lossless is for losers, by the way.


----------



## headdict

Guidostrunk said:


> First. Who ever said anything about samples of songs? Certainly not me. My comparisons were of full songs, and multiple songs.
> Secondly. I wouldn't call anyone a loser, based on their preference in audio format.
> 
> Definitely someone who is down right ignorant , deserves loser status


LOL. You were obviously not following the recent discussion on how dither works and I was just trying to apply what I have learned from it. Everything I wrote is of course utter nonsense.


----------



## Guidostrunk

Fair enough. I know nothing about dither, so I'm out of that discussion lol.


headdict said:


> LOL. You were obviously not following the recent discussion on how dither works and I was just trying to apply what I have learned from it. Everything I wrote is of course utter nonsense.


----------



## pinnahertz

Cutestudio said:


> Your generalisation from a single example is noted.
> Yes, the capability is there would have been sufficient.


It was YOUR generalization, and was incorrect.  The capability is there, but is _unusable_ for video DVDs!


Cutestudio said:


> Outside the HiFi box 24bit and high sample rates are common place, as they are in any pro-audio gear (obviously), which is often IME cheaper and better than the niche 'audio' gear.


You have a odd (means: wrong) definition of commonplace.  Higher resolution in production is not what this discussion is about.  It's about the release format into the hands of consumers. 


Cutestudio said:


> I bought a cheap USB to S/PDIF converter the other day with coax and optical out for a project, you know what, even that goes to 24bit / 96kHz.
> Yet here we are, double teaming the heretic who suggests 24bit has benefit over 16bit. Doh!


It's because the heretic doesn't seem to understand application of technology.  The fact that 24 bit consumer gear is available doesn't change the fact that it's benefits, if any, are inaudible to the user, are a strong marketing influence only, and simply occupy extra bandwidth and storage space.  And it's mostly not available to the consumer anyway.  Yes, we work in 24 bits in production.  Often at slightly higher bit rates too, but that doesn't mean it makes any sense to release that to the consumer, that he would want it, or could actually even play it.  You do know what devices most music is played on, right?  Or are we in our own world there too?


----------



## bigshot

The irony of this whole debate is that 16 bit audio even without dithering is for all intents and purposes good enough for listening to music in your living room. Dithering just gilds the lilly and makes the noise floor a little bit lower. I fail to see what kind of transients you're going to be eliminating at -70dB. It doesn't affect a 20kHz waveform either. This argument is pure theory stretched to absurd levels. No practical effect at all.


----------



## pinnahertz

bigshot said:


> The irony of this whole debate is that 16 bit audio even without dithering is for all intents and purposes good enough for listening to music in your living room. Dithering just gilds the lilly and makes the noise floor a little bit lower. I fail to see what kind of transients you're going to be eliminating at -70dB. It doesn't affect a 20kHz waveform either. This argument is pure theory stretched to absurd levels. No practical effect at all.


Yes!  

Small point perhaps but dithering raises the noise floor, not lowers.  But doesn't affect transients or 20kHz, and this has long since passed beyond theory, and even more importantly, passed beyond the realm of practical application of theory.   In most living rooms you can't tell if ti's 16, 16 dithered (which it likely is), or 24 (which it can't be other than the structure of the data itself).  

I think in honor of the death of logic and reason in this thread, you should play a good, loud Caruso acoustic recording for us on the mechanical/acoustic player (with it's own special kind of dither).  Point the horn east, perhaps I'll hear it.


----------



## old tech (Jun 25, 2017)

bigshot said:


> The irony of this whole debate is that 16 bit audio even without dithering is for all intents and purposes good enough for listening to music in your living room. *Dithering just gilds the lilly and makes the noise floor a little bit lower.* I fail to see what kind of transients you're going to be eliminating at -70dB. It doesn't affect a 20kHz waveform either. This argument is pure theory stretched to absurd levels. No practical effect at all.


I'm far from an expert in this area and this statement makes me question what I've learned to date.  I thought that dither raises the noise floor - ie correlating quantisation errors into random white noise?

I agree though that the noise floor of dithered 16bits is for all intents and purposes inaudible.  The only question I have in my mind though is why dither at all at 16bits when the dithered noise floor is inaudible, which by definition would mean that the audibility of quantised errors from undithered 16bits quantised would be even lower?


----------



## Don Hills

Cutestudio said:


> ... You see this is where I'm disagreeing with you, dither can only work over a number of samples because it's a statistical method.
> 
> I suspect the problem is that you haven't reviewed many digital waveforms close in and seen how much each point counts. Take the old Genesis 'Lamb Lies Down' album, very dynamic, plenty of very quiet parts where dither is really important for 16bit due to the inherent lack of resolution. The 'perfectly accurate' you speak of is a statistical measure than cannot apply to the exact shape of one-off transient events, only in general will the waveform be considered accurate. E.g. in a periodic waveform you can look at a dithered digital cycles, superimpose all the cycles and averaged together they are indeed accurate, but individually they still have to obey the quantising levels. If therefore you only have one of them you have no idea of knowing the true shape. ...



How many samples does it take to capture the shortest transient that will fit within the Nyquist bandwidth?


----------



## Strangelove424

old tech said:


> I'm far from an expert in this area and this statement makes me question what I've learned to date.  I thought that dither raises the noise floor - ie correlating quantisation errors into random white noise?
> 
> I agree though that the noise floor of dithered 16bits is for all intents and purposes inaudible.  The only question I have in my mind though is why dither at all at 16bits when the dithered noise floor is inaudible, which by definition would mean that the audibility of quantised errors from undithered 16bits quantised would be even lower?



It just comes down to what sounds more pleasing if you were theoretically going to hear it. Undithered errors would take the form of distortions that correspond to the signal, which is generally thought to be displeasing when compared to the homogeneity of white noise. We tend to associate white noise with analogue and randomized distortion with digital artifacts, and dithering is often thought of as a way of making the noise floor of digital sound more like traditional analogue. It's not audible, but in theory that's why distortion is traded for noise.


----------



## pinnahertz

Strangelove424 said:


> It just comes down to what sounds more pleasing if you were theoretically going to hear it. Undithered errors would take the form of distortions that correspond to the signal, which is generally thought to be displeasing when compared to the homogeneity of white noise. We tend to associate white noise with analogue and randomized distortion with digital artifacts, and dithering is often thought of as a way of making the noise floor of digital sound more like traditional analogue. It's not audible, but in theory that's why distortion is traded for noise.


Except that, apart from the very most basic dither, it's not just simply white noise.  There are several different ways of shaping the noise to get it to work well as a dither signal, but be less audible as a noise signal.

If you want to jump in with both feet...*this is a pretty good read*, although obviously product specific, the principles are there.


----------



## Strangelove424

They got it right in the first line of the intro, it's the audio equivalent of transmission fluid. Not even that, because you actually need transmission fluid, more like washer fluid. I'm sure anything can be made better with more intelligent plugins, it's a matter of being worth it or not or whether it will ever be heard, like whether or not the fancy blue washer fluid is worth it to clean bird poo off, or if I'm better off with distilled water and some blue food coloring (just kidding, I wouldn't put food coloring in my washer lines). Because they were so honest at the onset, however, and the content is interesting, I will read it through.


----------



## bigshot

A lot of audiophila is like working out the ratio of a circle's circumference to its diameter to way more decimal places than you really need to slice a nice piece of apple pi for dessert.


----------



## Whazzzup

I liked the life of Pi


----------



## WoodyLuvr (Jun 27, 2017)

Whazzzup said:


> I liked the life of Pi


So does that mean the McIntosh MHA150 in our story is really just a Schiit Magni/Modi stack and the HE-1s are HD6XXs?!


----------



## gregorio (Jun 27, 2017)

Cutestudio said:


> [1] I'm sensing you're all riled up about this for some reason and didn't actually read my post properly.
> [2] Perhaps you should have a think about the mechanism, dither can only span those quantisation gaps over time, time a transient may not have.
> [3] You appear to be disputing a mathematical fact.
> [4] You see this is where I'm disagreeing with you, dither can only work over a number of samples because it's a statistical method.
> ...



1. Unfortunately, your "sensing" is backwards, it's you who haven't read your own posts properly! It was you who quoted that dither prevents non-linear behaviour and since then you've tried to argue that it doesn't?

2. Again, you've got it backwards and YOU need to think about the mechanism! I notice that you have yet again failed to answer the question and presumably you do not realise that transients, like every other sound, only exist over time. If the frequency of the transient is too high/fast (beyond the Nyquist point) then it cannot be captured and cannot be dithered. There is no transient or any other audio waveform which can be captured with just a single sample, at least two samples are required. If it can be captured, then it can be dithered.

3. Again, completely backwards. You posted the quote, which is the end result of the "mathematical fact" and you are the one now disputing that quote/mathematical fact!!

4. And that's precisely why you're stupid to disagree! Yes, dither does take a number of samples BUT so does encoding a waveform as digital data in the first place and so does reconstructing it back into an analogue waveform. A to D and D to A conversions are themselves statistical processes. This really is digital theory lesson #1, which you clearly do not know and is presumably why you're coming out with all this nonsense ... and you accuse others that a little knowledge is dangerous, sheesh!

5. It's really impressive how much you are able to get so utterly wrong with just one simple sentence!! Firstly, it demonstrates that you don't understand even the absolute basics of digital audio theory and secondly, your "suspected problem" couldn't be more ludicrous if you deliberately tried! 1. You are looking at digital data points, not a waveform! That digital data represents coordinates which will only become a waveform once processed by a sinc function, until it is, then of course it will contain errors, digital data is NOT an analogue of the waveform!! Come on, this is the basics of digital audio developed 90 years ago! 2. I've "reviewed" digital audio data "close in" pretty much every working day of my life for the last 20 years!

6. "Very dynamic" compared to what? Compared to many modern pop music releases sure, but compared to the dynamic range offered by 16bit, NO, it has a tiny dynamic range, about 100 times less than 16bit!!

7. This just gets better! Now you're telling a mastering engineer what mastering engineers do.

8. Duh, I'm quoting YOU! "_I'm still not sure why you keep pushing 16bit, 24bit is clearly a better format - *even if you can't tell the difference*_...". Are you not reading your own posts?

9. Less delusion and ignorant nonsense please and more logic and facts!!
9a. How many commercial recordings can you name with more than 60dB dynamic range? If nothing else *answer this question*!

10. Been to many live concerts or houses with a 0dBSPL noise floor have you?
10b. You normally sit with your ear an inch or two away from the snare drum during a live gig do you? The only situation I can imagine where you might have your ears that close to a snare drum during a performance is if you were giving the drummer a blowjob! In which case, some ear defenders would be essential and a moist towel would probably come in handy as well!

11. It seems to me that you are arguing for exactly the same "fi" but just want it to be bigger and cost more!

12. Explain why a 24bit distribution format would be more useful if people want to post process their digital audio.

G


----------



## StanD

WoodyLuvr said:


> So does that mean the McIntosh MHA150 in our story is really just a Schiit Magni/Modi stack and the HE-1s are HD6XXs?!


A simulation.


----------



## MorrisL

I still didn't understand why exactly 24bit is useful in recording and mixing music. I record and mix all the time and I've tried both 16bit and 24bit and have compared them carefully. These doesn't seem to be any difference here either, just like in playback.


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## bigshot

If you record at a decent level and don't do a lot of filters, there probably isn't any real difference.


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## gregorio

MorrisL said:


> I still didn't understand why exactly 24bit is useful in recording and mixing music.



In terms of recording, the advantage is purely in terms of headroom, not directly in terms of audio quality. With 24bit we don't have to be concerned with the noise floor of the recording medium, so if we set a level which results in a max peak of say -20dB it's not an issue, whereas that could be an issue with 16bit, after leveling and compression are applied during mixing and mastering. So with 16bit recording we tend to record hotter but then that risks unexpected clipping.

In terms of mixing, it makes no real difference at all, assuming an appropriate signal to noise recording mentioned above. However, the bit depth of the mixing environment (as opposed to the bit depth of the audio file itself) does make a difference and most mix environments today are 64bit float and even going back 20 years, it was typically 48bit fixed. This added bit depth in mixing is required to combat the issue of successive rounding errors generating noise, which would likely become audible in a 16bit mix environment, due to the sheer number of processing steps in a typical mix. With a 64bit float mix environment tens of thousands of processors would be required for the rounding errors to reach audible levels and it therefore becomes a non-issue.

G


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## MorrisL

gregorio said:


> In terms of recording, the advantage is purely in terms of headroom, not directly in terms of audio quality. With 24bit we don't have to be concerned with the noise floor of the recording medium, so if we set a level which results in a max peak of say -20dB it's not an issue, whereas that could be an issue with 16bit, after leveling and compression are applied during mixing and mastering. So with 16bit recording we tend to record hotter but then that risks unexpected clipping.
> 
> In terms of mixing, it makes no real difference at all, assuming an appropriate signal to noise recording mentioned above. However, the bit depth of the mixing environment (as opposed to the bit depth of the audio file itself) does make a difference and most mix environments today are 64bit float and even going back 20 years, it was typically 48bit fixed. This added bit depth in mixing is required to combat the issue of successive rounding errors generating noise, which would likely become audible in a 16bit mix environment, due to the sheer number of processing steps in a typical mix. With a 64bit float mix environment tens of thousands of processors would be required for the rounding errors to reach audible levels and it therefore becomes a non-issue.
> 
> G


OK, I get it now. Thanks. I always record at 24bit/48hz, and recently considered going back to 16bit/44.1 because I couldn't tell a difference and didn't see a point in filling up hard drive space unnecessarily. Might as well stick to 24bit then.


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## gregorio

MorrisL said:


> OK, I get it now. Thanks. I always record at 24bit/48hz, and recently considered going back to 16bit/44.1 because I couldn't tell a difference and didn't see a point in filling up hard drive space unnecessarily. Might as well stick to 24bit then.



If what you record has fairly predictable max peak levels, then you'd have nothing to gain from recording in 24bit, you'd just be filling up hard disk space unnecessarily. In commercial situations we're often facing wildly different max peak levels from client to client, so poor predictability and clipping a take is unacceptable, so 24bit is useful purely from the point of covering our a**es! For over a decade, commercial engineers only had 16bit recording and had to dedicate more of their time to covering their a**es by being more assiduous with their input levels to start with but then we generally have a great deal less time to record today than we did 2+ decades ago.

G


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## MorrisL

gregorio said:


> If what you record has fairly predictable max peak levels, then you'd have nothing to gain from recording in 24bit, you'd just be filling up hard disk space unnecessarily. In commercial situations we're often facing wildly different max peak levels from client to client, so poor predictability and clipping a take is unacceptable, so 24bit is useful purely from the point of covering our a**es! For over a decade, commercial engineers only had 16bit recording and had to dedicate more of their time to covering their a**es by being more assiduous with their input levels to start with but then we generally have a great deal less time to record today than we did 2+ decades ago.
> 
> G


I may be misunderstanding you but do you mean to say that if my recordings are ever so slightly clipping, I might be able to fix that if they're in 24bit, as opposed to 16bit?


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## castleofargh

nope that's not what he meant ^_^. just that if you have "only" 16bit for your tracks, you will spend more time trying to get the recording not too far from 0dB, from fear of potentially audible background noise. while with 24bit, you just don't have to care. you can leave a good deal of headroom to be absolutely sure that you won't clip anything and also don't have to worry too much about the background noise becoming audible when you add some gain later on those tracks to match them with others. it's just an easier/safer process.


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## MorrisL

castleofargh said:


> nope that's not what he meant ^_^. just that if you have "only" 16bit for your tracks, you will spend more time trying to get the recording not too far from 0dB, from fear of potentially audible background noise. while with 24bit, you just don't have to care. you can leave a good deal of headroom to be absolutely sure that you won't clip anything and also don't have to worry too much about the background noise becoming audible when you add some gain later on those tracks to match them with others. it's just an easier/safer process.


Thanks. I understand it better now. Basically, I can record "colder" than before and not worry about the possibility of clipping. This is definitely worthwhile. Can't tell you how often a perfectly good take was almost ruined because of a few unexpectedly loud hits here and there.


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## gregorio

MorrisL said:


> Thanks. I understand it better now. Basically, I can record "colder" than before and not worry about the possibility of clipping. This is definitely worthwhile. Can't tell you how often a perfectly good take was almost ruined because of a few unexpectedly loud hits here and there.



Yep, castleofargh's response was spot on. And yes, that's exactly the benefit of 24bit for recording, no intrinsic quality benefit, just a lot more headroom to play with. Of course, that's only the recording medium signal to noise ratio you don't really have to worry about with 24bit, you've still got mic, environmental noise/noise-floor and pre-amp noise to worry about!

G


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## Dobrescu George

So.... Most bands won't get into recording into 24 bit any time soon because it won't become an ISO standard?...


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## 71 dB

In music production you want dynamic headroom as mentioned before and all digital effects increase noise floor. Say you filter frequencies below 50 Hz out on one track and quantization noise is added at the level of least significant bit. Everytime you do something noise is added and it cumulates. So, you want headroom in the lower end too to be able to drop the noisy least significant bits away when downsampling to 16 bit for CD.


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## pinnahertz (Oct 10, 2017)

71 dB said:


> In music production you want dynamic headroom as mentioned before and all digital effects increase noise floor. Say you filter frequencies below 50 Hz out on one track and quantization noise is added at the level of least significant bit. Everytime you do something noise is added and it cumulates. So, you want headroom in the lower end too to be able to drop the noisy least significant bits away when downsampling to 16 bit for CD.


While true in theory, in practice there are 4 to 6 LSB worth of noise already because there ain't but one ADC in the world capable of better than 20 bits of noise performance, even if it's a 24 bit ADC.  Add a mic preamp in front of the ADC, and it's even worse.  I doubt the low level noise added by any effect would change what's already there much at all.  Especially since internal DAW processing is already at 64 bit fp.


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## pinnahertz

Dobrescu George said:


> So.... Most bands won't get into recording into 24 bit any time soon because it won't become an ISO standard?...


I'm not sure what "most bands" means, but the default today is to record at 24 bits...because you can, and it's useful in production to do so.  All DAWs are up to 32 or 64 bit floating point internally.


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## bigshot

I've never done anything above 24/96 though. Not much reason to. It can only cause problems.


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## pinnahertz

bigshot said:


> I've never done anything above 24/96 though. Not much reason to. It can only cause problems.


You should try 25/97, it's just a bit better.


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## 71 dB

pinnahertz said:


> While true in theory, in practice there are 4 to 6 LSB worth of noise already because there ain't but one ADC in the world capable of better than 20 bits of noise performance, even if it's a 24 bit ADC.  Add a mic preamp in front of the ADC, and it's even worse.  I doubt the low level noise added by any effect would change what's already there much at all.  Especially since internal DAW processing is already at 64 bit fp.



Yeah, of course. 24 bits means the additional noise happens at 48 dB lower level. Prosessing can be at 64 bit floating point, but in the end you need to truncate + dither it to 24 bits.


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## gregorio

71 dB said:


> [1] Yeah, of course. 24 bits means the additional noise happens at 48 dB lower level. [2] Prosessing can be at 64 bit floating point, but in the end you need to truncate + dither it to 24 bits.



1. No it doesn't. When processing, 24bit or 16bit makes no difference because the processing occurs in a 64bit processing/mixing environment and the noise added from processing in the LSB is many, many hundreds of dB below audibility. Even with 1,000 channels of audio and multiple processors on each, still the noise would be way below audibility! If I remember correctly (and I may not as it was about 15 years ago) even with the older 48bit fixed mixing environment, you needed 277 processors for the resultant quantisation noise to reach -120dB.

2. Noise from truncating 64bit float to 24bit fixed would average the LSB, IE. Peak average of truncation noise would be at -138dB. As this is unresolvable, no one I know bothers dithering to 24bit. Even dithering to 16bit is only a standard procedure on the basis of "better safe than sorry", rather than because the resultant error is likely to be audible.



Dobrescu George said:


> So.... Most bands won't get into recording into 24 bit any time soon because it won't become an ISO standard?...



I can understand you not wanting to read the entire thread before posting but at least the OP and the previous handful of posts would be advisable!

G


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## 71 dB

gregorio said:


> 1. No it doesn't. When processing, 24bit or 16bit makes no difference because the processing occurs in a 64bit processing/mixing environment and the noise added from processing in the LSB is many, many hundreds of dB below audibility. Even with 1,000 channels of audio and multiple processors on each, still the noise would be way below audibility! If I remember correctly (and I may not as it was about 15 years ago) even with the older 48bit fixed mixing environment, you needed 277 processors for the resultant quantisation noise to reach -120dB.



I am talking about the the noise cumulation when several 64 bit processing are done. After every processing you truncate + dither the result and that's when 16 bit or 24 bit makes a big difference. At 16 bit the noise cumulates fast above -90 dBFS* , while at 24 bit the noise level is perhaps -110 dBFS, the original noise from mic amps etc. because the cumulated noise add the noise power so little.

* Let's assume you have 10 tracks at 16 bits truncated (no dither) to from 24 bit. Your quantization noise for each track is at level -98 dBFS. When you mix those tracks together, the noise power gets 10-fold and is at level -88 dBFS. If you have 100 tracks, the end result noise is at level -78 dBFS, which starts to be bad. 



gregorio said:


> 2. Noise from truncating 64bit float to 24bit fixed would average the LSB, IE. Peak average of truncation noise would be at -138dB. As this is unresolvable, no one I know bothers dithering to 24bit. Even dithering to 16bit is only a standard procedure on the basis of "better safe than sorry", rather than because the resultant error is likely to be audible.



The dynamic range at 24 bit without dither is 20 * log (2^24 * sqrt(3/2)) = 146.26 dB. Yes, 24 bit is fine without dither, but every CD use dither, which extends _usable_ dynamic range 10-20 dBs.

Are you saying 16 bits is just fine in studio, because prosessing is done at 64 bit? My understanding is that 24 bits allow nice headroom and no more than 32 bits is needed in prosessing. For consumer audio 16bit / 44.1 kHz is enough.


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## gregorio

71 dB said:


> [1] I am talking about the the noise cumulation when several 64 bit processing are done. [2] After every processing you truncate + dither the result and that's when 16 bit or 24 bit makes a big difference.



1. So am I or rather, I'm talking about many hundreds of 64bit processes being done!

2. No you do not truncate + dither after every processing and that's why there is no difference between 16 and 24 bit! The entire mixing process is carried out in the 64bit mixing environment. You don't perform a process on a track, bounce the track back to a 16 or 24 bit file (with truncation or dither), then apply some other process at 64bit and bounce/truncate it back to 16 or 24 bit again, ad infinitum until you finish the mix. If you did, it would take forever to create a mix and would end up, as you suggest, with significant noise. You would get significant noise even if you followed this process with 24 bit, which is why this is not how mixing works and why no mixing environments are 24 bit!! What actually happens is that your 16 or 24bit tracks are loaded into a 64bit mixing environment. You perform a process and the result stays in the mixer (in RAM at 64bit), you perform another process (or numerous other processes) at 64bit using the 64bit result from the last process and everything stays at 64bit all the time, including the truncation error. The only point at which there is any truncation above the 64th bit is when the mixing is complete and you bounce that completed mix out of the mixing environment to say an audio file (in 16 or 24bit). Therefore, there is no accumulation of 16 or 24bit truncation noise, all accumulated truncation noise occurs at 64bit and is completely inaudible even with many hundreds of tracks and processors!

The rest of your post is therefore irrelevant/incorrect!

G


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## 71 dB

gregorio said:


> 1. So am I or rather, I'm talking about many hundreds of 64bit processes being done!
> 
> 2. No you do not truncate + dither after every processing and that's why there is no difference between 16 and 24 bit! The entire mixing process is carried out in the 64bit mixing environment. You don't perform a process on a track, bounce the track back to a 16 or 24 bit file (with truncation or dither), then apply some other process at 64bit and bounce/truncate it back to 16 or 24 bit again, ad infinitum until you finish the mix. If you did, it would take forever to create a mix and would end up, as you suggest, with significant noise. You would get significant noise even if you followed this process with 24 bit, which is why this is not how mixing works and why no mixing environments are 24 bit!! What actually happens is that your 16 or 24bit tracks are loaded into a 64bit mixing environment. You perform a process and the result stays in the mixer (in RAM at 64bit), you perform another process (or numerous other processes) at 64bit using the 64bit result from the last process and everything stays at 64bit all the time, including the truncation error. The only point at which there is any truncation above the 64th bit is when the mixing is complete and you bounce that completed mix out of the mixing environment to say an audio file (in 16 or 24bit). Therefore, there is no accumulation of 16 or 24bit truncation noise, all accumulated truncation noise occurs at 64bit and is completely inaudible even with many hundreds of tracks and processors!
> 
> ...



Okay, I see. I didn't know that because I work with Audacity. Millionaires can work with their 64 bit monsters… …thanks for the lecture.


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## gregorio

71 dB said:


> Okay, I see. I didn't know that because I work with Audacity. Millionaires can work with their 64 bit monsters… …thanks for the lecture.



Logic Pro = $199.99
Pro Tools = $199 (annual subscription).
Reaper = $60

You don't need to be a millionaire! And 64bit isn't a monster, my iPhone is 64bit and cost more than these DAWs!

G


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## pinnahertz

gregorio said:


> Logic Pro = $199.99
> Pro Tools = $199 (annual subscription).
> Reaper = $60
> 
> ...


I find Audacity a useful tool on occasion, but too limiting for real work.  Great tool for free, but hardly an industry standard.   G, you'll be interested to know that when I needed deliberate clipping Audacity was the tool that did it, but I had to set it to 16 bit (it also does 32 bit float).


----------



## gregorio

pinnahertz said:


> G, you'll be interested to know that when I needed deliberate clipping Audacity was the tool that did it, but I had to set it to 16 bit (it also does 32 bit float).



Yep, taking it out and into another system/DAW is how I solved the problem at the time. I could of course just have looped back and clipped the ADC/DAC. I can also achieve it with a couple of the plugins I now own, which specifically provide an option to create digital clipping and various other sorts of overload distortion. This is just between you and me though! 

The point I was making is that the quoted statement was nonsense, it's impossible to accidentally clip a pro DAW and without specific tools or leaving the DAW it's even impossible (AFAIK) to clip deliberately!

G


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## 71 dB

My fiend has Logic Pro and I have heard the horror stories enough. Complex software means tons of bugs and when you uppgade your OS, anything can happen… 



pinnahertz said:


> I find Audacity a useful tool on occasion, but too limiting for real work.



That's why I have wrote maybe two dozen nyguist plugins for it (+ you'll find plugins only by other people).


----------



## mk3

gregorio said:


> It is in fact the fundamental tenet of the Nyquist-Shannon Sampling Theorem on which the very existence and invention of digital audio is based. From WIKI: “In essence the theorem shows that an analog signal that has been sampled can be *perfectly* reconstructed from the samples”. I know there will be some who will disagree with this idea, unfortunately, disagreement is NOT an option. This theorem hasn't been invented to explain how digital audio works, it's the other way around. Digital Audio was invented from the theorem, if you don't believe the theorem then you can't believe in digital audio either!!



Would it not rather be the case that, were the Nyquist-Shannon theorem incorrect, digital audio would still work, just imperfectly?
(That is, if the theorem were incorrect, it might still be the case that sampling can reconstruct the analogue waveform very well, nearly perfectly within the limits of human hearing, although with some relatively minor errors, and that imperfect reconstruction would still be sufficient to have excellent and usable digital audio?)


----------



## Dobrescu George

mk3 said:


> Would it not rather be the case that, were the Nyquist-Shannon theorem incorrect, digital audio would still work, just imperfectly?
> (That is, if the theorem were incorrect, it might still be the case that sampling can reconstruct the analogue waveform very well, nearly perfectly within the limits of human hearing, although with some relatively minor errors, and that imperfect reconstruction would still be sufficient to have excellent and usable digital audio?)



The basic idea from the theorem cannot be incorrect by itself. The theory states that a 22kHz signal can be faitfully reproduced by 22*2 = 44.000 samples per second. since the samples are a basic sine, the idea itself works. 

I think that there is much more that can be talked around how the sampling is done, how precise the microphones are, how precise the headphones are at reconstructing the original waveform, and even discussing how good a DAC or the ADC is...


----------



## mk3

Regarding the Nyquist theorem, this still troubles me.
If I understand correctly, the claim is that Nyquist theorem underlies the purported ability of digital audio to perfectly re-create the original analogue waveform.
Real-world analogue audio is not actually band-limited; but our hearing range is, maximally around 20Hz-20kHz.  The application of the theorem to digital audio leads to the claim that 44.1kHz (CD-quality, more than double the upper limit of human hearing) digital audio is sufficient to perfectly re-construct audio in the human audible range.

My concern/confusion:  A musical signal is complex, not simply a time-varying sine wave.  A musical signal is a sum or superposition of many (sometimes very many) time-varying signals of different shapes, volumes, and frequencies.  This is true of a single acoustic instrument like a violin, piano, or human voice; all the more complex in a full arrangement of multiple sound sources. Once you sum so many waveforms together, isn’t the result a waveform with time variation far more detailed than the sampling frequency?  That is, the rate of change of the final waveform can be much finer than the sampling rate time “slice”; within one slice there may be massive variation of the signal.  So won’t the sampling at Nyquist frequency (or even at any finite frequency) fail to capture these variations?  

I’ve tried to find a good definitive article or site to finally prove this to myself, but to no avail so far.  If you happen across one, please send it my way.  Pictures help my limited intellect.  Thank you for your patience; those who find these matters clear and obvious, please forgive my trying your patience. (Please be kind; I am not asking this to troll or ignite any unpleasantries).


----------



## csglinux

mk3 said:


> Regarding the Nyquist theorem, this still troubles me.
> If I understand correctly, the claim is that Nyquist theorem underlies the purported ability of digital audio to perfectly re-create the original analogue waveform.
> Real-world analogue audio is not actually band-limited; but our hearing range is, maximally around 20Hz-20kHz.  The application of the theorem to digital audio leads to the claim that 44.1kHz (CD-quality, more than double the upper limit of human hearing) digital audio is sufficient to perfectly re-construct audio in the human audible range.
> 
> ...



In theory, it doesn't matter how complex the signal is. You can (theoretically) perfectly reconstruct the original analog signal up to the Nyquist limit. Even so, there are a couple of issues I'm still on the fence with...

There have been a few attempts recently to fuzz around Shannon-Nyquist with claims about timing and the fact that the human ear/brain doesn't work like a sum of Fourier modes. It sounds plausible, but I've yet to see any conclusive scientific evidence for it.  The potential problem is that when we reconstruct sharp transients (like a square wave) with a sum of sinusoidal modes up to (say ) 22 kHz, you don't have a perfect representation of the square wave, but the leading and trailing edges have pre- and post-ringing due to Gibb's phenomena. In theory, filters and/or higher sampling rates can limit the extent of the pre-and post ringing - which most definitely are not natural. (E.g., if a blast wave were headed towards you, you'd have no clue until the instant it hit you.)

@gregorio I'd be interested in your thoughts on the following interview with Rob Watts:



Check it out at about the 6 minute mark. He makes claims about soundstage depth improvements by reducing the effect from noise shapers by 100, 200, 300 dB. Way below what we (even he) would expect. And yet he claims he hears an improvement. (I've not experienced this myself. I've heard a DAVE, but don't own one, so I can't conduct these kinds of tests.) I wonder if this isn't a little bit of marketing hyperbole? On the other hand, I hate to be a skeptic and I never like to tell somebody else what they did or didn't hear. I believe there could be subtleties to our hearing that we might not quite have nailed down yet.


----------



## siberianmoon (Oct 11, 2017)

mk3 said:


> My concern/confusion:  A musical signal is complex, not simply a time-varying sine wave.  A musical signal is a sum or superposition of many (sometimes very many) time-varying signals of different shapes, volumes, and frequencies.  This is true of a single acoustic instrument like a violin, piano, or human voice; all the more complex in a full arrangement of multiple sound sources. Once you sum so many waveforms together, isn’t the result a waveform with time variation far more detailed than the sampling frequency?


You are basically arguing that if two sound waves are combined, they can form a new soundwave with a higher frequency that would not be sampled accurately enough. This means the soundwave in question would have to have a frequency over 22 050 Hz – that's inaudible to humans any way.

Also, I think, it could be reduced to 'what if a single soundwave is not sampled at its peak but just before or after'. According to the theorem you quoted, it doesn't matter. The peak can be calculated from the two* consecutive samples that 'surround' it.

(* Actually I don't know how many samples it takes to calculate a wave form.)


----------



## pinnahertz

71 dB said:


> My fiend has Logic Pro and I have heard the horror stories enough. Complex software means tons of bugs and when you uppgade your OS, anything can happen…


Well..."it depends"... Yes, upgrades are sometimes side-grades, or back-grades in some ways, but if someone was going to use any software (not just DAW) for producing real work then the entire system should be  considered as the "tool", and once working (and a backup image made), the system should be "frozen" until such time as updates and upgrades and their impact is fully understood.  I installed several ProTools systems in professional studios and once working well, locked them down.  That way the tool is useful for the longest period of time.  

The same holds true for any complex software and OS. 

I once tried a simple multitrack project in Audacity.  Tracking went well, mixdown...well I exported it to something else to preserve my hair.


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## bigshot (Oct 11, 2017)

The idea that "complex sounds" are harder to reproduce than "simple sounds" is incorrect. 16/44.1 is able to perfectly reproduce *all* sounds that can be heard. If it couldn't reproduce some audible sounds as well as others, that would be expressed as distortion. Digital audio doesn't have audible levels of distortion.


----------



## castleofargh

mk3 said:


> Regarding the Nyquist theorem, this still troubles me.
> If I understand correctly, the claim is that Nyquist theorem underlies the purported ability of digital audio to perfectly re-create the original analogue waveform.
> Real-world analogue audio is not actually band-limited; but our hearing range is, maximally around 20Hz-20kHz.  The application of the theorem to digital audio leads to the claim that 44.1kHz (CD-quality, more than double the upper limit of human hearing) digital audio is sufficient to perfectly re-construct audio in the human audible range.
> 
> ...


the signal is indeed band limited and if that limit was too close to our hearing threshold and had fairly high amplitude signals, then it would be an audible issue. well a small one as all we'd have to do is jump to 48khz, boom problem solved. ^_^
as for complex signal vs simple signal. it is in effect irrelevant. the hardest signal to deal with is a full amplitude sine at a frequency close to the band limiting. something like a very loud 21khz is way more difficult for 16/44 to handle than a hundred people in a choir. what we subjectively identify as complex has no significance for a signal that is just in the end one amplitude moving over time. only how fast it has to move can be an issue if the band limiting is too close. complex signal doesn't increase the speed. the maximum frequency of the signal does. 
but in any case this has little to do with bit depth.


----------



## pinnahertz

mk3 said:


> Regarding the Nyquist theorem, this still troubles me.
> If I understand correctly, the claim is that Nyquist theorem underlies the purported ability of digital audio to perfectly re-create the original analogue waveform.
> Real-world analogue audio is not actually band-limited; but our hearing range is, maximally around 20Hz-20kHz.  The application of the theorem to digital audio leads to the claim that 44.1kHz (CD-quality, more than double the upper limit of human hearing) digital audio is sufficient to perfectly re-construct audio in the human audible range.


Actually the real world of audio is band limited, and so is hearing.  What you're missing is the specific degree of content above hearing range.  Things roll off quite a lot up there for many reasons, propagation of ultrasonics being a biggie.  Even if a set of jangling keys can be shown to produce ultrasonic content (it can), does that content get to our hears?  Turns out, not much. 


mk3 said:


> My concern/confusion:  A musical signal is complex, not simply a time-varying sine wave.  A musical signal is a sum or superposition of many (sometimes very many) time-varying signals of different shapes, volumes, and frequencies.  This is true of a single acoustic instrument like a violin, piano, or human voice; all the more complex in a full arrangement of multiple sound sources. Once you sum so many waveforms together, isn’t the result a waveform with time variation far more detailed than the sampling frequency?  That is, the rate of change of the final waveform can be much finer than the sampling rate time “slice”; within one slice there may be massive variation of the signal.


There is far less high frequency content in acoustic music than most people realize.  Complex sounds add together, but adding doesn't increase the total bandwidth unless there's a lot of nonlinear distortion.  If you were to mix 10kHz and 12kHz you don't get anything but 10kHz and 12kHz mixed unless the system is nonlinear and causes intermodulation products.


mk3 said:


> So won’t the sampling at Nyquist frequency (or even at any finite frequency) fail to capture these variations?


Sure it will, but it doesn't matter.  Ultrasonic content is already attenuated more than 40dB, typically, and more with rising frequency.  Add to that microphone response, propagation losses, and the highly directional nature of ultrasonic energy, as a practical matter there's nothing much there to capture, so filtering at Nyquist does not damage anything. 

The argument that we need that content is the thrust of the hi-res audio proponents, but of course, there's no real data to support that.


----------



## mk3

siberianmoon said:


> You are basically arguing that if two sound waves are combined, they can form a new soundwave with a higher frequency that would not be sampled accurately enough. This means the soundwave in question would have to have a frequency over 22 050 Hz – that's inaudible to humans any way.
> 
> Also, I think, it could be reduced to 'what if a single soundwave is not sampled at its peak but just before or after'. According to the theorem you quoted, it doesn't matter. The peak can be calculated from the two* consecutive samples that 'surround' it.
> 
> (* Actually I don't know how many samples it takes to calculate a wave form.)



Thank you for the reply.  
How do we characterize the frequency (and thus speak of the bandwidth limits) of a complex signal, one that has little (or no) repeating patterns?  (Can even such a signal exist, or are there always repeating patterns)?

My question also applies to relatively simple signals.

Let's say we sample pure waveform tone (perhaps a single oscillator of a perfect synthesizer) at 20kHz sampling frequency.
Now let that synthesizer produce a 10kHz  sine wave, and then after a time produce a triangle wave of equal amplitude.  

Supposedly the sampler (at double the frequency) can perfectly capture this waveform, allowing perfect re-construction according to the Theorem.

If the samples happen to be taken at the peaks and lows of the waveform, wouldn't the samples taken for the sine and triangle waves be identical?
If so, then how could any D/A algorithm "know" which waveform (sine, triangle, or other) to reconstruct?
I must be missing something obvious in my thinking here.


----------



## csglinux

mk3 said:


> Thank you for the reply.
> How do we characterize the frequency (and thus speak of the bandwidth limits) of a complex signal, one that has little (or no) repeating patterns?  (Can even such a signal exist, or are there always repeating patterns)?
> 
> My question also applies to relatively simple signals.
> ...



Up to Nyquist, your sawtooth wave and sine wave are the same. To differentiate the two, you'd have to have a higher sampling rate, since your triangular waveform involves higher frequency content than your sine wave.


----------



## castleofargh

mk3 said:


> Thank you for the reply.
> How do we characterize the frequency (and thus speak of the bandwidth limits) of a complex signal, one that has little (or no) repeating patterns?  (Can even such a signal exist, or are there always repeating patterns)?
> 
> My question also applies to relatively simple signals.
> ...



triangle wave, really? where does that even come from? 
this is just saying the same thing in a trick question. a triangle signal can be expressed as a series of sine waves, some of which will reach pretty high frequencies to perfectly represent the pointy part. if that frequency is bigger than 1/2 of the sample rate, Nyquist theorem is simply not followed. so you're really asking something like, can you encode 30khz with a 20khz band limiting? and of course the answer is no. 
now will it make any difference in what you would hear? that's unlikely because the headphone and your ear will both low pass/band limit the signal anyway. which one will do it sooner depends on ears and gears. but as long as the file does band limiting at a higher frequency, we won't notice. 

also you need to keep a clear view of what you're trying to achieve here. if you want perfect reproduction of any signals, then the highest resolution is obviously better. but as soon as you're talking human hearing, the limit is what we can hear. more is the same as a lot more and perfection was never invited.


----------



## pinnahertz

mk3 said:


> Thank you for the reply.
> How do we characterize the frequency (and thus speak of the bandwidth limits) of a complex signal, one that has little (or no) repeating patterns?  (Can even such a signal exist, or are there always repeating patterns)?


Signals with no repeating patterns are generally classified as "noise". Even noise can have a very well defined bandwidth.  You can't just consider wave forms when you think about the spectrum of a signal.


mk3 said:


> My question also applies to relatively simple signals.
> 
> Let's say we sample pure waveform tone (perhaps a single oscillator of a perfect synthesizer) at 20kHz sampling frequency.
> Now let that synthesizer produce a 10kHz  sine wave, and then after a time produce a triangle wave of equal amplitude.
> ...


What you're missing is the spectrum of each of those signals.  A 20kHz sine wave, assuming it's pure, is made up of just 20kHz.  If sampled at 44.1, it won't be just sampled at peaks and valleys, but eventually at every valid voltage point, since it's not related directly to the sampling frequency. 

Same with the 10kHz sine wave, except there will be twice as many samples per cycle.  Both will be correctly and losslessly reconstructed.

The theoretical 10kHz triangle is a bit different as it contains odd-order harmonics, the first significant being the third, and with intense harmonics extending to infinity at progressively lower intensity.  Since that third harmonic frequency is 30kHz, it and everything above is filtered off by the anti-aliasing filter, and the result is basically a sine wave.  However, that's also what you'd hear, since your hearing doesn't extend high enough to detect the third harmonic.  Same with a square wave, though the harmonic content is of greater amplitude, it's still all odd-order harmonics extending theoretically to infinity.  The only actual square or triangle wave sources that come even close to the theoretical would be lab-grade test instruments.  No synth can do it, and no digital generator can either, though they may come close enough. 

Now before anyone goes and tries to test this, like to see if you can hear the difference between a 10kHz sine wave and a 10kHz square or triangular wave,  it won't actually work right because it's impossible to transduce a 10kHz triangle or square wave correctly.  Transducers have limited bandwidth and tons of nonlinearity, so you'll get a mangled square wave out of it even if you supply something perfect going it, with all sorts of sub-harmonics that are audible.  You'd have to pre-filter the test signals and limit them to the most linear range of the transducer to be a meaningful test, otherwise you'll hear a difference even if it's a difference only generated in the transducer. 

But to be clear, those are very, very special cases.  They don't occur in actual audio, and the reproduced result is actually representative of what would be heard.

Again, it doesn't matter, we don't hear those signals anyway.


----------



## mk3

castleofargh said:


> triangle wave, really? where does that even come from?
> this is just saying the same thing in a trick question. a triangle signal can be expressed as a series of sine waves, some of which will reach pretty high frequencies to perfectly represent the pointy part. if that frequency is bigger than 1/2 of the sample rate, Nyquist theorem is simply not followed. so you're really asking something like, can you encode 30khz with a 20khz band limiting? and of course the answer is no.
> now will it make any difference in what you would hear? that's unlikely because the headphone and your ear will both low pass/band limit the signal anyway. which one will do it sooner depends on ears and gears. but as long as the file does band limiting at a higher frequency, we won't notice.
> 
> also you need to keep a clear view of what you're trying to achieve here. if you want perfect reproduction of any signals, then the highest resolution is obviously better. but as soon as you're talking human hearing, the limit is what we can hear. more is the same as a lot more and perfection was never invited.



Triangle waves: I must be spending too much time with synthesizers.

I didn't intend to be tricky. It seems I am misunderstanding what is meant by bandwidth limiting (hence my parallel question of what the "frequency" of a complex, or virtually non-repeating signal, really means); so the assumption is that such a limit is based on maximum _sine-wave_ frequency (not maximum frequency of _various_ wave shapes), presumably because sine waves are considered fundamental?

So then apparently a 20kHz sampler _cannot_ reproduce a 10kHz square wave, triangle wave, saw wave, etc.


----------



## bigshot

I think most of us listen to music. We use our equipment to play back music. Weird waveforms and ultra high frequency content are nice and all in theory. But what's the point of reproducing sound that our ears can't even reproduce? When it comes to reproducing the thing that really matters- music (even electronic music), 16/44.1 does a perfect job of it for human ears. That's all that matters for me.


----------



## Dobrescu George

castleofargh said:


> triangle wave, really? where does that even come from?
> this is just saying the same thing in a trick question. a triangle signal can be expressed as a series of sine waves, some of which will reach pretty high frequencies to perfectly represent the pointy part. if that frequency is bigger than 1/2 of the sample rate, Nyquist theorem is simply not followed. so you're really asking something like, can you encode 30khz with a 20khz band limiting? and of course the answer is no.
> now will it make any difference in what you would hear? that's unlikely because the headphone and your ear will both low pass/band limit the signal anyway. which one will do it sooner depends on ears and gears. but as long as the file does band limiting at a higher frequency, we won't notice.
> 
> also you need to keep a clear view of what you're trying to achieve here. if you want perfect reproduction of any signals, then the highest resolution is obviously better. but as soon as you're talking human hearing, the limit is what we can hear. more is the same as a lot more and perfection was never invited.



You seem like a knowledgeable person. 

A 30 kHz sound

A 25 kHz sound 

A 40 kHz sound 

A 50 kHz sound

Any of those can produce, depending on their loudness, any harmonics that would be audible to humans? I just want a straight answer. 



bigshot said:


> I think most of us listen to music. We use our equipment to play back music. Weird waveforms and ultra high frequency content are nice and all in theory. But what's the point of reproducing sound that our ears can't even reproduce? When it comes to reproducing the thing that really matters- music (even electronic music), 16/44.1 does a perfect job of it for human ears. That's all that matters for me.



Now I am going to respectfully say that I want saw waves. For music. For Mindless Self Indulgence albums. I cannot guarantee whether you will like their music or not, but I do, and I want perfect sine waves because the synths they used to that and they have a vrey interesting texture. Seems that the better the top end is for a IEM / Headphone, the better those sound., 

I do not believe that bit depth or if music is Hi-Res will have any impact on that and I do believe that it still is within the range of what humans can hear, but having good frequency response above 10 kHz, so in 10 - 20 kHz really helps this music.


----------



## pinnahertz (Oct 11, 2017)

You might not be aware of it, but there is no audio recorder that actually can accurately reproduce a 10 kHz square or triangular wave.  No analog recorder, tape or vinyl, or digital system. Not even digital at 20/192!
Nor  is there any need to record and accurately reproduce the theoretical test signal with  infinite harmonics  anywhere in the audio world.


----------



## 71 dB

pinnahertz said:


> I once tried a simple multitrack project in Audacity.  Tracking went well, mixdown...well I exported it to something else to preserve my hair.



I have never had problems with downmixing a multichannel project in Audacity.


----------



## pinnahertz

Dobrescu George said:


> You seem like a knowledgeable person.
> 
> A 30 kHz sound
> 
> ...


No.  But when those sounds occur in nonlinear systems, intermodulation products occur that are audible. Those are defects though.  Is that straight enough?


Dobrescu George said:


> I do not believe that bit depth or if music is Hi-Res will have any impact on that and I do believe that it still is within the range of what humans can hear, but having good frequency response above 10 kHz, so in 10 - 20 kHz really helps this music.


Of course having response above 10 kHz to 20 kHz helps music, but it does not help you reproduce triangular or square waves.


----------



## Dobrescu George

pinnahertz said:


> You might not be aware of it, but there is no audio recorder that actually can accurately reproduce a 10 kHz square or triangular wave.  No analog recorder, tape or vinyl, or digital system. Not even digital at 20/192!
> Nor  is there any need to record and accurately reproduce the theoretical test signal with  infinite harmonics  anywhere in the audio world.



I am not saying there is, but certain IEMs and headphones sound much better with Mindless Self Indulgence's music. 



pinnahertz said:


> No.  But when those sounds occur in nonlinear systems, intermodulation products occur that are audible. Those are defects though.  Is that straight enough?
> 
> Of course having response above 10 kHz to 20 kHz helps music, but it does not help you reproduce triangular or square waves.



So... There are audible stuffs there.

As for triangular waves, I may not know what Mindless self indulgence employs, but it is synth stuff and I like it. and I know that it looks like it has a lot of really saw-y stuff if viewed under Audacity or an oscilloscope


----------



## castleofargh

mk3 said:


> Triangle waves: I must be spending too much time with synthesizers.
> 
> I didn't intend to be tricky. It seems I am misunderstanding what is meant by bandwidth limiting (hence my parallel question of what the "frequency" of a complex, or virtually non-repeating signal, really means); so the assumption is that such a limit is based on maximum _sine-wave_ frequency (not maximum frequency of _various_ wave shapes), presumably because sine waves are considered fundamental?
> 
> So then apparently a 20kHz sampler _cannot_ reproduce a 10kHz square wave, triangle wave, saw wave, etc.


yup. because the sampling theorem is all about sine waves and they can be used to define the other signals. when we say 10khz square wave, it defines the repeating rate of the squared shape, not the frequency of all the sines that make up the square wave. a perfect square wave contains an infinite number of sines at increasing frequencies(and decreasing amplitude, probably one of the reasons why we don't miss them that much when some are missing). the lowest being indeed a 10khz sine(see Fourier for all the mathematical fun).  so no band limited signal can reproduce it perfectly, simply because a strict limit doesn't go well with the concept of infinity.  
and in real life aside from quantum theory, you'll be hard pressed to find anything that can go from one position to another one instantaneously. 
so instead we tend to care for what's audible, and try to almost perfectly reproduce that part at least for music. when it's about internet signal transmission, we want our signal to change amplitude even faster to be able to send a lot more data in a given delay. for that, band limiting at 44khz would suck. different needs, different standards. 

but to be clear, there is nothing wrong with increasing resolution and sample rate, as long at it doesn't lead to other issues. but we do have to keep in mind real life use starting with human limits. if our ear doesn't know what to do with the ultrasonic content of a signal, there is little benefit to making huge files just to have that extra content. just like there is little interest in having a TV that reproduces accurately ultraviolet frequencies or above. the eye is not going to notice anyway, and it could actually have some negative impact.


----------



## castleofargh

Dobrescu George said:


> You seem like a knowledgeable person.
> 
> A 30 kHz sound
> 
> ...


then I'm becoming better at pretending to know stuff. ^_^
if loudness isn't limited, then we can physically feel ultrasounds. that has been established. but the levels were not really applicable to typical musical content and humanly reasonable listening levels. 

not sure if I really get what you're pointing at. are we considering a perfect DAC, amp and headphone? or do we account for the distortions they could produce themselves in the audible range when fed with high amplitude ultrasounds? because that's another can of worms.


----------



## Jonman503

All my music in is 24 bit


----------



## Dobrescu George

castleofargh said:


> then I'm becoming better at pretending to know stuff. ^_^
> if loudness isn't limited, then we can physically feel ultrasounds. that has been established. but the levels were not really applicable to typical musical content and humanly reasonable listening levels.
> 
> not sure if I really get what you're pointing at. are we considering a perfect DAC, amp and headphone? or do we account for the distortions they could produce themselves in the audible range when fed with high amplitude ultrasounds? because that's another can of worms.



Okay.... 

Yes, I think I was referring much more to what the audible distortions would be if we fed the drivers and the DACs / AMPs with ultrasonics rather than just have them exist, because besides the typical harmonics, I was wondering if having Hi-Res music with what would probably be Hi-Res noise, would in fact lead to some effects on the drivers / DACs / AMPs, especially if the said drivers were not optimized to go beyond 20 kHz, or the DAC or the AMP (given capacitor input, how the roll-off filter impacts the sound and all).

I am asking not for me, but to understand why people hear Hi-Res better. My music is not in hi-res and I won't judge music I'm not accustomed to... 

I don't think the reason some IEMs present Mindless Self Indulgence with better subjective sound is related to ultrasonics then


----------



## Darren G

Personally I look at transducers. They have mass.  They can only be accelerated/decelerated so fast.  Likewise I look at my own ears, mass, limits, etc.  Personally I stopped fretting that there are limits on the frequencies I can hear, and just want the frequencies I can to sound good.


----------



## RRod

mk3 said:


> Triangle waves: I must be spending too much time with synthesizers.
> 
> I didn't intend to be tricky. It seems I am misunderstanding what is meant by bandwidth limiting (hence my parallel question of what the "frequency" of a complex, or virtually non-repeating signal, really means); so the assumption is that such a limit is based on maximum _sine-wave_ frequency (not maximum frequency of _various_ wave shapes), presumably because sine waves are considered fundamental?
> 
> So then apparently a 20kHz sampler _cannot_ reproduce a 10kHz square wave, triangle wave, saw wave, etc.



Sine waves are the fundamental units of decomposition for the Fourier transform. Nyquist is a theorem about Fourier transforms. The question it answers is "When can the Fourier transform of a function (and thus the function itself) be exactly reproduced from the discrete-time Fourier transform (DTFT)?" The answer it gives is "When the Fourier transform has bounded support (i.e. the function is bandlimited) and the sampling frequency is at least twice that bound."

Triangle/square/saws etc. are *NOT* bandlimited functions, and thus they violate Nyquist and are not perfectly representable by the DTFT, not at 44.1kHz, not at 96kHz, not at 1e100kHz. But that doesn't matter because no microphone (or natural phenomenon) ever produced a perfect triange/square/saw wave! Then add on top of that the capabilities of the human ear, and you find much fewer problems with audio than some people want you to believe.

As far as "virtually non-repeating signals", note that a large class of functions allow Fourier transforms, again certainly anything you're getting out of a mic.


----------



## RRod (Oct 11, 2017)

Dobrescu George said:


> Okay....
> 
> Yes, I think I was referring much more to what the audible distortions would be if we fed the drivers and the DACs / AMPs with ultrasonics rather than just have them exist, because besides the typical harmonics, I was wondering if having Hi-Res music with what would probably be Hi-Res noise, would in fact lead to some effects on the drivers / DACs / AMPs, especially if the said drivers were not optimized to go beyond 20 kHz, or the DAC or the AMP (given capacitor input, how the roll-off filter impacts the sound and all).
> 
> ...



What's missing is the level of the ultrasonic content. If it's low, then the IMD products should also be low unless you have some kind of unbounded distortion. The IMD products would also have to not be audibly masked by OTHER content, distortion or otherwise. So yes, if your benchmark is "can I play high intensity ultrasonic content without lower frequency content and end up with something audible", then, SURE! That isn't exactly what people who hear hi-res as 'better' are talking about, though…


----------



## pinnahertz

71 dB said:


> I have never had problems with downmixing a multichannel project in Audacity.


You need to experience a real pro DAW to understand how limiting Audacity is.


----------



## pinnahertz

Jonman503 said:


> All my music in is 24 bit


So...you up-sampled everything then?


----------



## bigshot

RRod said:


> What's missing is the level of the ultrasonic content. If it's low, then the IMD products should also be low unless you have some kind of unbounded distortion. The IMD products would also have to not be audibly masked by OTHER content, distortion or otherwise. So yes, if your benchmark is "can I play high intensity ultrasonic content without lower frequency content and end up with something audible", then, SURE! That isn't exactly what people who hear hi-res as 'better' are talking about, though…



If you can't hear inaudible frequencies loud, odds are you can't hear them at natural soft volumes either. It might sound better to a fruit bat that way, but not to human ears.


----------



## bigshot

Jonman503 said:


> All my music in is 24 bit



You're missing out on an awful lot of great music.


----------



## gregorio (Oct 12, 2017)

71 dB said:


> My fiend has Logic Pro and I have heard the horror stories enough. Complex software means tons of bugs and when you uppgade your OS, anything can happen…



One of the reasons that Pro Tools so dominated the pro DAW market was because it was so stable and in a pro studio stability is vitally important. If John Williams says, "I love the London Symphony Orchestra, that last take was perfect", the last thing you can afford to say is, "sorry we'll have to do it again, the DAW crashed"! There are of course some horror stories, as there are with any set of complex equipment but compared to much software, pro DAW (and Pro Tools in particular) is extremely stable.



mk3 said:


> Would it not rather be the case that, were the Nyquist-Shannon theorem incorrect, digital audio would still work, just imperfectly? ... The application of the theorem to digital audio ...



No, if the theorem were incorrect there would be no digital audio and no digital anything else! The theorem was not applied to digital audio, it was the other way around, digital audio was applied to the theorem. It was because the theorem was correct that digital audio was developed. There are a couple of points often missed by those in the audiophile community who have a vested interest in demonstrating that the theorem is incorrect/incomplete:

1. Nyquist suggested the basics of the theory in 1924 but in 1948 Caude Shannon mathematically proved it. Later still, when technology had advanced sufficiently, organisations started trying to find a way to engineer technology to fulfil that proven theorem. So we're NOT dealing with just a theory applied to or attempting to explain how digital audio works, we're dealing with a proven theorem upon which digital audio is designed and without which digital audio would not have been developed in the first place.

2. The audiophile community tends to look at the Nyquist-Shannon theorem purely in terms of their own particular interest, music reproduction but actually that is an almost incidental by-product of the theorem. In his 1948 paper "A Mathematical Theory of Communication" Shannon's proof of what is today called the Nyquist-Shannon Theorem does not just cover the perfect quantification, storage and communication of audio information but of ALL information!! Think about that for a moment! ... That proof of what Shannon called "Communication Theory" (but is today called Information Theory), is the basis of all digital technology and for this reason Shannon is sometimes called "the father of the digital age". Indeed, the basic unit of information and entropy, as defined by the IEC, is named the Shannon, although it's now known more popularly as a "bit".  I should therefore have more correctly titled this thread "16Sh vs 24Sh, the Myth Exploded"! Today this theorem crops up all over the place, in numerous fields, from neurobiology to our understanding of back holes. It is, arguably, one of the most important and influential theorems in human history! So no, if the theorem were incorrect there would not be any digital audio, in fact there would not be any digital anything, including the "digital age"!

Regarding waveforms: As essentially stated by others, the Nyquist-Shannon Theorem is correct, it is correct for ALL actual waveforms, irrespective of how simple or complex they are! It is therefore also correct for any actual square, triangle or sawtooth wave! Audiophiles (or those marketing to them) will often hold up some output plot as say; "there you are, that's not an accurate square wave." - which is absolutely true! It's not an accurate square wave because an accurate square wave does not and cannot exist, digital audio accurately captures all the information of what actually does exist, not what audiophiles only believe exists. A common problem in audiophilia I'm afraid and hence the use of the word "Myth" in the title of this thread!

G


----------



## old tech

Dobrescu George said:


> You seem like a knowledgeable person.
> 
> A 30 kHz sound
> 
> ...



You seem to be buying into another one of those audiophile myths (typically the anti CD brigade) which usually go something like this:

"we cannot hear frequencies above 20khz but they do affect frequencies we can hear, therefore the higher the frequency response the more accurate the music playback".

I won't comment whether these ultrasonic frequencies do or do not affect those we can hear and we'll put aside the far more powerful <20hz subsonic frequencies as a contrast and also assume instruments are creating ultrasonic harmonics which are not masked.  If then these ultrasonic frequencies do make a discernible difference then by definition the effect is within range of human hearing.  Therefore the effect and consequences are only relevant to a live acoustic event, not a recording.  The reason is that the recording will have had captured the effects of the ultrasonics, so why would the ultrasonics need to be reproduced as well?  It is a bit like looking at a colourless diamond.  It is clear but (depending on its luminosity) it will glow purple under ultraviolet light.  A picture of the diamond under the ultraviolet light with the purple glow can be taken and then played back on a PC screen.  Note that the playback of the glow (the effect of ultraviolet waveforms) did not require the camera to capture ultraviolet frequencies, nor the PC screen to playback ultraviolet frequencies, and if they did what difference would it make?

Actually that is an interesting question.  If (and it is a big if) ultrasonic frequencies do affect sound we hear and it is recorded with the music, would it double the effect (given the effect from the live event has already been captured)?  If the ultrasonic content is mainly noise, would it then distort the sound if it is not filtered above 20khz?


----------



## 71 dB

pinnahertz said:


> You need to experience a real pro DAW to understand how limiting Audacity is.



How I live my life is not your damn business! I am tired of you trying to make other people feel inferior. Jean-Michel Jarre created Equinoxe without a DAW. Tangerine Dream created Force Majeure without a DAW. It's about creativity and talent and that's something I don't have not matter how expensive "pro" DAW I use.


----------



## 71 dB

gregorio said:


> One of the reasons that Pro Tools so dominated the pro DAW market was because it was so stable and in a pro studio stability is vitally important. If John Williams says, "I love the London Symphony Orchestra, that last take was perfect", the last thing you can afford to say is, "sorry we'll have to do it again, the DAW crashed"! There are of course some horror stories, as there are with any set of complex equipment but compared to much software, pro DAW (and Pro Tools in particular) is extremely stable.



DAWs pay back themselves in 17 seconds if you make the soundtrack of the newest Star Wars or the new Katy Perry album with it.


----------



## Dobrescu George

RRod said:


> What's missing is the level of the ultrasonic content. If it's low, then the IMD products should also be low unless you have some kind of unbounded distortion. The IMD products would also have to not be audibly masked by OTHER content, distortion or otherwise. So yes, if your benchmark is "can I play high intensity ultrasonic content without lower frequency content and end up with something audible", then, SURE! That isn't exactly what people who hear hi-res as 'better' are talking about, though…



Okay, I think I got it 

It exists, but it is not what people are hearing with their Hi-Res files. 



bigshot said:


> You're missing out on an awful lot of great music.



Ahahahaha - true. This is why I cannot go to hi-res, not a lot of my favorite music is in that format. 



old tech said:


> You seem to be buying into another one of those audiophile myths (typically the anti CD brigade) which usually go something like this:
> 
> "we cannot hear frequencies above 20khz but they do affect frequencies we can hear, therefore the higher the frequency response the more accurate the music playback".
> 
> ...



Very interesting take! 

I actually have only CDs, I mean ONLY CDs right now, I was curious if it is worth to try to move to Hi-Res. Seems not so much


----------



## RRod

bigshot said:


> If you can't hear inaudible frequencies loud, odds are you can't hear them at natural soft volumes either. It might sound better to a fruit bat that way, but not to human ears.



Well the question is whether you can hear non-linearities due to the ear when presented with >20kHz content. My point was that even if you can, the details matter a lot to audibility with anything other than contrived tests with high-volume isolated high frequency content.


----------



## bigshot (Oct 12, 2017)

old tech said:


> If (and it is a big if) ultrasonic frequencies do affect sound we hear and it is recorded with the music, would it double the effect (given the effect from the live event has already been captured)?  If the ultrasonic content is mainly noise, would it then distort the sound if it is not filtered above 20khz?



If the inaudible was audible, how would recording the inaudible affect the audible?



71 dB said:


> Jean-Michel Jarre created Equinoxe without a DAW. Tangerine Dream created Force Majeure without a DAW.



Those albums were recorded when I was in high school. They didn't have ProTools then. I'm sure if those guys weren't in nursing homes and were still recording, they would be using ProTools just like everyone else. It's a standard tool, and much better than the 24 track tape decks used back in the days when dinosaurs ruled the Earth.



RRod said:


> Well the question is whether you can hear non-linearities due to the ear when presented with >20kHz content. My point was that even if you can, the details matter a lot to audibility with anything other than contrived tests with high-volume isolated high frequency content.



There's no reason to believe that presence of super audible frequencies has any impact at all on recorded music. And the range of frequencies we're talking about recording would only be an octave or two above 20kHz. That's a small fraction of the range that truly is audible.



Dobrescu George said:


> I actually have only CDs, I mean ONLY CDs right now, I was curious if it is worth to try to move to Hi-Res. Seems not so much



I thought you said all of your music was 24 bit?


----------



## Dobrescu George

bigshot said:


> If the inaudible was audible, how would recording the inaudible affect the audible?
> 
> Those albums were recorded when I was in high school. They didn't have ProTools then. I'm sure if those guys weren't in nursing homes and were still recording, they would be using ProTools just like everyone else. It's a standard tool, and much better than the 24 track tape decks used back in the days when dinosaurs ruled the Earth.



I actually agree, I'm pretty sure they would be using the latest tech if they could. 

[Not related to the topic] One thing I am disappointed as far as recording goes is Rings of Saturn's latest album. One of the poorest masterings out there...


----------



## pinnahertz

71 dB said:


> How I live my life is not your damn business! I am tired of you trying to make other people feel inferior. Jean-Michel Jarre created Equinoxe without a DAW. Tangerine Dream created Force Majeure without a DAW. It's about creativity and talent and that's something I don't have not matter how expensive "pro" DAW I use.


Whoa!  Chill, man!  I made no comment on your life, I just said you wouldn't understand the difference between free software and a full-on pro DAW without using one.  Just like I wouldn't understand why people spend 100K on a car without test driving one.  

Sure, Jarre created Equinoxe without a DAW, but I'm fairly sure he used one on Aero (it's a 5.1 mix) and the included reboots of Oxegene.


----------



## Dobrescu George

pinnahertz said:


> Whoa!  Chill, man!  I made no comment on your life, I just said you wouldn't understand the difference between free software and a full-on pro DAW without using one.  Just like I wouldn't understand why people spend 100K on a car without test driving one.
> 
> Sure, Jarre created Equinoxe without a DAW, but I'm fairly sure he used one on Aero (it's a 5.1 mix) and the included reboots of Oxegene.



I agree on test driving a car befgore buyiung. 

I'd do that for any car, even a 10.000$ car, I would drive it for at least a week before purchasing, along with getting a lending contract on the other options, at least to have some idea what I'm getting into.


----------



## pinnahertz

Dobrescu George said:


> I agree on test driving a car befgore buyiung.
> 
> I'd do that for any car, even a 10.000$ car, I would drive it for at least a week before purchasing, along with getting a lending contract on the other options, at least to have some idea what I'm getting into.


It's just a little harder to test drive a DAW.  Not impossible, but just a bit more effort is required than a free download.


----------



## castleofargh

Dobrescu George said:


> I actually agree, I'm pretty sure they would be using the latest tech if they could.
> 
> [Not related to the topic] One thing I am disappointed as far as recording goes is Rings of Saturn's latest album. One of the poorest masterings out there...


 we sent 2 golden vinyls to aliens, they never replied because they're scared that if they do reply, they'll immediately receive 2 more every month and have to pay for them+delivery.


----------



## pinnahertz

castleofargh said:


> we sent 2 golden vinyls to aliens, they never replied because they're scared that if they do reply, they'll immediately receive 2 more every month and have to pay for them+delivery.


Intergalactic Columbia House!  Yikes, just think of the shipping charges...


----------



## Yuri Korzunov

Bit reserve need to "transparent" work with musical stuff - don't worry about target format.

General rule: target 16 bit - record 24 bit.

64-bit processing allow to forget about distortions, noise and overload issues (float point, especially).

If target format is 16-bit, 24-bit processing demands permanent control to prevent of sound damaging, many programming issues.

Of course, bit depth reducing can some improve performance (lesser calculation time resources). But it is justifiable in case of resource lack only.


----------



## RRod

bigshot said:


> There's no reason to believe that presence of super audible frequencies has any impact at all on recorded music. And the range of frequencies we're talking about recording would only be an octave or two above 20kHz. That's a small fraction of the range that truly is audible.



Yes exactly: one has to consider the actual context of the HF content, rather than make generalizations based on specific test conditions.


----------



## 71 dB

bigshot said:


> Those albums were recorded when I was in high school. They didn't have ProTools then. I'm sure if those guys weren't in nursing homes and were still recording, they would be using ProTools just like everyone else. It's a standard tool, and much better than the 24 track tape decks used back in the days when dinosaurs ruled the Earth.



No they didn't have Pro Tools. The first digital samplers were coming out. The hardware they used to make music looked like ENIAC computer. Edgar Froese of Tangerine Dream died in 2015, but other members have continued making music. Just today got three new CDs. Jean-Michel Jarre keeps making music, for example Oxygene 3 recently. He uses both old and new stuff as far as I know.


----------



## 71 dB

pinnahertz said:


> Whoa!  Chill, man!  I made no comment on your life, I just said you wouldn't understand the difference between free software and a full-on pro DAW without using one.  Just like I wouldn't understand why people spend 100K on a car without test driving one.



Why is it so important to understand the difference between free software and a full-on pro DAW? How does that understanding benefit me?

My Mac is old, too old for DAW I think (HW requirements?). I am learning. I am an idiot with music theory. Scales and chords are difficult stuff for me. Maybe someday I am ready to learn DAW.



pinnahertz said:


> Sure, Jarre created Equinoxe without a DAW, but I'm fairly sure he used one on Aero (it's a 5.1 mix) and the included reboots of Oxegene.



Jarre is one of the best selling artists off electronic music. Of course he can spend hundreds of thousands of dollars on top notch hardware and software. Anything less would be silly.


----------



## bigshot

ProTools is a recording device, not a musical instrument. The equivalent of ProTools back then would be this kind of thing...


----------



## pinnahertz

71 dB said:


> Why is it so important to understand the difference between free software and a full-on pro DAW? How does that understanding benefit me?


You're the one claiming Audacity is so fantastic.  I'm just saying, it's great for free, it's frustrating for anyone doing a serious project.  And nobody has to be wealthy to afford great real DAW tools.  But also, you can't compare the free stuff vs the paid pro stuff unless you actually experience it.  Its just a matter of know of what you speak. 


71 dB said:


> My Mac is old, too old for DAW I think (HW requirements?).


Well, I run Audition on a 2008 Mac Pro under 10.7.5.  Runs fine.  I had ProTools "Free" 15 years ago on a G4, ran just fine (no longer available, but neither is the G4). I think if you wanted to, you could get better software that would run just fine on any Mac running today.  If it'll run iMovie, it'll run a DAW up to at least 24 channels.  You might not get the latest version, but you could get something that works.  Part of a real DAW is the ability to interface with a control surface.  Those would run you a bit of money, but even they are within the means of anyone serious about recording and producing music.  


71 dB said:


> Jarre is one of the best selling artists off electronic music. Of course he can spend hundreds of thousands of dollars on top notch hardware and software. Anything less would be silly.


I'm not the one who suggested he didn't use a DAW.


----------



## bigshot

I thought Jarre was dead. I must have been thinking of Mike Oldfield.


----------



## pinnahertz

bigshot said:


> ProTools is a recording device, not a musical instrument. The equivalent of ProTools back then would be this kind of thing...


Oooh!  I'll bring the denatured alcohol, you bring the swabs!


----------



## pinnahertz

bigshot said:


> I thought Jarre was dead. I must have been thinking of Mike Oldfield.


Nope, both are alive.  You must be thinking of Abe Vigoda.


----------



## Darren G

Transducers do an utterly crap job of trying to keep up with square waves at 30hz, and once you see why (that fricken mass thing, and needing energy to accelerate/decelerate mass), it becomes clear why it's nervousa to fret over ultrasonic frequency reproduction.  Any portion of those frequencies that are below the maximum frequencies we can hear are already captured, and any portion after that, well my ears I'm sure can't hear it


----------



## old tech

bigshot said:


> If the inaudible was audible, how would recording the inaudible affect the audible?



I meant in regard to the claim made by some audiophiles that these inaudible frequencies affect frequencies that are audible.  It is not a claim I buy into, I was just pointing out if it did, it is not relevant to a recording/playback.


----------



## old tech

Dobrescu George said:


> Okay, I think I got it
> 
> I actually have only CDs, I mean ONLY CDs right now, I was curious if it is worth to try to move to Hi-Res. Seems not so much



This is where many audiophiles get led astray.  The quality of the recording and production/mastering is far more important than the media (to an extent, we aren't talking about 8 track cartridges).

Sometimes the best sound is on hi res media, not because of hi res per se but because of better mastering tailored to a more home listening environment.  Other times the best sound of a particular album is on CD, sometimes even LP.  For serious listeners it pays to have playback equipment of different formats and choose the format based on the mastering rather than the other way round.


----------



## gregorio

71 dB said:


> Why is it so important to understand the difference between free software and a full-on pro DAW? How does that understanding benefit me?



Well for starters, it would stop you making incorrect assumptions, posting them publicly on a science forum and then posting inapplicable equations and explanations to defend those incorrect assumptions.



bigshot said:


> I thought Jarre was dead.



He is but that's Maurice Jarre, the 3 time oscar winning film score composer, Jean-Michel's father.

G


----------



## 71 dB

pinnahertz said:


> You're the one claiming Audacity is so fantastic.  I'm just saying, it's great for free, it's frustrating for anyone doing a serious project.  And nobody has to be wealthy to afford great real DAW tools.  But also, you can't compare the free stuff vs the paid pro stuff unless you actually experience it.  Its just a matter of know of what you speak.



Audacity is a good free software and I have made it even better for my needs by writing nyquist plugins for it. Frustrating? What? Defending myself from your attacks is frustrating, not working with Audacity.

What's the point of joining a discussion if you know everything? I have learned something about DAWs here, so the time hasn't been wasted. Perhaps someone else following our discussion has also learned? I am not a teen anyone thinking I know almost everything. Life has humbled me. When someone points out I am wrong I simply thank them for correcting me because I have learned that people are quite wrong about many things most of the time. There it too much knowledge in the world for us. If you use your time learning about DAWs, you have less time learning about sharks or black holes.



pinnahertz said:


> Well, I run Audition on a 2008 Mac Pro under 10.7.5.  Runs fine.  I had ProTools "Free" 15 years ago on a G4, ran just fine (no longer available, but neither is the G4). I think if you wanted to, you could get better software that would run just fine on any Mac running today.  If it'll run iMovie, it'll run a DAW up to at least 24 channels.  You might not get the latest version, but you could get something that works.  Part of a real DAW is the ability to interface with a control surface.  Those would run you a bit of money, but even they are within the means of anyone serious about recording and producing music.



Ok, thanks for the info.



gregorio said:


> Well for starters, it would stop you making incorrect assumptions, posting them publicly on a science forum and then posting inapplicable equations and explanations to defend those incorrect assumptions.



I have to express my incorrect assumptions so people like you can correct me.

My equations are correct, just not applicable for modern DAWs. I'm sure DAWs haven't been 64 bit always, so maybe 25 years ago my equations were applicable? I stopped defending my assumptions the moment I realised they are incorrect.


----------



## gregorio

71 dB said:


> [1] I have to express my incorrect assumptions so people like you can correct me. ... I stopped defending my assumptions the moment I realised they are incorrect.
> [2] I'm sure DAWs haven't been 64 bit always, so maybe 25 years ago my equations were applicable?



1. Obviously these two statements are contradictory.
2. Your equations would have been applicable if there were any DAWs 25 years ago.

G


----------



## 71 dB

gregorio said:


> 1. Obviously these two statements are contradictory.
> 2. Your equations would have been applicable if there were any DAWs 25 years ago.
> 
> G


1. Thanks again for correcting me  Which statement is false?

2. Were any? When did the first DAW come out? According to Wikipedia, The first Pro Tools came out January 20, 1989; almost 29 years ago.

https://en.wikipedia.org/wiki/Pro_Tools


----------



## Darren G (Oct 13, 2017)

I admit, I am open to the value of taking more samples over time, if that helps hardware manufacturers re-construct the original waveform in hardware or software, but the bit-depth is purely about dynamic range.  16 bits is ~96 some db, 24 bits is some 140 db (sorry don't have the actual correct numbers on hand, but it's irrelvant to my point).

Think about that.  Nobody listens at a dynamic range of 140 db.  We are talking hearing loss (and pain!) at that absolute level, or at a more listenable range, suppressing noise that is above what is achievable, or even if someone could if cost was no object, what is the point?  Go to a live concert.  The noise levels of the crowd alone would drown out any low level sounds beyond a certain dynamic range.

I won't even bother with our brains/ears already mask low level details when processing louder sounds.  The point doesn't change.


----------



## pinnahertz

71 dB said:


> Audacity is a good free software and I have made it even better for my needs by writing nyquist plugins for it. Frustrating? What? Defending myself from your attacks is frustrating, not working with Audacity.


It's all about point of view.  Each tool has its limits.  If Audacity were on par with ProTools, ProTools would have died years ago and everyone would be using the free tool.  

You have a very personal view of my posts: that they are attacking you.  I don't expect you to appreciate them, but this isn't a private message either, it's public, and others might appreciate more information.


71 dB said:


> What's the point of joining a discussion if you know everything?


I don't, never claim to.  I do share what I do know, hopefully someone else may benefit from my life long learning process.  


71 dB said:


> I have learned something about DAWs here, so the time hasn't been wasted. Perhaps someone else following our discussion has also learned? I am not a teen anyone thinking I know almost everything. Life has humbled me. When someone points out I am wrong I simply thank them for correcting me because I have learned that people are quite wrong about many things most of the time. There it too much knowledge in the world for us. If you use your time learning about DAWs, you have less time learning about sharks or black holes.
> 
> I have to express my incorrect assumptions so people like you can correct me.


That's a very fine humble attitude.  And I'm sure we are all far more likely to run into a DAW, or Audacity, in our lives that run into a shark or black hole.


----------



## gregorio

71 dB said:


> 2. Were any? When did the first DAW come out? According to Wikipedia, The first Pro Tools came out January 20, 1989; almost 29 years ago.



I suppose that depends on how you define Digital Audio Workstation. Pro Tools was first released in 1991 but only offered 4 tracks and was little more than a non-destructive editor rather than a DAW. It wasn't until 20 years ago that it developed to the point of being a true DAW and at that time it's mix engine was updated to essentially 48bit.

G


----------



## pinnahertz

71 dB said:


> 1. Thanks again for correcting me  Which statement is false?
> 
> 2. Were any? When did the first DAW come out? According to Wikipedia, The first Pro Tools came out January 20, 1989; almost 29 years ago.
> 
> https://en.wikipedia.org/wiki/Pro_Tools


Yes, and it was immediately nicknamed "SlowTools".  A simple 1 second crossfade took like 30 seconds to "render".  All processing was in the computer.  Other systems of the time used external DSP boxes to speed things along, and ProTools got there too eventually with DSP Farm cards.  I'm not sure of the internal bit depth, but it was already known that DSP on 16 bit audio needed more bits to work well.  These were the days of dedicated hardware DAWs, it was the only way to keep up productivity, and they were out of reach of anyone but well financed studios.


----------



## 71 dB

Darren G said:


> I admit, I am open to the value of taking more samples over time, if that helps hardware manufacturers re-construct the original waveform in hardware or software...



All you need to help reconstruction process is to oversample original 44.1 kHz to 88.2 kHz and the reconstruction filter requirements become much more relaxed.



Darren G said:


> ..., but the bit-depth is purely about dynamic range.  16 bits is ~96 some db, 24 bits is some 140 db (sorry don't have the actual correct numbers on hand, but it's irrelvant to my point).



The formula for dynamic range (without dither) of N bit data is 20*log (2^N * sqrt(3/2)) = 6.0206*N + 1.761.

16 bit => DR = 98.09 dB
24 bit => DR = 146.26 dB
32 bit => DR = 194.45 dB



Darren G said:


> Think about that.  Nobody listens at a dynamic range of 140 db.  We are talking hearing loss (and pain!) at that absolute level, or at a more listenable range, suppressing noise that is above what is achievable, or even if someone could if cost was no object, what is the point?  Go to a live concert.  The noise levels of the crowd alone would drown out any low level sounds beyond a certain dynamic range.



There's not even amplifiers with such a low noise floor to be able to achieve this kind of dynamic range. The A-weighed background noise of a quiet living room is about 30 dB and the loudest peaks of real symphony orchestra is about 110 dB, so in order to reproduce that dynamic range you need about (110-30) dB = 80 dB of dynamic range. So, (optimally used) 14 bits is pretty much how much is needed. On CD there's 2 more bits to increase the dynamic range by 12 dB so we are good. Dither increases _usable_ dynamic range of CD about 10-20 dB (because the signal doesn't granulate (doesn't modulate quantization noise), it remains "pure", just accompanied by the dither noise, with can be made very quiet for human hearing by noise shaping. This means you can reproduce sounds at level 100 dBFS and below!). 

Don't believe? Try it yourself on Audacity: Generate a 1000 Hz test tone at level 0.00001 (-100 dBFS) at 32 bits and transfer it to 16 making sure you have noise shaping dither enabled (Preferences / Quality / High-Quality Conversion / Dither / Shaped). The spectrum of the result looks like this:




 

You see the 1 kHz test tone at level -100 dBFs + shaped dither noise. If you amplify the signal by 50 dB you can hear the test tone at level -50 dBFS + high frequency hiss (dither). Yes, the noise level is higher than the test tone, but at very high frequencies so that the test tone is not only audible, but very distortion-free considering we are operating at levels below the dynamic range of 16 bit audio.


----------



## Darren G (Oct 14, 2017)

71 dB said:


> _*There's not even amplifiers with such a low noise floor to be able to achieve this kind of dynamic range. *_



Exactly


----------



## TheSonicTruth

gregorio said:


> It seems to me that there is a lot of misunderstanding regarding what bit depth is and how it works in digital audio. This misunderstanding exists not only in the consumer and audiophile worlds but also in some education establishments and even some professionals. This misunderstanding comes from supposition of how digital audio works rather than how it actually works. It's easy to see in a photograph the difference between a low bit depth image and one with a higher bit depth, so it's logical to suppose that higher bit depths in audio also means better quality. This supposition is further enforced by the fact that the term 'resolution' is often applied to bit depth and obviously more resolution means higher quality. So 24bit is Hi-Rez audio and 24bit contains more data, therefore higher resolution and better quality. All completely logical supposition but I'm afraid this supposition is not entirely in line with the actual facts of how digital audio works. I'll try to explain:
> 
> When recording, an Analogue to Digital Converter (ADC) reads the incoming analogue waveform and measures it so many times a second (1*). In the case of CD there are 44,100 measurements made per second (the sampling frequency). These measurements are stored in the digital domain in the form of computer bits. The more bits we use, the more accurately we can measure the analogue waveform. This is because each bit can only store two values (0 or 1), to get more values we do the same with bits as we do in normal counting. IE. Once we get to 9, we have to add another column (the tens column) and we can keep adding columns add infinitum for 100s, 1000s, 10000s, etc. The exact same is true for bits but because we only have two values per bit (rather than 10) we need more columns, each column (or additional bit) doubles the number of vaules we have available. IE. 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024 .... If these numbers appear a little familiar it is because all computer technology is based on bits so these numbers crop up all over the place. In the case of 16bit we have roughly 65,000 different values available. The problem is that an analogue waveform is constantly varying. No matter how many times a second we measure the waveform or how many bits we use to store the measurement, there are always going to be errors. These errors in quantifying the value of a constantly changing waveform are called quantisation errors. Quantisation errors are bad, they cause distortion in the waveform when we convert back to analogue and listen to it.
> 
> ...



Agreed!  If 24bit and 16bit versions of the exact same recordings were simultaneously published, the average consumer would be hard-pressed to hear a difference.  So why do some people allege to hear a difference?  The mastering!  Something must be done to make it sound 'different' enough for it to sell.   And typically it is the same compress-brickwall-limit job that was done during 'remastering' of earlier 80s CD releases.  

Buyer beware!


----------



## old tech

TheSonicTruth said:


> Agreed!  If 24bit and 16bit versions of the exact same recordings were simultaneously published, the average consumer would be hard-pressed to hear a difference.  So why do some people allege to hear a difference?  The mastering!  Something must be done to make it sound 'different' enough for it to sell.   And typically it is the same compress-brickwall-limit job that was done during 'remastering' of earlier 80s CD releases.
> 
> Buyer beware!


Apart from that, is the recording truly hi res?  If the recording originally was sourced from an analog tape or 16/44 master (ie most pre 1990 albums) then it is only the container that is hi res, not the actual music.

The other question is why does every hi res player have a light or some sort of indicator to let the listener know it is a hi res file being played?  Surely if the difference is real, the listener wouldn't need the indicator.  However, the indicator is a very important component for the placebo effect.


----------



## gregorio

TheSonicTruth said:


> If 24bit and 16bit versions of the exact same recordings were simultaneously published, the average consumer would be hard-pressed to hear a difference.



In my OP I tried to explain why the average (and any other) consumer would not just be "hard-pressed" to hear a difference but that it would be impossible to hear a difference.

G


----------



## Darren G

gregorio said:


> In my OP I tried to explain why the average (and any other) consumer would not just be "hard-pressed" to hear a difference but that it would be impossible to hear a difference.
> 
> G



For whatever it's worth, I really enjoyed reading your post on bit-depth and dynamic range.  Very well explained.


----------



## 71 dB

old tech said:


> The other question is why does every hi res player have a light or some sort of indicator to let the listener know it is a hi res file being played?  Surely if the difference is real, the listener wouldn't need the indicator.  However, the indicator is a very important component for the placebo effect.



The light indicates that the placebo effect is turned on. 

If you don't know better it's healthty to assume 90 % of heard differences are caused by placebo effect. Often plabeco is responsible of 99-100 % of the differences (these totally disappear in proper blind tests), but _sometimes_ differences in audio are real.


----------



## old tech

71 dB said:


> The light indicates that the placebo effect is turned on.
> 
> If you don't know better it's healthty to assume 90 % of heard differences are caused by placebo effect. Often plabeco is responsible of 99-100 % of the differences (these totally disappear in proper blind tests), but _sometimes_ differences in audio are real.


Yes, sometimes the differences are real.  What amazes though is why people spend thousands on equipment upgrades without even doing basic due diligence, not realising that their initial excitement may be clouding perceptions.


----------



## gregorio

71 dB said:


> If you don't know better it's healthty to assume 90 % of heard differences are caused by placebo effect. Often plabeco is responsible of 99-100 % of the differences (these totally disappear in proper blind tests), but _sometimes_ differences in audio are real.



Except in the case of a deliberate fake, the differences in the audio are always real. That's not the issue, the issue is; are those differences audible? Is random noise 30-1000 times lower than the noise floor of the recording itself audible? Or, in the case of sample rates, is relatively small amounts of ultrasonic content audible in the presence of much larger amounts of content in the audible band? In the case of bit depth, it would require the recording to be reproduced at ear damaging levels for the difference to be even theoretically audible. In the case of higher than standard sample rates, logic and ALL the considerable reliable evidence obtained over many decades indicates that it is not audible and hence why we have the term "ultrasonic" in the first place.

It is therefore unhealthy to assume anything other than 100% perception bias (placebo effect).

G


----------



## 71 dB

old tech said:


> Yes, sometimes the differences are real.  What amazes though is why people spend thousands on equipment upgrades without even doing basic due diligence, not realising that their initial* excitement* may be clouding perceptions.



You kind of answered your own question. People want excitement.



gregorio said:


> Except in the case of a deliberate fake, the differences in the audio are always real. That's not the issue, the issue is; are those differences audible? Is random noise 30-1000 times lower than the noise floor of the recording itself audible? Or, in the case of sample rates, is relatively small amounts of ultrasonic content audible in the presence of much larger amounts of content in the audible band? In the case of bit depth, it would require the recording to be reproduced at ear damaging levels for the difference to be even theoretically audible. In the case of higher than standard sample rates, logic and ALL the considerable reliable evidence obtained over many decades indicates that it is not audible and hence why we have the term "ultrasonic" in the first place.
> 
> It is therefore unhealthy to assume anything other than 100% perception bias (placebo effect).
> 
> G



I hear you grecorio, but I was actually talking about differences in audio in general, not only bits and sample frequencies. These are in no way exact numbers, I tend to think like this:

5 % of differences people experience in audio are real
5 % of differences people experience in audio are real, but emphasized by placebo effect
90 % of differences people experience in audio are pure placebo and disappear in blind tests

"HI-RES" audio exploits that 90 % heavily. For golden ears 90 % is maybe "only" 70-80 %

Again, these numbers are not accurate nor do they come from any science studies. I just need "some reasonable" numbers to model the world and other people can use other numbers if they feel like it. Many people deny the existence of placebo effect and their "placebo profile" is 100-0-0 instead of 5-5-90 or 8-12-80. In Star Wars force is always with you. In real life placebo effect is always with us.


----------



## gregorio

71 dB said:


> I hear you grecorio, but I was actually talking about differences in audio in general, not only bits and sample frequencies. These are in no way exact numbers, I tend to think like this:
> 5 % of differences people experience in audio are real
> 5 % of differences people experience in audio are real, but emphasized by placebo effect
> 90 % of differences people experience in audio are pure placebo and disappear in blind tests ...



I personally don't find such a huge generalisation useful, as it's actually misleading in many specific cases. In the case of 16 vs 24bit for example, or even hi-res in general. Where are those 5 or 10%, why have we never encountered them in any of the countless formal DB tests performed over many years? There's no reliable evidence (of which I'm aware) of even a single person being able to tell a difference, let alone 5 or 10%!

G


----------



## bigshot

I tend to think that it's more a matter of thinking process than it is a statistical figure. Everyone who employs logical problem solving techniques get good results. Everyone who doesn't gets randomized results.


----------



## 71 dB

gregorio said:


> I personally don't find such a huge generalisation useful, as it's actually misleading in many specific cases. In the case of 16 vs 24bit for example, or even hi-res in general. Where are those 5 or 10%, why have we never encountered them in any of the countless formal DB tests performed over many years? There's no reliable evidence (of which I'm aware) of even a single person being able to tell a difference, let alone 5 or 10%!
> 
> G



Maybe I wasn't clear enough: 16 vs 24bit is pretty much all in the 90 % while differences among loudspeakers "occupy" the 5 %. So, whenever you think you hear difference you should your placebo profile, say 5-5-90 and then think what is you are hearing differences of, is it loudspakers, cables, sample frequences or bit? Then you can go like this: Loudspeakers => yeah, the difference is real. Bits => it's placebo.


----------



## pinnahertz

71 dB said:


> You kind of answered your own question. People want excitement.
> 
> 
> 
> ...


Those are very specific number for being  "in no way exact", "not accurate nor coming from science studies".  

We all have the need for reasonable numbers, but making stuff up doesn't differentiate true science from audiophoolery and marketing hype.  That's what those guys do every day.


----------



## 71 dB

pinnahertz said:


> Those are very specific number for being  "in no way exact", "not accurate nor coming from science studies".
> 
> We all have the need for reasonable numbers, but making stuff up doesn't differentiate true science from audiophoolery and marketing hype.  That's what those guys do every day.



You keep whining no matter what. I made clear these are not specific numbers and still you keep whining! No matter what I say or how I said it you need to whine about it. How about if I say it like this:

Very little of differences people experience in audio are real
Very little of differences people experience in audio are real, but emphasized by placebo effect
Most of differences people experience in audio are pure placebo and disappear in blind tests

What would you whine about that? I am curious...


----------



## castleofargh

97.6% of made up values are BS.  ^_^


----------



## pinnahertz

71 dB said:


> You keep whining no matter what. I made clear these are not specific numbers and still you keep whining! No matter what I say or how I said it you need to whine about it. How about if I say it like this:
> 
> Very little of differences people experience in audio are real
> Very little of differences people experience in audio are real, but emphasized by placebo effect
> ...


Making up a list of numbers to suit your own purpose. That's no different from marketing lies.  Why bother? We  already have plenty of marketing lies to contend with.


----------



## csglinux

castleofargh said:


> 97.6% of made up values are BS.  ^_^



My wife has some shampoo that makes your hair 27.8% silkier. (It really says that on the bottle.) I've always been impressed that any cosmetics company could measure silkiness so accurately


----------



## bigshot

They did a poll of people asking them if they were going to go to heaven. 80% of people said they were. When they asked those people how many other people were going to go to heaven, they said 20%. Math rarely adds up!


----------



## chaos215bar2

That's not math. That's marketing.


----------



## JaeYoon (Oct 17, 2017)

Sony's website is a great place to start checking out marketing too


----------



## TheSonicTruth

JaeYoon said:


> Sony's website is a great place to start checking out marketing too



The monkey at the mastering DAW will have more of an impact on the final sound than that thing!


----------



## bigshot

What part of my body will I "feel every note" in?


----------



## pinnahertz

bigshot said:


> What part of my body will I "feel every note" in?


Your auditory nerve for sure, after that it's a question of SPL. 

But the real question is: do you want to feel every note?  That probably is the way we _should_ listen to Beethoven's 9th...as the creator "heard" it.


----------



## bigshot

Beethoven bit his piano to hear the music, right? Maybe we could come up with an audiophile dental appliance of some sort!


----------



## JaeYoon (Oct 17, 2017)

I love you guys.

Dead.....

Here is an audiophile salesman hat. Anyone wanna get started?

You see purchase this Audiophile Dental Chompers!! Bite into your own stereo system and ensure audio nirvana! But wait!!! Before you touch your keyboard and search google for snake oil!

You see Ludwig Van Beethoven bit into his own piano!! In order to hear how it sounds!
Now you can hear music the way the artist intended it to! Just wait! Call now and receive these cool audiophile certified bumpers. You wouldn't want jitter and EMI and that nasty crosstalk to infiltrate your music now!!

Order now for only...wait for it! Wait for it!
10,000 dollars. Yes! If you order now we can perform surgery and connect these expensive audiophile quality cables right into your chompers as well! Buy now right away!! What are you waiting for!!! Go on!! Throw your money!! Did I forget that these chompers can play back HIGH RESOLUTION 24 bit audio and dsd natively!!! Wow you must be pushing that buy button twice!!

Don't forget to buy an ESD proof audiophile certified case for your chompers too!


----------



## Don Hills

bigshot said:


> Beethoven bit his piano to hear the music, right? Maybe we could come up with an audiophile dental appliance of some sort!



Done that. Clamped the outlet of a PA compression driver between my teeth. Made my eyesight go blurry and my teeth ache.


----------



## ev13wt

JaeYoon said:


> I love you guys.
> 
> Dead.....
> 
> ...







You and me, we are going to make billions.


----------



## RRod

bigshot said:


> Beethoven bit his piano to hear the music, right? Maybe we could come up with an audiophile dental appliance of some sort!


Tuning fork to the teeth does wonders for PRaT!


----------



## OddE

71 dB said:


> The formula for dynamic range (without dither) of N bit data is 20*log (2^N * sqrt(3/2)) = 6.0206*N + 1.761.
> 
> 16 bit => DR = 98.09 dB
> 24 bit => DR = 146.26 dB
> ...



-I am a bit late to the party, but I'd just like to chime in that the Benchmark AHB2 is getting close enough for all practical purposes (and then some!) - they claim THD+N at -118dB or so at rated output. (Which is where you'd need to be at in order to take advantage of the offered dynamic range.)

Obviously, you'd also be in severe pain and your ears wouldn't be anywhere near capable of relaying said range to your brain - but it would be there. (nevermind the horrific distortion introduced by any transducer driven that hard; again, you'd have other priorities. Like shutting the system down, for instance.)


----------



## Cutestudio

71 dB said:


> Dither increases _usable_ dynamic range of CD about 10-20 dB (because the signal doesn't granulate (doesn't modulate quantization noise), it remains "pure", just accompanied by the dither noise, with can be made very quiet for human hearing by noise shaping. This means you can reproduce sounds at level 100 dBFS and below!).


The original article is faulty in it's primary premise, something I have pointed out before to the blind.
Dither is a _statistical_ method and while your statement is superficially correct there is an important caveat: We are still representing the data in 16bit digital so the quantisation errors are _still present_.

Logically then it means that any transient HF shape is still locked to those quantised steps, and therefore _still the wrong shape_. As a tone it's not an issue (it's a _statistical_ method), but as a one off, non repeated shape, it's _still_ quantised to the wrong shape. The question of whether this creates an audible effect can be addressed by the question: "Can we heard the difference between no dither and (the various types of) dither?" If we can, we _must_ be able to also hear the error. My next point however makes clear that this error is also irrelevant.



71 dB said:


> The A-weighed background noise of a quiet living room is about 30 dB and the loudest peaks of real symphony orchestra is about 110 dB, so in order to reproduce that dynamic range you need about (110-30) dB = 80 dB of dynamic range. So, (optimally used) 14 bits is pretty much how much is needed. On CD there's 2 more bits to increase the dynamic range by 12 dB so we are good.



A good post but this is where theory and practice diverge. This thread is about the LSB, bit0, and the worries that swapping that bit for 8 bits (i.e. going to 24bit) may be required or not: i.e. it's about the _details_ of the _tiniest_ sounds. This theory is inapplicable to modern pop CDs however because _they are all mastered in 15 bits, having thrown away bit 15, the MSB_.
There's even a _very_ good chance that those loudest peaks of a symphony orchestra will be simply cut off.

So while the discussion of dither and the misunderstanding of it's limitations is indeed interesting; as a subject I feel it's rather irrelevant to modern digital music. 286 pages on the LSB when the MSB went a long time ago: i.e. we've got a bigger problem that 16 bit dither at -96dB, a problem that's almost exactly +6dB in size.

Interestingly I find most of life is like this, the elephant in the room is usually quite invisible. Today there are many elephants.


----------



## gregorio

Cutestudio said:


> [1]The original article is faulty in it's primary premise, something I have pointed out before to the blind.
> Dither is a _statistical_ method and while your statement is superficially correct there is an important caveat: We are still representing the data in 16bit digital so the quantisation errors are _still present_.
> [1a] Logically then it means that ...
> [2] This theory is inapplicable to modern pop CDs however because _they are all mastered in 15 bits, having thrown away bit 15, the MSB_.



1. Unfortunately, you appear to have completely misunderstood how dither works and it's purpose. After the correct application of dither there is NO quantisation error, nada, none whatsoever, response is perfectly linear down to the digital noise floor.
1a. "Logically then", the rest of what you say on this matter is nonsense because it's based on your misunderstanding of digital audio/dither. There are no quantisation steps in the waveform after conversion and it is NOT the wrong shape, regardless of whether it's a transient or any other waveform shape!
BTW, it's not a good idea to call others "blind" when you yourself obviously have a poor grasp of the issue!

2. I've got no idea where this assertion comes from, it's complete nonsense but at least it's original complete nonsense, I've never heard this one before. Where did you get the idea that the MSB is "thrown away" during the mastering process?

G


----------



## 71 dB

Cutestudio said:


> The original article is faulty in it's primary premise, something I have pointed out before to the blind.
> Dither is a _statistical_ method and while your statement is superficially correct there is an important caveat: We are still representing the data in 16bit digital so the quantisation errors are _still present_.



Dithering breaks correlation between the signal and noise. It means errors can't be appointed to the signal, but they form dither noise which is a separate entity of the signal. Quantization errors mean noise that correlates with the signal. When you remove the correlation, quantization noise becomes, well just noise present with the signal.



Cutestudio said:


> Logically then it means that any transient HF shape is still locked to those quantised steps, and therefore _still the wrong shape_. As a tone it's not an issue (it's a _statistical_ method), but as a one off, non repeated shape, it's _still_ quantised to the wrong shape. The question of whether this creates an audible effect can be addressed by the question: "Can we heard the difference between no dither and (the various types of) dither?" If we can, we _must_ be able to also hear the error. My next point however makes clear that this error is also irrelevant.



Well, yes. Signal + noise is not the same as signal alone meaning we have "wrong shape". To correct this you need infinite bits, infinite dynamic range in theory. In real life worrying about wrong shapes at this scale  (16 bit) is meaningless.






Look at the magnitude spectrum above: I genereted 1000 Hz sinusoid, level -120 dB (amplitude 0.000001) at 32 bit mode. Then I made it 16 bit using shaped dither noise and the spectrum of that 16 bit signal is above! The 1000 Hz sinusoid is clearly there. If I amplify this signal by 50 dB, I can listen to it and hear (just) the sinusoid behind the dither noise. The sinusoid sounds pure, because it is very very pure. What in this spectrum shows quantization errors? Quantization errors (no dither) look like this:



 

I Generated 1000 Hz sinusoid, level -80 dB at 32 bit. Then truncated it to 16 bit without dither. This is pure quantization error. You see 3000 Hz, 5000 Hz, 7000 Hz etc. (odd harmonics) dominating as you would expect. Below the same sinusoid at 16 bit with triangle dithering:





The noise _power_ is similar to the non-dither case, but the specrum shape is very different, because the dither noise doesn't correlate with the signal (that would cause harmonics).



Cutestudio said:


> A good post but this is where theory and practice diverge. This thread is about the LSB, bit0, and the worries that swapping that bit for 8 bits (i.e. going to 24bit) may be required or not: i.e. it's about the _details_ of the _tiniest_ sounds. This theory is inapplicable to modern pop CDs however because _they are all mastered in 15 bits, having thrown away bit 15, the MSB_.



Huh? Are you talking about loudness war?



Cutestudio said:


> There's even a _very_ good chance that those loudest peaks of a symphony orchestra will be simply cut off.



Not on the CDs of orchestral music I have. Stop buying bootleg crap. 



Cutestudio said:


> So while the discussion of dither and the misunderstanding of it's limitations is indeed interesting; as a subject I feel it's rather irrelevant to modern digital music. 286 pages on the LSB when the MSB went a long time ago: i.e. we've got a bigger problem that 16 bit dither at -96dB, a problem that's almost exactly +6dB in size.



I wonder who has demonstrated most misunderstanding of dither? MSB = + 6 dB of dynamic range. LSB = + 6 dB of dynamic range. They are equally important.


----------



## TheSonicTruth

71 dB said:


> Dithering breaks correlation between the signal and noise. It means errors can't be appointed to the signal, but they form dither noise which is a separate entity of the signal. Quantization errors mean noise that correlates with the signal. When you remove the correlation, quantization noise becomes, well just noise present with the signal.
> 
> 
> 
> ...



"
↑
There's even a _very_ good chance that those loudest peaks of a symphony orchestra will be simply cut off.
Not on the CDs of orchestral music I have. Stop buying bootleg crap. "

He may be right.  My classical CDs are all original albums released in the mid to late 1980s, yet they all have rock DR values: DR11-13.

I *know* that a typical symphony sitting features much greater actual DR swings than what was put on my CDs.  *Something* was done to them!


----------



## bigshot (Oct 18, 2017)

When people speak of the dynamic range of a symphony orchestra, they're talking about how loud it is from a close perspective. Put your ear right next to a trumpet or tympani and you'll hear some pretty loud sound. But in practice, the louder instruments are arranged further back and the recording is made from a little distance to get a more listenable perspective. The actual dynamics of an orchestral recording is in the same range as any other dynamic music- 35-40dB max. It isn't comfortable to listen to music with wildly exaggerated dynamics.

I have one surround sound orchestral recording that has the stupidest dynamics I've ever heard. It's a violin concerto, and they arrayed the band all around the conductor in a circle and miked the conductor's perspective. Someone had the bright idea of placing all the instruments at the same distance from the mic. They put the tympani right next to the violin soloist, so every once in a while there's a massive kettle drum hit that blasts you out of your chair. Unlistenable.


----------



## TheSonicTruth

bigshot said:


> When people speak of the dynamic range of a symphony orchestra, they're talking about how loud it is from a close perspective. Put your ear right next to a trumpet or tympani and you'll hear some pretty loud sound. But in practice, the louder instruments are arranged further back and the recording is made from a little distance to get a more listenable perspective. The actual dynamics of an orchestral recording is in the same range as any other dynamic music- 35-40dB max. It isn't comfortable to listen to music with wildly exaggerated dynamics.
> 
> I have one surround sound orchestral recording that has the stupidest dynamics I've ever heard. It's a violin concerto, and they arrayed the band all around the conductor in a circle and miked the conductor's perspective. Someone had the bright idea of placing all the instruments at the same distance from the mic. They put the tympani right next to the violin soloist, so every once in a while there's a massive kettle drum hit that blasts you out of your chair. Unlistenable.



  Technique counts for something! lol


----------



## RRod

TheSonicTruth said:


> "
> He may be right.  My classical CDs are all original albums released in the mid to late 1980s, yet they all have rock DR values: DR11-13.
> 
> I *know* that a typical symphony sitting features much greater actual DR swings than what was put on my CDs.  *Something* was done to them!



And here's a 1979 classical disc (Soundstream recording) with DR15. And here's a 1986 recording with a track with DR20, but of course DR is not really a dynamic range measure since you can add material and have the DR go down…


----------



## pinnahertz

RRod said:


> And here's a 1979 classical disc (Soundstream recording) with DR15. And here's a 1986 recording with a track with DR20, but of course DR is not really a dynamic range measure since you can add material and have the DR go down…


The DR meter is a tool for measuring a very specific case of dynamic range, short term dynamic range that is strongly affected by loudness war processing. But it does not measure total dynamic range, that's not it's function..


----------



## castleofargh

pinnahertz said:


> The DR meter is a tool for measuring a very specific case of dynamic range, short term dynamic range that is strongly affected by loudness war processing. But it does not measure total dynamic range, that's not it's function..


you're telling it to the wrong guy ^_^.


----------



## pinnahertz

castleofargh said:


> you're telling it to the wrong guy ^_^.


I know, but I'm also aware that he's not the only one here.


----------



## TheSonicTruth

pinnahertz said:


> I know, but I'm also aware that he's not the only one here.




And what does he mean by "wrong guy"?


----------



## castleofargh

you, I mean you of course. you're the one who came arguing against dynamic compression with the DR meter while giving a wrong explanation of what it does. the post right after cleared everything there was to clear, but it still took many more to get you to just consider that maybe you needed another tool. 
and yet, here we are. you clearly comparing the dynamic of a orchestra IRL, and a DR meter number on CDs. making me feel that you haven't listened to anything we discussed about the DR meter tool just a few days ago.


----------



## PETEBULL (Nov 12, 2017)

audiosampling said:


> *16bit vs 8bit, *the myth exploded (again) !
> 
> http://www.audiocheck.net/blindtests_16vs8bit.php


Damn it! You can have FLAC the size of MP3! Freaking EPIC!!!
Oh wait. Sometimes you get artifacting ("herding_calls" sample)  but not on brickwalled audio.
Yeah, I hear some noise on Daft Punk - Instant Crush. That "gangam style" is way too brickwalled and thus fraudy.
Upd. Using dithering can fix that noise  though it occupies ~100 Kbps more. Still the noise can be an audiocassette fetish for some.


----------



## amirm

gregorio said:


> Not unless you can tell the difference between white noise and white noise that is well below the noise floor of your listening environment!!


Sorry for going back to the first post in this thread.  I just got here .  But turns out we can absolutely do that.  Noise in your recording comes from your speakers, i.e. point source.  Noise that is in the environment in your room is diffused all around you.  Putting aside the important bit that the spectrum of noise in our living room is anything but "white" (it is heavily biased towards low frequencies because walls and doors don't filter it as much), our hearing system is capable of distinguishing between point noise sources and diffused one.  Think of the ancient man needing to hear a dangerous animal coming at them, making noises, over the noise of the environment.  With two ears and a brain, we can distinguish between them because what each ear hears is different and that differential lets us identify the noise and its direction.

This has been researched with controlled testing and published in the Journal of Audio Engineering Society: _Dynamic-Range Issues in the Modern Digital Audio Environment, ” _Fielder, Louis D., JAES Volume 43 Issue 5 pp. 322-339; May 1995






So in summary, listeners could detect white noise coming out of speakers at levels ranging from minus 2 to plus 9 db SPL whereas the environmental noise was at 20 to 35 dbA SPL.

This also undermines the thesis of the original post of taking average room noise SPL, adding it to CD dynamic range and such to say we don't need that kind of dynamic range.  That math may make sense in our stomach, but it does not when you look at the science of it (for more see https://audiosciencereview.com/forum/index.php?threads/dynamic-range-how-quiet-is-quiet.14/).


----------



## bigshot

A listening room with a 20 to 30dB noise floor is quieter than any listening room in a home. That's like a recording booth, not a listening room.


----------



## castleofargh

amirm said:


> Sorry for going back to the first post in this thread.  I just got here .  But turns out we can absolutely do that.  Noise in your recording comes from your speakers, i.e. point source.  Noise that is in the environment in your room is diffused all around you.  Putting aside the important bit that the spectrum of noise in our living room is anything but "white" (it is heavily biased towards low frequencies because walls and doors don't filter it as much), our hearing system is capable of distinguishing between point noise sources and diffused one.  Think of the ancient man needing to hear a dangerous animal coming at them, making noises, over the noise of the environment.  With two ears and a brain, we can distinguish between them because what each ear hears is different and that differential lets us identify the noise and its direction.
> 
> This has been researched with controlled testing and published in the Journal of Audio Engineering Society: _Dynamic-Range Issues in the Modern Digital Audio Environment, ” _Fielder, Louis D., JAES Volume 43 Issue 5 pp. 322-339; May 1995
> 
> ...



while I agree that simply adding the noise of the room to the music's dynamic is an oversimplified idea. countering it with audibility measured for white noise where the room's own noise floor is now the loudest signal, that's IMO even more inadequate to depict realistic listening or listening needs. 
sure enough noise isn't some opaque paint spread over the stuff below, but masking is still very much a reality, so is the human's limited instantaneous dynamic range.


----------



## bigshot

The easiest way to tell how much dynamic range is enough is to sit in your living room, put on a CD and turn the volume up to the loudest comfortable volume and listen to see if you hear the noise floor of the CD. (hint: you won't)


----------



## csglinux

bigshot said:


> The easiest way to tell how much dynamic range is enough is to sit in your living room, put on a CD and turn the volume up to the loudest comfortable volume and listen to see if you hear the noise floor of the CD. (hint: you won't)



One day, with the right equipment, and the right recording, and an appropriate period of (what should be) silence in the recording, you might.

I posted a link a while back to a video interview with Rob Watts where he claims noise shapers at -130, -140 dB made an audible difference. He thinks subtle cues from the multiple reverberations in a large cathedral were responsible for the improved depth perception of a pipe organ recording. Now, I've not heard those demos myself, so maybe all this is just marketing. I'm all for a healthy dose of skepticism. Is there another plausible explanation?


----------



## RRod (Nov 13, 2017)

With my PM-3s on, which have pretty good isolation in the meaty audible range, I could hear white noise with -110dBFS peak / -114 dBFS RMS pretty readily in my not too-too quiet work environment. This is with the pot set for the highest I set it for music. Any actual sound of course swamps it immediately, including the piped-in white noise for work ^_^


----------



## castleofargh

csglinux said:


> One day, with the right equipment, and the right recording, and an appropriate period of (what should be) silence in the recording, you might.
> 
> I posted a link a while back to a video interview with Rob Watts where he claims noise shapers at -130, -140 dB made an audible difference. He thinks subtle cues from the multiple reverberations in a large cathedral were responsible for the improved depth perception of a pipe organ recording. Now, I've not heard those demos myself, so maybe all this is just marketing. I'm all for a healthy dose of skepticism. Is there another plausible explanation?


the most plausible explanation without some hyper specific conditions, would be that he's wrong, obviously. 
  maybe you can abuse noise shaping at -130dB and move it all to some high level energy in the ultrasounds. rand maybe then because of a different issue and bad filtering you can end up making some audible IMD or whatever. maybe what he calls -130dB is in fact a loud signal on some totally silent track with the loudest stuff already at -70dB? IDK what he tested, or how, or what else he measured before deciding that it was fine to suggest some silly hypothesis. 
in short, the claim needs information about the test related to it. else it means nothing at all. and the hypothesis is only that, no reason to take it more seriously than any other idea anybody else could have.


----------



## csglinux

Except something like this could be easily disputed by anybody else doing the same listening tests. I may be wrong, but I doubt Rob would throw his (fairly considerable) reputation under the bus like that.


----------



## bigshot

csglinux said:


> One day, with the right equipment, and the right recording, and an appropriate period of (what should be) silence in the recording, you might.



Anything can happen in theory. But in practice, I think saying that I would never need more than 16 bits is a VERY safe bet. I've had CDs for decades and I haven't run into that sort of situation yet.


----------



## csglinux

bigshot said:


> Anything can happen in theory. But in practice, I think saying that I would never need more than 16 bits is a VERY safe bet. I've had CDs for decades and I haven't run into that sort of situation yet.



The question is - how much better can recorded music playback possibly get? We won't know until we get there, but we still seem to be making incremental improvements, even when the specs say we shouldn't be able to hear a difference. E.g., look at the specs of a Chord Hugo 2. Then go A/B it with a Chord Dave. There's something going on there I don't understand.


----------



## bigshot

csglinux said:


> The question is - how much better can recorded music playback possibly get?



Equipment can be improved infinitely I suppose. But human ears aren't likely to evolve to be able to hear better in our lifetimes. Since digital audio is capable of perfectly reproducing sound for human ears, we're sitting smack dab in the middle of audio nirvana. But there are still going to be people who think they need better than that.


----------



## csglinux

Leaving aside the obvious improvements in speakers and headphones, have you not experienced any improvement at all in DACs and amps in recent years? Maybe we should have, but it doesn't feel like we've hit the wall yet. At least, an awful lot of folks on headfi don't think so. Are we all suckers?! 

I'm probably getting off-topic. These improvements are most likely nothing to do with bit depth. They may even have nothing to do with sample rate or timing - I suspect there are far worse crimes being committed by the internals of the electronic circuitry. But, I've often wondered if - one day, with the perfect recording - how nice it would be to crank the volume all the way up in a very quiet/silent passage and still not hear any noise floor.


----------



## bigshot (Nov 13, 2017)

Multichannel sound, digital equalizers and DSPs are the only electronic/digital improvement in sound I've noted in the past couple of decades. Every digital audio product I've had since I got my Macintosh 8500AV in 1995 has been audibly transparent. I'm sure there are bad ones out there, but on the whole, sound quality is pretty much perfect.


----------



## Don Hills

[QUOTE="csglinux, post: 13848731, member: 399084"...  But, I've often wondered if - one day, with the perfect recording - how nice it would be to crank the volume all the way up in a very quiet/silent passage and still not hear any noise floor.[/QUOTE]

You must be a fan of John Cage's music:
https://en.wikipedia.org/wiki/4′33″


----------



## amirm

castleofargh said:


> while I agree that simply adding the noise of the room to the music's dynamic is an oversimplified idea. countering it with audibility measured for white noise where the room's own noise floor is now the loudest signal, that's IMO even more inadequate to depict realistic listening or listening needs.
> sure enough noise isn't some opaque paint spread over the stuff below, but masking is still very much a reality, so is the human's limited instantaneous dynamic range.


Masking is content dependent.  When it is not there, then the noise floor is there to be heard.

What we are trying to do is define a channel which can be shown to be error-free.  Once there, the system will maintain its performance regardless of what content you play.

Notion of masking is very useful when we are looking at a signal and noise/distortion around it.  Such is not the case here.  I can have low frequency music notes, and mid to high frequency noise that would be audible with it due to lack of masking.  

As an analogy, we can say that the restaurant doesn't need to wash their dishes if we limit ourselves to food that is able to kill the bacteria in it.  That restriction would not fly in that situation, and as such, doesn't work here either.


----------



## JaeYoon

amirm said:


> Masking is content dependent.  When it is not there, then the noise floor is there to be heard.
> 
> What we are trying to do is define a channel which can be shown to be error-free.  Once there, the system will maintain its performance regardless of what content you play.
> 
> ...


I'm not sure about that analogy.
One could potentially cause bodily harm to patrons that would consume that food.

Someone listening to masked frequencies would be in no chance of same potential of danger of sickness.


----------



## bigshot (Nov 15, 2017)

amirm said:


> What we are trying to do is define a channel which can be shown to be error-free.  Once there, the system will maintain its performance regardless of what content you play.



If you define a system that outperforms the thresholds of human hearing, they don’t have to be error free.

A better analogy would be... if you ditch classes at school to go fishing and your mother doesn't find out, you don't get in trouble. What she doesn't know can't hurt you! That's masking in a nutshell.


----------



## Darren G

A question/thought for the group here (and a 90 degree thought)...

At one point in time I was working on a MPEG-1 decoder.  There was limited content available to test with at that point in time, but as someone starring at the output of decoding every day, what I became aware of is macro blocks, and motion artifacts.  I would show this to others and their reaction was very different.  In their minds it was so clear, and 'perfect', keeping in mind they were use to seeing video played back from films, or tapes, where noise was the norm.

What I realized, or at least started to question was, our brains are very good pattern matching machines.   What bothered me about MPEG-1 wasn't the errors so much as the repetitive nature of those errors.  Eventually that is all my brain locked in on.

Okay so the leading question then is this...

Measuring noise is only part of the story.  The more difficult measurement is repetition of noise.  See I am thinking our brains are very good at masking noise, but if the noise is repetitious, that is what our brains lock on. 

If we ask the question is any bit of noise below audible level, yes it is inaudible, but if there is a pattern to the noise, do our brains pick up on that, and either find it annoying or pleasant?

--

Just some thoughts to throw into the mix...


----------



## 71 dB

The coding artifacts of MPEG-1 video are of course perceptually gigantic compared to quantization errors of 16 bit audio.


----------



## Darren G

Agree 71 dB.  We are well into the area of diminishing returns.   Still, my question is subtle...  and I'll use analogy to explain what I am thinking (it is not perfect for sure).

Imagine a clear window, it is near perfect, but there is a speck of dust.

Many of us ignore the speck of dust, but what we have no clarity (or clearly definable measurement for), is do some people's brains focus almost entirely on that speck of dust, while others just see the clear window?

It's our human norm to believe our perception is universal, but we have many examples of rare individuals who excel in certain areas because their brains are wired differently.  

I am not suggesting the music industry focus on the rare .000(add 0's here)1 percent, only that I do think it is a fair question.


----------



## bigshot

I would say that inaudible is inaudible no matter what the pattern. There certainly are ways that noise can be more annoying, but if it's below the threshold, it's like the tree falling in the forest with no one there to hear it.


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## castleofargh

Darren G said:


> Agree 71 dB.  We are well into the area of diminishing returns.   Still, my question is subtle...  and I'll use analogy to explain what I am thinking (it is not perfect for sure).
> 
> Imagine a clear window, it is near perfect, but there is a speck of dust.
> 
> ...


 when the little bunny poops in the woods, on the mountain I see from my window maybe 10km away.  I don't think the change in panorama is something I should care about. I know that this fracking rabbit is taking a crap on my panorama probably several times a day, I can be mad about the reality of it and if I go walk in the mountain, I could end up finding poop, somehow proving how right I was to be mad about it. I can convince myself that because it is measurable, and even noticeable in a totally different context, then it matters. I can even feel like the beautiful scenery now sucks because of the rabbit. I can buy traps and catch the rabbit, and it would probably make me feel like the view is better than ever.
there is no limit or rule to what someone must consider significant and how much mess is going on in his own mind. but there is absolutely a limit to what a human eye can see when looking at my window. and the rabbit on the mountain on the other side isn't one of them. 

to be clear there is nothing wrong with going highres, or trying to do even better than what's necessary or relevant. the simple idea of going beyond a limit is in itself a strong motivation for many people. but humans have thresholds and those thresholds are determined through practical testing, not through wishful thinking, sighted tests, and made up hypotheses. personally I am not against highres, I'm against pushing highres for BS reasons.


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## Cutestudio

The issues with arguing that all noise is the same and that statistical methods can make 16bit audio as accurate as 24bit remain as spurious as ever. This 16vs24 bit subject however is a discussion in theory only as in the real world we have a very limited choice of most music, generally between mangled mp3, mangled CD spec, not-so-mangled Mastered-For-iTunes, or mangled 'hi-res' versions sold by various chancers. Vinyl is available for some sure, some is just a 3rd rate copy of the mangled CD spec, but some is better mastered. Norah Jones's 'The Fall' is totally mashed on CD - it makes the Chilli Peppers Californication seems quite listenable - but the vinyl is quite good. Or it would be if they'd used some decent vinyl instead of recycled sunbeds and coffee cups to press it on.

I've talked with people who have bought (quite a lot of) hi-res audio and then found it's just the same as the regular 44.1/16 when viewed in a waveform editor like Audacity. You can't see the fine detail but you can see the usual brick shaped waveform with the usual damage as the dynamics were rolled out of it by a mastering 'engineer' so everything turns into a nice dull flat noise to be turned down and ignored by the sucker who's bought it..

Genuine hires can have the advantage of a frequency response. The dull 'you can't hear above 20kHz' mantra may have been fine in the 1930s, but since stereo was invented you can have all types of beat frequencies interacting between the channels, so there's a ready explanation of why a higher HF limit may sound better. The luddites would never wear a distribution format upgrade though.

However still the main problem with both hires and the regular 44.1/16 is the lack of the MSB (due to the insidious and endemic loudness war), so you end up with compression/clipping forming a series of flats in the waveform that puts many a mediaeval castle's castellations to shame. People who don't realise that the MSB is missing don't spend their days looking at the waveforms, it's not my opinion: it's a simple fact for most modern music today. The people who 'know' this isn't true need not reply, merely look at their own tracks in a waveform editor to see for themselves.

There is an argument I've heard that the loudness war is partially caused by 16bit recordings sounding a little lifeless when not fully driven but I think there are a number of factors that have left us in the situation today whereby we can argue about 16bit vs 24bit for years: but still not have the opportunity to buy a decent unclipped/compressed recording of much at all in either format.

24 bit does have some tangible benefits to digital audio however, even just as an output format: one of which is the ability to use a digital level control without the degradation of signal that a 16bit output exhibits. You can demonstrate this by simply turning down the digital level as you turn up the analog level. The degradation of the 16bit signal is quite pronounced but the 24bit is far more usable as a level control without impacting quality too greatly.


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## bigshot (Nov 15, 2017)

Hey Cutestudio! I would chop up your post and answer it line by line, but that would be annoying and would just make you mad. It's a lot nicer just to point out that if you're interested in this subject, you should read the article and watch the two videos in my sig file. The videos have downloadable audio files associated with them so you can hear the truth for yourself. Obviously, you've spent some time thinking about how digital audio works, but I think you'll be surprised at how much more there is to learn than what you already know. It's always good to learn, right?

Start with the article, and after you have a chance to digest it, then watch the videos. Sound Science is fun!


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## Darren G

castleofargh said:


> when the little bunny poops in the woods, ...



LOL, what a great analogy.  I am objective (prove it) type, so you'll always find me on the side of the mountain that doubts anyone is hearing a gnat fart a mile away.

On the flip side, it is quite interesting how *quantization noise* is something some human brains may pick up on, and the solution is as simple as adding some random noise.  I am over extending my knowledge here so sorry in advance for any misinformation.

What I am trying to understand is vague anyway, but the gist of it is, is there a threshold range of noise that is essentially inaudible, but if it is repetitious in nature, that (for some) elevates what should be inaudible into the perceptual realm of audible (conscious or not) for some?


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## amirm

bigshot said:


> If you define a system that outperforms the thresholds of human hearing, they don’t have to be error free.


And that is exactly what we are doing.  We measure the system level of noise, compare it to threshold of hearing, then determine the loudest real life music we can find, and the required dynamic range falls out of that.  From Fielder paper's summary (which is also in my article): 







Using the simple math of 6 db/bit we see that we need 122/6 = 20 bits.  16 bit audio at 96 dB will not do it.

We can use noise shaping to improve 16 bit dynamic range in audible band but then we run into the limited bandwidth of 44.1 Khz.  And second real life problem that you have no assurance that when the music was produced, noise shaping was used.

This is what I call error free channel.  It is perceptually so.  An objective error free channel of course does not exist when it involved analog signals we hear.


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## bigshot

I've said it before and I'll say it again. If you aren't listening to your music at 122dB, you don't need a dynamic range of 122dB. And the noise floor of a typical living room is between 35 and 40dB, not 20 to 30. Your numbers are wrong. If you get out a SPL meter and check for yourself what those numbers represent in real world sound, you'll see I'm right.


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## Darren G

Another aside, but such a ton of valuable information here.  Even if there is no agreement, this sub-forum, reading the insights from the audio engineers is cool crap how much depth and detail goes into mastering audio.


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## 71 dB

Who the hell hears 4 dB sounds when exposed to peaks over 120 dB? Your ears will be still ringing the next day! That Conclusion 7 calculation is ridiculous.


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## amirm

bigshot said:


> And the noise floor of a typical living room is between 35 and 40dB, not 20 to 30.


That's just wrong.  I have explained this many times to you.  Yet you keep ignoring the science/psychoacoustics.  Single number noise levels like this are useless. You must, must decompose them into their spectrum, convert them to equiv. sine wave levels, and then see how they compare.  That is exactly what I did in my article on noise levels of our listening rooms.  And that is based on peer-reviewed, journal of AES paper, written by ex-president of Audio Engineering Society.   Again, he is Fielder in his paper:





And the figure:





As you see, even the average home has 10 db SPL of noise in the mid-frequencies where our hearing range is most sensitive (as shown by the solid line).

It doesn't matter if you agree or disagree on this topic as a whole.  At least learn this bit of audio science and don't keep repeating objectivists myths of using one number to determine noise audibility.

*Your SPL meter lies to you. * It absolutely does.  It doesn't have your brain or your two ears.  *In acoustic science, measurements can get you in trouble and fast. *  While said in a different context and purpose, Dr Toole says this best in his book:

This disagreement between what is measured and what is heard has been
the motivation for much scientific investigation of the acoustics of rooms, both
large and small. In some ways, *our problems with rooms, especially small
rooms, began when we started to make measurements.*​
I will say it again, please don't keep using single value SPL numbers as meaning anything with respect to audibility of noise.  It is just junk forum objectivists banter that has no basis in real science.


Even worse in this context is the fact that we are in headphone forum.  Headphones can block amazing amount of noise in those mid-frequencies.  Once there, you can easily hear the faintest level of noise created by the equipment.  Here is the data from my IEM, the etymotic E4SR:





That is a whopping 55 to 60 db of noise reduction at mid frequencies!!!

Furthermore, while stereo playback allows the sound in one channel to mask the noise in the other, no such thing happens with headphones as each ear hears what is intended for it.

So whatever argument you have here, simply does not hold when it comes to headphones.

Honestly, I have shown you all of this science and measurements before yet you repeat the same argument over and over again.  Do you have any references to back your assertion and if not, why insist on it over and over again?


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## Strangelove424

How is the "loudest real life music" instrument measured? I'm assuming acoustic sources only, then which instrument and how was it mic'd up? Did they stick a mic directly into the drums, or rest it on the skin? These numbers are insanely high for a music listening scenario. 120db is way beyond the safe limit. To give you some perspective, here's a French air raid siren at 120db, or this car-mounted train horn. Who listens to music at 120db? They must be deaf. I want to meet that person. I'm sure they would ask me "what?" alot. BTW, 24 bit is artificially reduced to 123db SNR (20 bit equivalent). A true 24 bit dynamic range is an illegal signal and dangerous for equipment/hearing. So with an instant deafness-inducing 140db SNR ruled out, it's simply a choice between a painful 96db or tinnitus-inducing 120db.


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## bigshot (Nov 15, 2017)

120dB is a VERY UNCOMFORTABLE LISTENING LEVEL. Find another cite that says you need 120dB of dynamic range to accurately reproduce recorded music. Happy hunting!

Honestly, if you knew what the numbers on the page represented, you would have a much better batting average at sorting out the wheat from the chaff.


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## 71 dB

amirm said:


> Honestly, I have shown you all of this science and measurements before yet you repeat the same argument over and over again.



Yes you have but you don't understand the relevance of the numbers you have shown. People don't hear the noise floor of 16 bit audio when they listen to music, not even when the track fades away. 16 bits would be enough even without dither and dither gives a few bits worth of extra practical dynamic range (up to ~120 dB!). In comparison the dynamic range of vinyl is 10 bits at best and vinyl lovers think even that is enough!


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## amirm

71 dB said:


> Yes you have but you don't understand the relevance of the numbers you have shown. People don't hear the noise floor of 16 bit audio when they listen to music, not even when the track fades away.


How did you determine that?  Or is this based on some research I can read?


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## JaeYoon

bigshot said:


> I've said it before and I'll say it again. If you aren't listening to your music at 122dB, you don't need a dynamic range of 122dB. And the noise floor of a typical living room is between 35 and 40dB, not 20 to 30. Your numbers are wrong. If you get out a SPL meter and check for yourself what those numbers represent in real world sound, you'll see I'm right.


Can confirm, I installed a SPL meter app on android and can confirm 35 - 46 DB in my room.


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## bigshot

I don't think amirm owns an SPL meter. Even a cheap one would give you enough info to be able to sort out hooey like that. Have you tried getting a level on a loud listening level JaeYoon? It sure isn't 120dB!


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## Don Hills

When you look at an SPL meter in a "quiet" room, you're looking at the peak SPL at whatever frequency it happens to occur at. If you look at the graph posted by Amir, you can see that the SPL increases at lower frequencies, and that's what the meter is showing. It's also why you get a dramatic difference in the reading when you switch between A and C weighting. You should be using A weighting for hearing threshold type measurements because it more closely matches the ear's own sensitivity curve. It's still not close to a perfect match though. So in this case Amir is correct, you need to measure the SPL at multiple frequences and plot the curve to get a true picture of how audible the SPL in the room actually is.


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## 71 dB

amirm said:


> How did you determine that?  Or is this based on some research I can read?



I have been listening to CDs for 3 decades and never have I heard the noise floor of 16 bit audio. All the noise I have heard is part of the recording and would be there no matter how many bits were used. I have also never heard anyone complain about hearing 16 bit noise floor. As an acoustic engineer I have some understanding of the practical demands of dynamic range in audio and anyone who has played with 16 bit audio in a wave editor, downsampled 24 bit to 16 bit using dither understands that 16 bit is enough.

The overall dynamic range of human hearing is 120-130 dB from hearing threshold to pain threshold, but not all of it is usable. If you are exposed to loud sounds, your hearing threshold raises temporalily. After having been some time in a silent environment you can hear silent sounds again. That's why you really need less dynamic range than 16 bit audio offers and why the "10 bits at best" dynamic range of vinyl seems to be enough for many.

I can understand why someone who knows nothing about digital audio or human hearing would think 24 bit consumer audio is better than (has benefits over) 16 bit consumer audio. I guess it's human nature to insist on "intuitive" concepts even when your undertanding and knowledge tells you otherwise. Companies making money on ignorance of people surely want to promote 24 bit high-res as "superior", but we who say 16 bit is enough have no such motives. Our only motive is telling the truth, because we feel it's our duty to use our knowledge that way against financial interests and lies.


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## gregorio

amirm said:


> Noise in your recording comes from your speakers, i.e. point source.  Noise that is in the environment in your room is diffused all around you.  Putting aside the important bit that the spectrum of noise in our living room is anything but "white" (it is heavily biased towards low frequencies because walls and doors don't filter it as much), our hearing system is capable of distinguishing between point noise sources and diffused one.



Obviously my OP was simplified for brevity. It would have been entirely possible to write a short book just on noise, because it's a complex subject with a large number of variables: On the purely digital side of things we've got the variable of noise-shaped dither, which has been standard recommended practise for well over 15 years and first started being used over 20 years ago. On the recordings themselves we got the noise variables of mics, mic pre-amps and mic positioning, the noise of the recording location and the instruments and artists present in that location and then we have the noise induced by processing during mixing and mastering. Lastly, we've got the noise of the playback system, the DAC, amp, listening environment and transducers. 

Nearly all of these variables are not just variable but highly variable and their interaction is complex. Yes, the typical environment noise of a listening environment is more pink-ish than white. On the other side of the coin, due to equal loudness contours at low levels, we're often not going to hear it as pink-ish, although that depends on where we live and the exact distribution of the noise. On the transducer side; yes, very well sealed IEMs might reduce listening environment noise by as much as 50dB or so, while open backed headphones might only reduce it by 10dB or so.

Despite my severe over-simplification, the basic point I was trying to convey still stands. Even if a certain combination of consumer equipment provides a listening noise floor of 0dBSPL, still 16 bit digital would allow for a peak level of 120dBSPL which, with such a low noise floor, would be way beyond comfortable and into the range of potentially damaging.



amirm said:


> We measure the system level of noise, compare it to threshold of hearing, then determine the loudest real life music we can find, and the required dynamic range falls out of that.



No, that's what you're doing, it also appears to be what some scientists do BUT it is NOT what "we" do. Your apparent definition of "real life" music is bizarre, to say the least! Sure, if we place a mic a CM or so from many instruments, we could in theory record levels up 120dB or more. Heck, place a mic a CM or so from someone's mouth when they're speaking just a little loudly and we can get also easily get peak levels of 120dB+ ... *BUT, *put together your variables of: An extremely quiet listening environment, someone speaking a bit louder than normal and put your ear just a centimetre or two from their mouth and observe for yourself the results, or rather *DON'T, DO NOT TRY THIS*!! It's not a safe experiment, if it's a quiet environment NEVER allow someone to put their mouth very close to your ear and speak, unless they whisper very much more quietly than their normal speaking level! So, what is "real life" music? You typically place your ear a CM or a couple of inches from the instrument/s during a performance do you? How is that in any way "real life"??

In most scientific papers, the output levels and freqs are typically measured by a mic placed very close to the instrument/s and this is the max level/freq quoted. In real life, the situation is very different. In the case of most popular music genres, we do often record with a mic placed closely to the sound source BUT in order for it to sound subjectively pleasing, we then have to heavily process such recordings: EQ, artificial reflections, very significant level changes and various stages of compression to "shape" the transients. With acoustic music genres (classical, jazz, etc.), we tend to not apply heavy processing during mixing and mastering, instead, by positioning mics far more distantly we effectively achieve natural compression, transient shaping, input levels, EQ and reflections, due to absorption, and then all that's needed is just simply balancing the output of those mics during mixing/mastering with very little (or sometimes even no) additional EQ and compression.

In short, you're talking about the real life levels achievable with close mic'ing and adding this to the real life acoustic music recording technique of little significant processing BUT the problem is that this combination of real life situations do NOT actually occur in real life! 

Additionally, you are ignoring some of the science here; while we have a large dynamic hearing range, it's not all usable at once. Just like with our sight, we can see in a near completely dark room and we can also see in bright sunlight but go from a very dark room into bright sunlight and what's the result?. Our hearing maybe capable of 120dB or more of dynamic range but the most optimistic scientific figures I've seen puts this at a max of about 60dB at any one time and I don't believe it's coincidental that commercial recordings almost never exceed this same, roughly 60dB, dynamic range. This brings us back to the content which is being listened to, it is not designed to test the limits of the human ear, it's designed as entertainment and to be comfortable. 



amirm said:


> That's just wrong.  I have explained this many times to you.  Yet you keep ignoring the science/psychoacoustics.



And I've explained to you in another thread all of what I've just said above and yet here you are again, stating and quoting exactly what you did to start with, completely oblivious to what you've been told! You quote only the science/psychoacoustics which supports your agenda (whatever that may be) and reference it to "real life", while deliberately ignoring other science, psychoacoustics and "real life" practicalities (which doesn't support your agenda) and then you accuse others of "keep ignoring the science/psychoacoustics"??! If you're going to quote "real life" then you HAVE to quote "real life", not just certain cherry picked aspects of real life which suit your agenda but which are NOT real life because you are omitting the other factors which would actually make it real life! 

G


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## 71 dB

60 dB test demonstrates a lot: Make 2 seconds of pink noise in a wave editor and attenuate the latter 1 seconds by 60 dB. Then play the noise in a loop: LOUD - quiet - LOUD - quiet - … and try to hear the quiet noise. Yes, you can make it audible by turning the volume up so that the LOUD part is uncomfortably loud. This is "vinyl" dynamic range. 16 bit digital audio even without dither does much much better and with dither perceptually even better than that! That's why 16 bits is enough, perhaps even 12 bit optimally used and dithered could be I think.


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## OddE

bigshot said:


> 120dB is a VERY UNCOMFORTABLE LISTENING LEVEL. Find another cite that says you need 120dB of dynamic range to accurately reproduce recorded music. Happy hunting!
> 
> Honestly, if you knew what the numbers on the page represented, you would have a much better batting average at sorting out the wheat from the chaff.



-Just to put @bigshot 's post in further perspective:

I have just checked out a hydraulic power unit for my employer, ensuring all of the instrumentation we applied to it actually works. (It does! Coffee for everyone!)

This particular unit had three 270kW pumps running at full blast; that is sufficient energy to accelerate 150 metric tons of very expensive stuff away from the seabed, double haste. Standing between the pumps is a very physical experience - you can feel your whole body being pounded from the vibrations, and even wearing (proper!) hearing protection (foam plugs + peltors), the noise level is such that I find it slightly uncomfortable.

I took a reading. 119dB(A).

However, this is loudness wars taken to extremes; hardly much by way of dynamic range - so I could see myself listening to music with the occasional peak around the 120dB mark. If I had stuffed foam earplugs in my ears and then put Peltors over them, that is!

Now, all I need is to find someone who wants to record the sound of butterfly wings fluttering inbetween same pumps, and I've found a business case for 24 bit audio...


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## TheSonicTruth

PETEBULL said:


> Damn it! You can have FLAC the size of MP3! Freaking EPIC!!!
> Oh wait. Sometimes you get artifacting ("herding_calls" sample)  but not on brickwalled audio.
> Yeah, I hear some noise on Daft Punk - Instant Crush. That "gangam style" is way too brickwalled and thus fraudy.
> Upd. Using dithering can fix that noise  though it occupies ~100 Kbps more. Still the noise can be an audiocassette fetish for some.




And this brings up a point:  This whole thread ignores the elephant-in-the-room:  The CONTENT.  16bit VS 24bit is a moot point if the master is hyper-compressed and has had the top 4-8 dB brick-wall-limited off of it!  

 Now I do realize that that has to be done mainly at the behest of the clients(the artist, the band, or the producer or label) that want volume level 10 loudness at a volume knob setting of 2, but it belays digital audio's ability to actually sound great.

Most of us on here have probably read that on-line article about the Nirvana 'Nevermind' 24bit high-rez downloads, and what was lost. But that example proves that there is no point in a high-res deliverable if the content itself is, under the marketing guise of 'remastering', treated to a steam roller and a lawn mower set too low!  

That said, 24-32bit/96-192khz sampling will continue to be the obvious choice for tracking, mixing, and post, while 16/44.1 will still be more than adequate for deliverable(CD, download, etc) for the forseeable future.

Once both artists and engineers get past this addiction to everything in a song - from the lead vocals to the rhythm section to the melody and backing vocals - being dialed up to eleven, and digital's true dynamic range is seriously exploited, then we can start talking about 'higher res' deliverable formats.


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## 71 dB

In everyday life the difference of loud and quiet is 30 dB, 5 bits.


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## castleofargh

to put a perspective, not on hearing, but on point of views, @amirm is clearly considering all the extremes in isolated conditions. it makes the practical results somehow ludicrous for music listening, yet true in the sense of noticeable to human if all the conditions are met. it's like jitter tests where people will achieve a given result for a given type of jitter in a specific test with specific signal, but then we play music instead of the test signal and the same people will stop noticing stuff a good magnitude or two sooner. it doesn't make the first test false, only very conditional and not at all representative of our practical needs for usual listening of musical content.
of course it doesn't mean that when I'm a in club (France laws require clubs to stay below 102dB(a) / 118dB(c) over 15mn), I'll get pissed off by my friend breathing very quietly 3 meters away from me and ruining the music. we all understand that hearing just doesn't work that way including @amirm . but as music could in principle be any sound at any non too painful loudness, @amirm is just covering all situations including those which will never happen. I don't think he said anywhere that we were able to notice both extremes at all time so let's not put words in his mouth. 

 on a practical level with my favorite songs at my desired listening levels, I have the hardest of times noticing anything at -80dB. I need instead to listen loud over a very quiet passage for such a stuff to start making a difference in my musical experience. what he says and what I say aren't contradicting each others, they simply relate to clearly different listening conditions and so we reach different results. not that controversial. I'm of the opinion that typical musical content already places us well outside of his concerns, but as the media to solve his concerns already exists(for recording and playback it's a different matter, but the media is fine), of course he should use 24bit. and I'll be sticking to 16bit for a while longer because my needs are already covered that way.


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## Cutestudio

TheSonicTruth said:


> And this brings up a point:  This whole thread ignores the elephant-in-the-room:  The CONTENT.  16bit VS 24bit is a moot point if the master is hyper-compressed and has had the top 4-8 dB brick-wall-limited off of it!
> 
> Now I do realize that that has to be done mainly at the behest of the clients(the artist, the band, or the producer or label) that want volume level 10 loudness at a volume knob setting of 2, but it belays digital audio's ability to actually sound great.
> 
> ...



Good to read at least one other person in this thread has spotted the elephant. At last. The clipping and compression reduce the peaks by around 6dB, which can be seen with a small amount of research easily enough. Madonna's original 'True Blue' vs the remashed 'True Blue' is a case in point, compare them and you can see about 6db (the MSB) has been lost and the result is about 1/2 the peak level that was there in the first place for the same RMS level. The difference between 16 and 24 bit is an extra 256 levels below the LSB (-96dB) in 16 bit which is useful, but we have a +6dB problem with todays digital music releases.

Additionally the argument about 120dB total range is rather pointless, if you have even a simple digital level control 24 bit is the way to go even if you start by feeding 16bit in at the start. Most DACs will be what - 18 to 20bit so any replay-gain style levelling will do best output as a 24bit source. I.e. 1/16th of a 16bit signal is 12bits, from 24bits you still have 20bits. So due to further audio processing after the CD the 16bits is rather a moot point because clearly 24bit allows digital EQ (including digital speaker crossovers) and attenuation so the aim is to escape from 16bit ASAP. 

Given then that 16bit is only relevant as a compact distribution medium the lack of the top bit is a serious and major problem for any HiFi. What's the distortion of a clipped or limited peak - 50%? So given we have a 50% grade distortion problem It's fascinating how we got to page 290 of a discussion about distortion levels of around 0.003%. The elephant is truly invisible.


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## RRod

There are genres that aren't compressed to hell. I have plenty of stuff mastered near -30dB RMS with 0dBFS peaks in my classical collection.



castleofargh said:


> on a practical level with my favorite songs at my desired listening levels, I have the hardest of times noticing anything at -80dB. I need instead to listen loud over a very quiet passage for such a stuff to start making a difference in my musical experience. what he says and what I say aren't contradicting each others, they simply relate to clearly different listening conditions and so we reach different results. not that controversial.



Yes, the first thing you need is music that makes you turn the pot up high in the first place. You then need a room quiet enough to hear the errors. No real mystery, but it's simply not reality for most folks.


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## gregorio (Nov 16, 2017)

castleofargh said:


> on a practical level with my favorite songs at my desired listening levels, I have the hardest of times noticing anything at -80dB.



That figure would easily cover the vast majority IMO, more than 99% of people in real life listening.



TheSonicTruth said:


> And this brings up a point:  This whole thread ignores the elephant-in-the-room:  The CONTENT.



No it doesn't, I've mentioned the content many times in this thread and indeed my last post was largely about the content. Of course though, there's different types of content, the vast majority of music uses no more than about 8 bits, a lot of it effectively uses no more than about 6 bits but some, mainly symphonic music, can use up to 10 bits or so. This is rather a simplification though, there's a whole range of noise and noise floors to consider and many of the posts in the last few days have effectively centered around the decay/reverb tails of notes fading into these noise floors. How far/quickly they fade into the noise floor in real life (say at a performance) verses how far it's possible to fade them in mixing/mastering on a recording and then how far science tells us it's possible to hear.



Cutestudio said:


> [1] The clipping and compression reduce the peaks by around 6dB, which can be seen with a small amount of research easily enough. Madonna's original 'True Blue' vs the remashed 'True Blue' is a case in point, compare them and you can see about 6db (the MSB) has been lost and the result is about 1/2 the peak level that was there in the first place for the same RMS level.
> [2] The difference between 16 and 24 bit is an extra 256 levels below the LSB (-96dB) in 16 bit which is useful, but we have a +6dB problem with todays digital music releases.
> [3] Additionally the argument about 120dB total range is rather pointless, if you have even a simple digital level control 24 bit is the way to go even if you start by feeding 16bit in at the start. Most DACs will be what - 18 to 20bit so any replay-gain style levelling will do best output as a 24bit source. I.e. 1/16th of a 16bit signal is 12bits, from 24bits you still have 20bits. So due to further audio processing after the CD the 16bits is rather a moot point because clearly 24bit allows digital EQ (including digital speaker crossovers) and attenuation so the aim is to escape from 16bit ASAP.



1. You can't put a figure on it but typically, in the case of popular music, it would be way, way more than 6dB. Madonna's original mix would have had way more than 6dB of compression to start with, even before the original mastering, let alone after the remastering. And no, there's no way to know how much compression has been applied without access to the original recorded, unprocessed tracks. Audiophiles have got to get away from the notion that compression and distortion are bad things, on the contrary, they are essential and have been for 50 years or so. Do you really want all the guitar solos in rock and pop music to sound like a virtually inaudible series of very short duration twangs, because that's what an electric guitar sounds like without compression and distortion? And what about all the other elements/instruments in the mix? Without compression and distortion the popular/rock music you listen to would not sound anything like pop or rock music, so please stop with the "we've lost the MSB" because we've lost a whole lot more than the MSB and gained a far greater amount from that loss!!

2. Nope, even the OP discussed this. The number of quantisation values doubles for each bit of data: 16bit = 65,536 values, 17bit would therefore = 131,072 and 24bit = 16,777,216. So the difference between 16 and 24bit is obviously not 256 potential values but 16,711,280!

3. Again, no! You've made a good argument for a processing environment to be greater than 16 bits but not the distributed audio files themselves. This is why today's pro mixing/processing environments are typically 64bit float and even going back around 20 years they were 32bit float or 48bit fixed.

G


----------



## bigshot (Nov 16, 2017)

TheSonicTruth said:


> And this brings up a point:  This whole thread ignores the elephant-in-the-room:  The CONTENT.  16bit VS 24bit is a moot point if the master is hyper-compressed and has had the top 4-8 dB brick-wall-limited off of it!



It isn't even an issue with dynamic recorded music. The most dynamic recorded music doesn't exceed a dynamic range of 50dB or so, and music is usually normalized to a few dB of the zero line. With redbook, that leaves a whole lot of room to play "hide the noise floor". I've done a lot of mixes where we worked at 24 bit and then bounced down to 16 at the end. We carefully compared the 24 bit mix to the 16 bit bounce down each and every time. Never a difference. That's on studio equipment with a room full of critical ears doing the checking. 16 bits is totally sufficient to cover the dynamics in any recorded music.



castleofargh said:


> to put a perspective, not on hearing, but on point of views, @amirm is clearly considering all the extremes in isolated conditions. it makes the practical results somehow ludicrous for music listening



And we're in a forum to discuss how science can help improve the perceived quality of sound in recorded music. Arguing for extremes like this belongs in a forum for research scientists or perhaps studio engineers, not for home stereo enthusiasts. Demanding that a DAP have a noise floor that far down is no different than demanding that Dark Side of the Moon requires 24/192 for distribution and that frequencies above 20kHz are worth going to the mat for. That kind of obsessive pursuit of the unhearable is patently absurd.

Cutestudio, compression is a recording tool, just like EQ, reverb and levels. It isn't good or bad unto itself. It's all about how it's applied. You wouldn't like the sound of Madonna's song if absolutely no compression was applied. You'd struggle to understand the words and the subtler sound of the instruments would be plowed under. Also, I don't know what you mean by missing "top bit". Dynamic range in digital audio extends downward, not upward like burning in on analogue tape. The difference between 16 bit and 24 bit is in the quietest parts, not the loud ones. The part of the dynamic range that 16 and 24 share at the top are identical. Did you have a chance to read the article in my sig called CD Sound Is All You Need yet?


----------



## amirm

71 dB said:


> I have been listening to CDs for 3 decades and never have I heard the noise floor of 16 bit audio. All the noise I have heard is part of the recording and would be there no matter how many bits were used. I have also never heard anyone complain about hearing 16 bit noise floor.


So when you said "people" can't hear this, you were merely reflecting yourself.  In that case, it would be best to say it that way.

Now as to yourself, you are not explaining any experiment that would back what you said: "_People don't hear the noise floor of 16 bit audio when they listen to music, *not even when the track fades away.* "  _Have you done any conversions of 24-bit content to 16 bits while the content fades away as to know there is no difference there?

Also, how do you know what was in the recording vs what was in the channel?

Finally how did go from people can hear to don't "complain?"  How would they complain without a reference otherwise?



> As an acoustic engineer I have some understanding of the practical demands of dynamic range in audio and anyone who has played with 16 bit audio in a wave editor, downsampled 24 bit to 16 bit using dither understands that 16 bit is enough.


Great.  Please provide an example of these where you couldn't hear the difference and let us do a listen.


----------



## amirm (Nov 16, 2017)

Let me state my position without the back and forth as to make it more clear.

It is important in the context of audio "science" that we are true and correct to what that is.  Paper after paper from luminaries in audio show that 16 bits is insufficient dynamic range (without noise shaping) to be transparent to listeners.  There is also research that shows high resolution content resampled to 44.1 Khz can have audible consequences.  If you are going to jump up and down and say this isn't so, please don't bother unless you have research you can put forward to the contrary.  Or controlled listening tests you have performed.  Otherwise, it really is useless to give me anecdotal information about what you think or hear.  That is not material.

It is also true that vast majority of people and this includes audiophiles will have a heck of time telling the difference between high resolution content and CD rate.  We as listeners simply don't know what these effects are and much of what we say we hear in sighted listening is not because of what we hear, but what we think we hear.  Given this, why do I insist on the first paragraph above?  Simple: if as objectivists we wear the cloth of science, science should not be the first thing we sacrifice to promote our message.  We need to be truthful and knowledgeable about what the science says in this regard.

In my past career, I have done a ton of controlled testing and found the above to be very true.  But what was also true was that I and the rest of trained listeners in my group and elsewhere in the industry could readily hear and identify artifacts that vast majority of people could not.  Unless you have been exposed to this class of people (and a few gifted individuals who have these abilities without training), you can't generalize to what "people can hear." 

Heck, I can teach you to hear some of the things you say are impossible to hear!  I suggest not going there though as it is not good to learn to hear small differences. 

So in summary, pull back a bit from extremism here.  Our case doesn't hold when we go there.

Finally, all of this talk is immaterial anyway.  CD as a format has had its useful life and there is no reason for us to continue to melt plastic to make it.  We can deliver content online without such a constraint and vast majority of our devices already knows how to play high-res.  To that end, I like to get my hands to stereo mixes prior to CD mastering.  Whatever that sample rate is, I want it!    If I want it at 16/44.1, I can convert it myself or download that version which usually is available anyway.  I don't want my content to have been subjected to loudness compression which sadly comes with mastering the CD.  By constantly defending the CD as a format, we work against this ideal.  That is not right in my book.


----------



## 71 dB

amirm said:


> So when you said "people" can't hear this, you were merely reflecting yourself.  In that case, it would be best to say it that way.



Yeah, but you are also hinting you _can_ hear it. Why else would you speak for 24 bit audio if 16 bit was enough to you?



amirm said:


> Now as to yourself, you are not explaining any experiment that would back what you said: "_People don't hear the noise floor of 16 bit audio when they listen to music, *not even when the track fades away.* "  _Have you done any conversions of 24-bit content to 16 bits while the content fades away as to know there is no difference there?



Yes, I have and that's why I know I know this stuff (I knew it long before because my university studies included digital audio), but the conversion tests confirmed it for me). I studied how dither allows sounds below quantization noise floor to be heard. The noise floor must be amplified A LOT to be heard even when the signal is quieter than the noise itself!



amirm said:


> Also, how do you know what was in the recording vs what was in the channel?



Well you can't, except there are recordings without noise to those demonstrate how 16 bit can do it.


----------



## Darren G

Again a 90 degree view...

I've owned gear that annoyed me to no end, and it was NOT because of noise. 

All transducers had a sound signature, some I enjoy, some I found annoying, and (yes some of you won't like this), but something about the amp, or yes the DAC, had a repetitious quality that I found annoying.

Today I have gear that I enjoy much, but the noise level?  Irrelevant.  As long it is noise, random, I don't even listen to it.  Sometimes it is as recorded, some times it's just noise, but noise doesn't bug me.  Go to hear live music and there is massive noise, but who cares?

None of this is a sound engineer's problem to resolve, but I use an amp with tubes in the pre-amp section.  I don't care if the tubes are imperfect.   It is completely irrelevant to me, as a listener.  What I do care about is it sounds good, and so I listen to more music.  I prefer 6922 tubes because of the lower noise floor, but I am not listening to the noise anyway.   I am listening to does this sound good to me?

Some gear just annoys me.  My brain objects.  Some gear sounds pleasant.  It may well be the poorer performing gear, but again, nobody goes to a live concert and doesn't enjoy it because there is noise coming from the speakers.

Also while I have been to live performances that may reach that 120db peak, I've also left because it was so loud it was painful.  Dynamic range is one variable, but it is not the only one that matters.  A great master CD at 16 bits of dynamic range still kicks butt over a crap one at 24 bits.


----------



## amirm

bigshot said:


> It isn't even an issue with dynamic recorded music. The most dynamic recorded music doesn't exceed a dynamic range of 50dB or so, and music is usually normalized to a few dB of the zero line.


Where is the research and data to back this?  

Heck, what is the tool that was used to determine this?  Have you converted your music to less than 10 bits which this represent and failed to tell the difference?


----------



## amirm (Nov 16, 2017)

71 dB said:


> Yeah, but you are also hinting you _can_ hear it. Why else would you speak for 24 bit audio if 16 bit was enough to you?


As I just explained, I ask for 24 bits because that is what is used to create the music in the first place.  I have no use for someone doing the conversion to 16 bits for me with who knows what scheme.  Or for what reason.

But yes, I have run a public test put forward by objective  blogger, Archimago and here are the results:  http://archimago.blogspot.com/2014/06/24-bit-vs-16-bit-audio-test-part-i.html

foo_abx 1.3.4 report
foobar2000 v1.3.2
2014/08/02 13:52:46

File A: C:\Users\Amir\Music\Archimago\24-bit Audio Test (Hi-Res 24-96, FLAC, 2014)\01 - Sample A - Bozza - La Voie Triomphale.flac
File B: C:\Users\Amir\Music\Archimago\24-bit Audio Test (Hi-Res 24-96, FLAC, 2014)\02 - Sample B - Bozza - La Voie Triomphale.flac

13:52:46 : Test started.
13:54:02 : 01/01 50.0%
13:54:11 : 01/02 75.0%
13:54:57 : 02/03 50.0%
13:55:08 : 03/04 31.3%
13:55:15 : 04/05 18.8%
13:55:24 : 05/06 10.9%
13:55:32 : 06/07 6.3%
13:55:38 : 07/08 3.5%
13:55:48 : 08/09 2.0%
13:56:02 : 09/10 1.1%
13:56:08 : 10/11 0.6%
13:56:28 : 11/12 0.3%
13:56:37 : 12/13 0.2%
13:56:49 : 13/14 0.1%
13:56:58 : 14/15 0.0%
13:57:05 : Test finished.

----------
Total: 14/15 (0.0%)

As you see, I achieved near perfect (14 correct answers out of 15) results.  To be clear, I can't always do this.  Not all content or even most of them are revealing of such differences.  What is uses is what is available for download freely (from 2L site I think) and is not necessarily content that can show such differences.

And oh, both files say "24 96" but one of them is converted to 16 and then back up to 24.  In other words it has no more information beyond 16 but it is presented as 24 bit as to make computer analysis very, very difficult.  And to be clear, the above results are pure listening test with nothing but Foobar player playing the content and me listening on my laptop with my IEMs.

I think the files are still there.  If not, let me know and I will provide them on dropbox.  I would be curious to see you run them and report on whether you can or cannot hear the difference.


----------



## 71 dB

amirm said:


> Finally, all of this talk is immaterial anyway.  CD as a format has had its useful life and there is no reason for us to continue to melt plastic to make it.  We can deliver content online without such a constraint and vast majority of our devices already knows how to play high-res.  To that end, I like to get my hands to stereo mixes prior to CD mastering.  Whatever that sample rate is, I want it!    If I want it at 16/44.1, I can convert it myself or download that version which usually is available anyway.  I don't want my content to have been subjected to loudness compression which sadly comes with mastering the CD.  By constantly defending the CD as a format, we work against this ideal.  That is not right in my book.



Where do you store you downloads? You always need material to store information be it a CD disc, USB memory stick or a hard drive. Loudness war is a commercial thing. People with more sophisticated music taste buy more music which does not suffer from it. I'm listening to Joseph Schwantner's Orchestral Music on Naxos while writing this and I don't hear ANY loudness compression. If there is any, it certainly doesn't do harm. Sounds dynamic!

We defend 16 bit digital audio in this thread. Doesn't matter if it's delivered on a CD or if it's a FLAC file and_ anything_ can be mastered to sound horrible.


----------



## gregorio (Nov 16, 2017)

amirm said:


> It is important in the context of audio "science" that we are true and correct to what that is. Paper after paper from luminaries in audio show that 16 bits is insufficient dynamic range (without noise shaping) to be transparent to listeners.
> 
> 
> > bigshot said: ↑ It isn't even an issue with dynamic recorded music. The most dynamic recorded music doesn't exceed a dynamic range of 50dB or so, and music is usually normalized to a few dB of the zero line.
> ...



Yes, this is a sub forum of sound science but it is not a forum for ONLY sound scientists, it's a forum for audiophiles to discuss their hobby, to understand how sound works and how it applies to their hobby. The members here are listening for enjoyment to commercial music/audio releases in consumer environments, albeit typically better environments than the average consumer. This means they are not dealing with pure science, only scientific test signals, only laboratory listening environments or the ultimate limits of human hearing. They are dealing with content created by artists and engineers for normal people in normal listening environments, listening normally and for some/much of what we're dealing with there's relatively little scientific research.

For example, your question to bigshot; I've never seen scientists sit and measure the noise floor and peak levels of say a live symphony performance in the middle of the audience. Maybe there is such research but I've not seen it, have you, can you quote it? What we do have is tens of thousands of sound recordists/mixers, many decades of practical, real life knowledge/experience and the millions of albums (plus millions of films and TV shows) to which everyone here is listening. Is this knowledge/experience all utterly invalid because scientists have not confirmed it by publishing studies of real life performances? I cannot disagree with bigshot's assertion on this point, there is not the science to contradict it and it's in line with my professional experience and those with whom I've worked for more than two decades.

How is 16bit with noise-shaped dither, theoretically 120dB or so of dynamic range, not enough for any real life music recording/reproduction? And, if noise-shaped dither has not been applied to a particular master, why do you think that is?

G


----------



## bigshot (Nov 16, 2017)

amirm said:


> Where is the research and data to back this?



You want a scan from a book taken completely out of context or some fudged ABX test results? I'm afraid that ain't my style!

I've supervised sound mixes. I know how the system of recording and mixing works. More dynamics doesn't make for a better mix. A proper balance of dynamics does. Sound engineers want to create a mix that stays within the ear's natural dynamic range *without having to take a couple of minutes to let them adjust to a new dynamic range. *Human ears can hear about 45dB of dynamic range *at a time.* If you want to push beyond that, you have to give the ears a couple of minutes to adjust to the change. You can't just say, "I've been working at peak level and now I want to put some quiet detail at -70dB." No one will hear it. Music is recorded and mixed for human ears, not oscilloscopes and SPL meters. That means you work within the limits of human hearing to create a balance that is comfortable to listen to.

We're talking about recorded MUSIC, not abstract numbers on a page. There are aesthetic issues involved that far surpass the technical ones in relative importance. Before you raise the argument... Yes, creative decisions CAN be made in an objective manner. The biggest way that's accomplished in a recording studio is by considering the listener and creating a carefully controlled pace of contrasts to keep things interesting without being jarring or blowing out the listeners' hearing. I can tell you quite clearly that for the purposes of listening to recorded music in the home, your figures are all wrong. The simplest way to prove that is to take your favorite SACD and put it on your home stereo and turn the volume up to a peak level of 120dB. Even assuming that your living room has a dead silent noise floor and you can hear all the way down to 0dB (which it isn't and you can't), 120dB is not something you'll be able to tolerate for more than a minute or two. Go ahead and try to listen to the whole album. I bet you five bucks you can't.

Your figures do not reflect any sort of understanding of what we are talking about here... _shopping for a player and speakers or headphones to listen to music in our own living room._ You have some idea in your head that unless you make sure that an amp or DAC can perform properly with a test tone in an anechoic chamber (or in deep space!) it's inadequate. Sorry. That is audiophool thinking. There's a point where better numbers on the page don't result in better sound for ears. Continuing to double down on crossing every T and calculating pi out another fifty decimal places and taking into account every extreme circumstance just becomes ludicrous. You've gone so far over that line, it seems that you have no concept of what the purpose of reviewing home audio equipment is all about any more.

I've already said that I don't trust your tests. I think you enjoy skewing the truth to "prove" your _absurdism ad extremum._ I also don't trust your judgement. I don't think you have any sense of proportion about what is important and what is totally irrelevant. I guess what I'm saying here is that you talk and talk and nothing of any value comes out. Feel free to keep doing that if you're enjoying yourself, but don't kid yourself into thinking what you're doing is useful to anyone in any practical way.


----------



## RRod

@amirm Would you agree that where you set your pot matters for all this? For example, I have been able to ABX 16 vs 14 bits *without* shaped dither using open cans in a quiet bedroom on orchestral material. But if I try the same thing with a metal track, I won't be able to do it, because where I put my pot for the metal track can be > 20dB lower! Now, if I had a room that was 15dBA instead of 35 dBA, maybe I'd have a shot. Add in noise-shaped dither, and what chance do I have? Truncation errors in fade-outs aren't some mysterious thing: turn up the volume enough and you'll hear them. The question is whether that volume is a listening volume or not, and that depends on the material.


----------



## bigshot

It is really easy to detect the difference between 16 bit and 24 bit. All you do is clip a tiny bit of the fade out just before it goes into silence and turn the volume up real loud. Am I right, amirm?


----------



## Cutestudio

gregorio said:


> 1. You can't put a figure on it but typically, in the case of popular music, it would be way, way more than 6dB. Madonna's original mix would have had way more than 6dB of compression to start with, even before the original mastering, let alone after the remastering. And no, there's no way to know how much compression has been applied without access to the original recorded, unprocessed tracks. Audiophiles have got to get away from the notion that compression and distortion are bad things, on the contrary, they are essential and have been for 50 years or so. Do you really want all the guitar solos in rock and pop music to sound like a virtually inaudible series of very short duration twangs, because that's what an electric guitar sounds like without compression and distortion? And what about all the other elements/instruments in the mix? Without compression and distortion the popular/rock music you listen to would not sound anything like pop or rock music, so please stop with the "we've lost the MSB" because we've lost a whole lot more than the MSB and gained a far greater amount from that loss!!


Your literal interpretation of 6dB is rather sad as you have entirely missed the point being made by a person who looks at them almost daily. Well recorded pop music tends to have a dynamic range of around 16dB and no clipping and no visible (rough) flat tops, poorly mastered and remastered pop music tends to have a dynamic range of < 10dB and clipping or visible (rough) flat tops. The difference between good and bad is around 6dB and can be seen by comparing a remaster to an original well mastered track, something you have clearly failed to a) take on board or b) look at for yourself. The difference is about 6dB, and consists of all of the peaks being cut off, which is equivalent of omitting the MSB from the digital word. You probably wouldn't know this because instead of looking you're here busy disagreeing with people. Go take a look: learn something.

No one expects the full dynamic range of a snare drum to be reproduced in their living room so lets stick to the reality of recorded music instead of claiming that somehow I said all compression was bad. I didn't. No one did. EXCESSIVE COMPRESSION IS BAD BECAUSE THE MUSIC JUST SHOUTS AND IS WEARING TO LISTEN TO BECAUSE EVERYTHING IS ALWAYS TOO LOUD. Do you understand?



gregorio said:


> 2. Nope, even the OP discussed this. The number of quantisation values doubles for each bit of data: 16bit = 65,536 values, 17bit would therefore = 131,072 and 24bit = 16,777,216. So the difference between 16 and 24bit is obviously not 256 potential values but 16,711,280!


Skim reading my posts as you decide the best way to argue against them causes this very type of reading comprehension mistake - as self appointed arbiter of truth do you really think me, a programmer of over 30 years doesn't know the difference between the max values of a 16 and 24 bit word? Seriously? READ the post, I clearly state that below the LSB there is 256 extra values, because each step of a 16bit word is represented by 256 steps in 24 bit, because - duh - it's *8* bits longer. To scale a 16bit word to a 24 bit word one multiplies by 256 (or shifts right by 8 bits) which means the very last bit - bit0 - the LSB, now has a value of 256, giving it 256 potential values. Exactly as I said.
[/quote]



gregorio said:


> 3. Again, no! You've made a good argument for a processing environment to be greater than 16 bits but not the distributed audio files themselves. This is why today's pro mixing/processing environments are typically 64bit float and even going back around 20 years they were 32bit float or 48bit fixed.
> G


Your obsession with disagreeing s curious, what's up with that? I was not discussing studio mixing here, or even distribution in 24bit: did you actually read any of my post properly? You are failing to have a discussion here, you are merely imposing your erroneous preconceptions onto the thread, having the biggest ego is one thing, the desperation of showing it is another.


----------



## castleofargh

bigshot said:


> It isn't even an issue with dynamic recorded music. The most dynamic recorded music doesn't exceed a dynamic range of 50dB or so, and music is usually normalized to a few dB of the zero line. With redbook, that leaves a whole lot of room to play "hide the noise floor". I've done a lot of mixes where we worked at 24 bit and then bounced down to 16 at the end. We carefully compared the 24 bit mix to the 16 bit bounce down each and every time. Never a difference. That's on studio equipment with a room full of critical ears doing the checking. 16 bits is totally sufficient to cover the dynamics in any recorded music.
> 
> 
> 
> ...


we're on a forum where anybody can start a discussion about anything he likes(within TOS). you and I are on the more practical end of the spectrum, some are very passionate about the most that can be achieved, and some are even interested in theoretical science. the place is big enough to accommodate everybody.





amirm said:


> Let me state my position without the back and forth as to make it more clear.
> 
> It is important in the context of audio "science" that we are true and correct to what that is.  Paper after paper from luminaries in audio show that 16 bits is insufficient dynamic range (without noise shaping) to be transparent to listeners.  There is also research that shows high resolution content resampled to 44.1 Khz can have audible consequences.  If you are going to jump up and down and say this isn't so, please don't bother unless you have research you can put forward to the contrary.  Or controlled listening tests you have performed.  Otherwise, it really is useless to give me anecdotal information about what you think or hear.  That is not material.
> 
> ...


 if I need training to notice something, then was it audible for me before the training? I can't do a backflip so I find it false to claim that I can. but of course if I trained, I could surely do one in a matter of days. still it's a little silly to assume I can do it before I can.
audibility is the same. just because once you bring up the most extravagant conditions including training, something becomes noticeable, doesn't mean we have to agree that it is noticeable anytime we're outside of your conditions. which is pretty much always for a dude just enjoying a song.


----------



## amirm

gregorio said:


> I cannot disagree with bigshot's assertion on this point, there is not the science to contradict it and it's in line with my professional experience and those with whom I've worked for more than two decades.


I am sorry but I don't know how any of this translates to a precise measurement of dynamic range of music we all have.  If it is a measurement, then you or he should be able to explain how you arrived at it.  Since I know there is no tool to make such a measurement, I hope you appreciate my skepticism on that.

The way we all can easily determine if our music has 50 db or less of dynamic range is to convert it to 8 bits.  8 bits gives us 8*6+1.76 = 50 db of dynamic range. 

I took a random high-res track from my library, the "Broadway" track from David Chesky's Jazz in the Harmonic album, which is in 32 bit format at 192 Khz.  I kept the sampling rate the same but converted the bit depth to 8 bits using TPDF dither.  I then did a double blind ABX test of the two using my laptop default headphone output and my Etymotic ER4S IEMs.  Here are the results:

foo_abx 1.3.4 report
foobar2000 v1.3.2
2017/11/16 11:27:05

File A: C:\Users\Amir\Documents\Test Music\Chesky 8 bit Test\Original 192 32 bit file.wav
File B: C:\Users\Amir\Documents\Test Music\Chesky 8 bit Test\Converted 192 8 bit file wit TPDF dither.wav

11:27:05 : Test started.
11:27:19 : 01/01  50.0%
11:27:27 : 02/02  25.0%
11:27:34 : 03/03  12.5%
11:27:40 : 04/04  6.3%
11:27:45 : 05/05  3.1%
11:27:50 : 06/06  1.6%
11:27:54 : 07/07  0.8%
11:28:00 : 08/08  0.4%
11:28:05 : 09/09  0.2%
11:28:12 : 10/10  0.1%
11:28:17 : 11/11  0.0%
11:30:39 : Test finished.

 ---------- 
Total: 11/11 (0.0%)

As you see, not only did I get 11 out of 11 answers right, but I did so at lightning speed.  There is only 4-5 seconds for each trial which includes listening and voting to go to the next trial.  In other words, it is 100% and clearly audible increase in noise floor.

Did you ever perform such a test and failed to hear such a drastic difference?  This is so easy to tell that everyone here should be able to hear it.

So no, I don't think Bigshot is right.  He has done no verification of what he is saying, nor is his data based on any kind of research.  It is just based on simplistic assumptions that may feel right, but are just incorrect.


----------



## bigshot

If simplistic is a synonym for practical experience, I’ll agree


----------



## amirm

bigshot said:


> If simplistic is a synonym for practical experience, I’ll agree


No, it means lay assumptions that are incorrect as a matter of audio science and research -- both medical and enjoyment fields.


----------



## amirm

castleofargh said:


> if I need training to notice something, then was it audible for me before the training? I


I don't know if you do or do not need training.  I don't know that you know if you do or do not either.   I have had people with no training beat me and beat me good in hearing audio impairments.  When I write something in forums I have to include the possibility of such people being readers of what I say.  I can't just exclude them and worse yet, say audio science says they don't exist.

And let's put all of this in context.  Who on earth are we protecting here?  Almost all of us have the capability to play at > 16 bits /44.1 Khz.  There is no royalty or any cost associated with playing high resolution content.  So what is the purpose here?  A cause célèbre to make a name for ourselves on forums while we sacrifice what we know in research?


----------



## bigshot

Compression in pop music is one part stylistic and one part practical. The attitude of “play it loud” is a big part of a lot of pop music, particularly hip hop. Pop music is designed to be played on kids’ phones on trains and busses.

Other types of music have different styles of recording and practical compromises. Classical music is the opposite side of the spectrum, and it can have as much as a 50dB range.

No one disagrees that hot mastering is an overreaction. There’s no excuse for clipping. But compression is often a creative choice. Compression is not necessarily the same thing as loudness wars.


----------



## Cutestudio

71 dB said:


> Loudness war is a commercial thing. People with more sophisticated music taste buy more music which does not suffer from it.


Amazing. I have no words. Is this really a HiFi forum?



71 dB said:


> We defend 16 bit digital audio in this thread.


No we don't, some of us are acutely aware of its limitations.
Additionally the defence of 16bit is irrational and has no purpose. Digital has moved on. Welcome to 2017.


----------



## bigshot

I train myself to understand and appreciate a wide range of music. That’s like learning a language. The more you hear and process, the deeper your understanding and more fluent you become. Training your ears instead of your brain just involves turning up the volume at the right spot. It may help you detect things, just like using a magnify glass does. But it doesn’t improve your understanding of music at all. It’s a mechanical exercise, not an intellectual one.


----------



## Cutestudio

bigshot said:


> Compression in pop music is one part stylistic and one part practical. .



If that was true then the remasters would not lose 6dB of dynamic range over the originals.
But they do.

So your statement has no bearing. You appear to be blissfully unaware of what the Loudness War actually is in practical terms on a digital waveform level. That's fine, not many people are, but please stop pretending that you do. Amirm and one or two others have got it, but there seems to be a camp fighting a corner based on a similar method to one employed by the Ostrich.


----------



## bigshot

amirm said:


> No, it means lay assumptions that are incorrect as a matter of audio science and research -- both medical and enjoyment fields.



Post some charts and diagrams related to enjoyment. That is actually my profession- to entertain. I’m sure you have some fascinating insights into that!


----------



## bigshot

Cutestudio said:


> If that was true then the remasters would not lose 6dB of dynamic range over the originals.
> But they do.



I’d be happy to explain that to you. When a record is mixed, the balances are judged in a best case scenario- on the calibrated monitors of a professional recording studio. In the early days of CDs, mastering for the limitations of LPs was no longer necessary. And the people who owned CD players were mostly people with nice stereos. The market for CDs in the 80s skewed older and more affluent than the market for digital music today. Mastering engineers today are working to get around the limitations of the equipment their customers use. Ear buds and cell phones require a different sort of mastering than high end stereos do. CD sales are plummeting. Streaming is where the market is. Mobile is where they’re listening. That means more compression is desirable. It may not suit your purposes as well, but audiophiles are now a niche market. There are specialty retailers for that. This is what I meant by practicality. You serve your demographic.


----------



## TheTrace

Cutestudio said:


> Amazing. I have no words. Is this really a HiFi forum?
> 
> 
> No we don't,* some of us are acutely aware of its limitations.*
> Additionally the defence of 16bit is irrational and has no purpose. Digital has moved on. Welcome to 2017.


What are those limitations?


----------



## 71 dB

bigshot said:


> Compression in pop music is one part stylistic and one part practical. The attitude of “play it loud” is a big part of a lot of pop music, particularly hip hop. Pop music is designed to be played on kids’ phones on trains and busses. Other types of music have different styles of recording and practical compromises. Classical music is the opposite side of the spectrum, and it can have as much as a 50dB range.



Agreed. I do listen to commercial pop music because it's fun, uplifting, energizing and easy to listen to. I know it is not dynamic music and I accept that. Despite of limited dynamic range, modern pop music can be surprisingly sophisticated for it's sound design and finetuned to perfection. Sometimes however I want to hear intellectual dynamic music and pop is changed to for example classical music or perhaps some marginal non-commercial electric music.



Cutestudio said:


> No we don't, some of us are acutely aware of its limitations.
> Additionally the defence of 16bit is irrational and has no purpose. Digital has moved on. Welcome to 2017.



The purpose of defending 16 bit audio is to make people aware of how additional bits don't make a difference in sound quality. The only reason why digital has moved on is money. 24 bit files make it easier to milk ignorant people once again...


----------



## 71 dB

TheTrace said:


> What are those limitations?



You can't record butterflies between 270 kW pumps making 119 dB of A-weighed noise.


----------



## bigshot

The biggest limitation of CD sound is that it isn't hip with the cool kids. 16/44.1 is SO 1982! Get with the times! Double the numbers is double the fun!


----------



## TheTrace

71 dB said:


> You can't record butterflies between 270 kW pumps making 119 dB of A-weighed noise.





bigshot said:


> The biggest limitation of CD sound is that it isn't hip with the cool kids. 16/44.1 is SO 1982! Get with the times! Double the numbers is double the fun!


Lmfaoooo


----------



## TheSonicTruth (Nov 16, 2017)

bigshot said:


> I’d be happy to explain that to you. When a record is mixed, the balances are judged in a best case scenario- on the calibrated monitors of a professional recording studio. In the early days of CDs, mastering for the limitations of LPs was no longer necessary. And the people who owned CD players were mostly people with nice stereos. The market for CDs in the 80s skewed older and more affluent than the market for digital music today. Mastering engineers today are working to get around the limitations of the equipment their customers use. Ear buds and cell phones require a different sort of mastering than high end stereos do. CD sales are plummeting. Streaming is where the market is. Mobile is where they’re listening. That means more compression is desirable. It may not suit your purposes as well, but audiophiles are now a niche market. There are specialty retailers for that. This is what I meant by practicality. You serve your demographic.




"_Mastering engineers today are 
working to get around the limitations 
of the equipment their customers use. 
Ear buds and cell phones require a 
different sort of mastering than high 
end stereos do. CD sales are plummeting. 
Streaming is where_-"flush!!!

I wonder why CD sales are plummeting and why  LP sales are resurging. Hmmmm. And it has little to do with streaming services, and everything to do with the more capable format(redbook digital audio) getting the short shaft in the mastering suite - regardless of who calls the shots or pays the bill.

And I also see what side you're on, denying that "content is where it begins", when content is everything! 

I also have yet to hear any difference between 16 and 24bjt versions of the same recording, unless some monkey at the controls of the mastering DAW creates a difference - internally or with plugins to the daw.

Compress on, bigshot, brickwall on, squash on!    I've already updated 90% of my CD collection, replacing excuses for remasters with original/first-run CDs.  Why?  Because they sound better!  Not just because of the higher DR values, etc.

Next time you're sitting in the porcelain library, feast your eyes here:

https://m.facebook.com/2016SaveOurMusicNoRemasters/?ref=content_filter


----------



## amirm

bigshot said:


> Training your ears instead of your brain just involves turning up the volume at the right spot.


No, that is not remotely training.  Training is the same as being a critical listener.  At the same volume and conditions I can hear distortions that others cannot due to that training//ability.

What you describe is a tool.  There is a difference between that and skill.

I realize this is a foreign concept for you so think of a more ready example: people who taste ice cream, beer, hot sauce, wine, etc. at the end of an assembly line.  They know what to taste for.  They don't rely on some other than you and I would for their jobs.

Ultimately unless you apply yourself and learn what training is about, and develop ability to hear audio impairments, you will be lost in this topic with no intuition.


----------



## bigshot (Nov 16, 2017)

A critical listener understands music and how it is structured and balanced to optimize clarity and expressiveness. Just listening for noise and artifacts doesn't require training. That just requires paying attention to stuff like that. If you're going to wrap your ego around passive perception then make sure your brain and ideas are involved. Focusing on schmutz isn't thinking and you shouldn't expected to elicit awe from those around you for that.


----------



## TheSonicTruth

bigshot said:


> The biggest limitation of CD sound is that it isn't hip with the cool kids. 16/44.1 is SO 1982! Get with the times! Double the numbers is double the fun!





TheTrace said:


> Lmfaoooo




I know.  I really crave over-compressed heavily limited DR7 music in a high res format.


----------



## bigshot (Nov 16, 2017)

TheSonicTruth said:


> I wonder why CD sales are plummeting and why  LP sales are resurging. Hmmmm. And it has little to do with streaming services, and everything to do with the more capable format(redbook digital audio) getting the short shaft in the mastering suite - regardless of who calls the shots or pays the bill.



I think you're confusing the result with the cause there. People are embracing streaming because it's more convenient for them. They haven't stopped buying CDs because they sound bad. They've stopped buying them because a more convenient format has replaced it. Also, if you look at my post, you'll notice that I was referring to pop music when I talked about compressing for the audience's intended use. Classical and Jazz are still mixed and mastered pretty much the same as they always have been. In fact, the introduction of multichannel sound has ushered in a whole new level of audio quality.

This really isn't anything new. In the days of record albums you could buy two kinds of records... LPs and 45s. The LPs were mixed and mastered for listening on a good speaker system in the home. 45s were designed for jukeboxes and radio airplay. Each of them had their own mixing and mastering to suit their intended purpose. 45s were generally more compressed for radio airplay and they had fuller bass for jukeboxes. The higher speed of 45s gave them an edge in the bass and the slightly wider grooves allowed for more volume. Some groups like The Beatles in their later albums sounded better on LPs. Others like the Rolling Stones were better served by 45s. It's all a matter of matching the engineering to the purpose of the music.

I realize that everyone wants the music industry to serve their own personal interests, but music is a business and it serves the market. Audiophiles aren't the market. They're a niche. I know I do my part to support the market for that niche, but I'm just one person.


----------



## gregorio

amirm said:


> [1] I am sorry but I don't know how any of this translates to a precise measurement of dynamic range of music we all have.
> [2] I took a random high-res track from my library, the "Broadway" track from David Chesky's Jazz in the Harmonic album, which is in 32 bit format at 192 Khz.



1. It doesn't and that's my point! There is no precise definition of dynamic range and therefore there cannot be a precise measurement of it. This is true of a lot of what we perceive when we listen to music/audio, we cannot measure soundstage for example. Musicians define dynamic range as the quietest note to the loudest note. You appear to be defining it quite differently, the highest peak transient level to the digital noise floor.
2. Case in point! I'm sure we could hear the noise floor of 32bit, if we concentrated only on some latter part of a fade to digital silence and amplified it greatly. However, leave the amplifier at that level, play the whole track and if you had a system which could actually reproduce that dynamic range (which obviously you don't) then it would literally kill you! Assuming an amp and speakers/headphones add zero noise, a 0dB listening environment, a 0dB recording environment, mics and mic preamps which add zero noise, the instruments and musicians make zero noise and instruments which can actually produce such a dynamic range, then we can only reproduce a maximum of 21 bits of dynamic range any way. What are the extra 11 bits for? And of course, all those qualifying assumptions simultaneously are hardly "real life".

I notice you ignored every single one of the 4 questions I asked, which seems to be a trend in your responses!



amirm said:


> [1] Who on earth are we protecting here?  Almost all of us have the capability to play at > 16 bits /44.1 Khz.
> [2] There is no royalty or any cost associated with playing high resolution content.  So what is the purpose here?



1. I'm not sure if my system will output 120dBSPL at my listening position, I've never tried and would never want to!
2. Really, all your hi-res content cost the same as the 16/44 versions? I don't think that matches the experience of most here.



amirm said:


> As I just explained, I ask for 24 bits because that is what is used to create the music in the first place.



No, it's not! Typically today and for a number of years it's created at 64bit float and before that it was typically created in 32bit float or 56bit fixed. I don't know of a time music was created in 24 bits but it must have been more than about 20 years ago.



Cutestudio said:


> Your literal interpretation of 6dB is rather sad ...



So you quote 6dB and the MSB all over the place and then say it's sad I interpret your statements literally as 6dB? How else should I interpret a figure you've repeatedly quoted? Are you saying that when you state 6dB I should interpret that as anything from say 3dB up to 30dB?

G


----------



## 71 dB

gregorio said:


> No, it's not! Typically today and for a number of years it's created at 64bit float and before that it was typically created in 32bit float or 56bit fixed. I don't know of a time music was created in 24 bits but it must have been more than about 20 years ago.


I think *amirm* means the tracks that are imported to DAW are 24 bit, not the internal processing bit depth of a DAW.


----------



## Cutestudio

bigshot said:


> I think you're confusing the result with the cause there. People are embracing streaming because it's more convenient for them. They haven't stopped buying CDs because they sound bad.


I think you're confused.
Since I discovered that CDs and digital downloads/streaming (the media is irrelevant) were engineered with horrific distortion built in I stopped about 95% of my new purchases and started instead scouring the second hand and foreign markets for less mangled product.
Apple knew this too which is why they spent millions on 'Mastered For iTunes' so they could sell non mangled product, but hey, what does Apple know eh?

Selling mangled, clipped and highly distorted product as CD quality is simple fraud and has obviously impacted sales. I know of many people who won't touch new masters with a barge pole. What really puzzles me however is your appearance on a HiFi forum to justify the selling of these fraudulant LoFi product which clearly is 'Not As Advertised' and arguably break a number of consumer laws related to merchantable quality and expectations of the quality of the waveforms within them.

Today it's impossible to buy most modern pop without severe clipping distortion and a dynamic range (Ratio of Peak to RMS in this case) of around 10dB. Tracing the original non-remastered music reveals the missing 6dB odd of signal that has been butchered. 



bigshot said:


> Classical and Jazz are still mixed and mastered pretty much the same as they always have been. In fact, the introduction of multichannel sound has ushered in a whole new level of audio quality.


Norah Jones's The Fall is jazz/blues and probably has the worst clipping damage of any modern tracks today.
The 'whole new level' of audio quality on multichannel sound is due to a simple lack of attention (and therefore mangling) of the source.



bigshot said:


> This really isn't anything new. In the days of record albums you could buy two kinds of records... LPs and 45s. The LPs were mixed and mastered for listening on a good speaker system in the home. 45s were designed for jukeboxes and radio airplay.



You are not listening. The existence of the 45rpm single cannot and does not rationalise mastering damage to digital music 50 years later.
Radio today have a bank of compressors and have NO NEED for a compressed source. Think about it: They have a guy shouting straight into a microphone every 10 minutes, why on earth would they need the digital tracks pre compressed?? You think they can't compress and limit themselves LOL?

Many car stereos today have features to repair poor quality digital music, if the record industry had released a non-damaged product they'd have put a compression setting there instead. As it is the choice now is between 'mangled' and 'heavily mangled' in the car.



bigshot said:


> It's all a matter of matching the engineering to the purpose of the music. I realize that everyone wants the music industry to serve their own personal interests, but music is a business and it serves the market. Audiophiles aren't the market. They're a niche. I know I do my part to support the market for that niche, but I'm just one person.


Compression and clipping is turning people away from music in droves. It's now a commodity. The self-harm of the record companies actually spreads MP3 pirating because people today can't tell the difference between mangled lossless and mangled lossy music, indeed encoding through mp3 may even smooth some of the worst clips out.

Makers of MP3 players would be free to add compression and limiting as required to their devices, the blanket ruining of the sound lowers everyone to the same low standard, Audiophiles are the canary that is now dead in the cage. Your claim to be on the side of audiophiles is belied by all of your posts which push the mangled product in 16bit format and tell anyone who objects that they are wrong and unimportant.

The record industry has serious issues with quality of product that in my view is fraudulent misrepresentation and harms their own business model. When I see people defend this practice on HiFi forums I realise that sound quality will continue to fall and the companies will eventually go bust. Already bands have been reduced to vocals and lyrics only with young talentless singers and part of this is due to the mangling of the product, the instruments have no character, dynamics are ironed out and all we have left is a dumb 12 year old singing vacuous dirges for radio muzak.

HiFi used to be something people aspired to because it sounded good. The industry was never that careful about the source but the novelty of the CD and the old engineering skill to avoid clipping lasted for quite a few years until todays Master Manglers. With EVERYTHING TURNED UP TO A CONTINUOUS NOISE SO AN ACOUSTIC GUITAR STRING IS THE SAME LEVEL AS A SNARE DRUM PEOPLE DON"T NEED TO BOTHER WITH HIFI OR MUSIC COLLECTIONS, I MEAN WHAT WOULD BE THE POINT WHEN ALL THE MUSIC IS THE SAME MANGLED WALL OF NOISE THAT NO ONE CARES ABOUT ANYMORE?

Have you got it yet?
You are defending the slow suicide of the entire HiFi and music industry on a HiFi forum.
Why?


----------



## reginalb

TheSonicTruth said:


> ...I wonder why CD sales are plummeting and why  LP sales are resurging...



Because hipsters like turntables. Why do you think one of the big sellers of vinyl is Urban Outfitters? It has absolutely zilch with the mastering on CD's.


----------



## TheSonicTruth (Nov 17, 2017)

Cutestudio said:


> I think you're confused.
> Since I discovered that CDs and digital downloads/streaming (the media is irrelevant) were engineered with horrific distortion built in I stopped about 95% of my new purchases and started instead scouring the second hand and foreign markets for less mangled product.
> Apple knew this too which is why they spent millions on 'Mastered For iTunes' so they could sell non mangled product, but hey, what does Apple know eh?
> 
> ...



Why?  Because he works in the business, turning what his clients are paying him to.  If I were in bigshot's shoes, I'd be collecting government support because as an engineer I would not be willing to destroy music that way for the sake of sheer loudness!  Why do you think M.E. Bob Katz has such a 'select' clientele?  Because he won't turn out and sign off on sh_t, that's why!  I'm proud to say I shook Katz' hand at the AES convention in New York a couple years ago.  I'd love to sit him and bigshot down in a room together and see what transpires.

If you, as an engineer, know that doing certain things to a music project is bad, and you demonstrated that to your client, and you do it anyway, you are just as implicit as that client in promoting loudness and distorted clipped GARBAGE.

And for his information, rips from my first generation classic rock, jazz, classical, and rap CDs sound just fine and 'translate' well on my portable players without any additional limiting or compression to make them louder(remastering), thank you very much!


----------



## TheSonicTruth

reginalb said:


> Because hipsters like turntables. Why do you think one of the big sellers of vinyl is Urban Outfitters? It has absolutely zilch with the mastering on CD's.




Can you prove that?  I get lots of customers on a daily basis who say 'nothing, not even CD, sounds quite like vinyl'.    You must be a squasher-ehem, cough! - 'mastering' engineer also.  (facepalm!)


----------



## gregorio

71 dB said:


> I think *amirm* means the tracks that are imported to DAW are 24 bit, not the internal processing bit depth of a DAW.



The channels are in 24bit format but of course the actual recorded material is far fewer bits than that. After the initial tracking, all the subsequent processing, balancing and mixing occurs in the 64bit mix environment though, so the mix is 64bit float but of course we can't export that.



Cutestudio said:


> With EVERYTHING TURNED UP TO A CONTINUOUS NOISE SO AN ACOUSTIC GUITAR STRING IS THE SAME LEVEL AS A SNARE DRUM



You seem to be missing the point and then hilariously stating that others don't understand your posts. An acoustic guitar string is significantly louder than an electric guitar string but I don't hear anyone complaining when an electric guitar is massively distorted, compressed and brought up to the same level as a snare drum. No one here, including bigshot, is arguing for the loudness war or massive over-compression, some of us have been arguing against it far longer than you. It's been an issue since well before pirated MP3s were common and has been very popular with the vast majority of consumers, which is why record companies continued to do it and why it didn't just die to start with! The problem is that it's impossible to quantify what is over-compression, a perfectly acceptable amount of compression for one song might be ridiculous over-compression in another. Some entire genres require several applications of heavy compression, others virtually none whatsoever. And incidentally, Mastered for iTunes does NOT stipulate appropriate amounts of compression or directly affects/combats the loudness war.

It seems you yourself have relatively little understanding of the issue and are confused, best then not to accuse others of what you are guilty of!

G


----------



## reginalb

TheSonicTruth said:


> Can you prove that?  I get lots of customers on a daily basis who say 'nothing, not even CD, sounds quite like vinyl'.    You must be a squasher-ehem, cough! - 'mastering' engineer also.  (facepalm!)



All I have to do is look at how many exclusives on vinyl they get at Urban Outfitters. I am not an engineer, and attached no value judgement to my statement, so calm yourself down. I understand business, and all you have to do is look at how vinyl is marketed to understand where the sales are happening. 

In 2016, vinyl record sales hit a 25 year high, with 3.2 million sold. Spotify alone has 100 million users and 30 million paid users. Apple has another 30 million users for Apple Music, Deezer has over 3 million, Tidal is north of a million users, And Google Play Music is somewhere in there as well, though they don't say how many they have. So, you're talking ~140 million people streaming music, vs. 3.2 million records sold in a year. Since the average owner will buy more than one record, you're looking at a number of people actively using records in the hundreds of thousands. 

I'd wager the evidence points to streaming as the future of music. And it makes sense. With Play Music All Access, I can often choose from multiple masters, I can upload for streaming 50,000 of my own songs, and have access to a vast library of music for a monthly cost instead of having to buy individual albums. For me, that does mean vinyl - since I have more "music money" leftover. I own a turntable, but it's because I think it's fun to collect and use records, not because I've deluded myself in to thinking that it's somehow better than other formats. 

I also get my hands on multichannel mixes whenever I can, I made sure to get a Blu-ray player that supports SACD, so that I can play any type of multichannel audio I come across. Unlike other audiophiles, though, I don't pretend that my preferences are the same as the market as a whole. They just aren't.


----------



## 71 dB

Cutestudio said:


> HiFi used to be something people aspired to because it sounded good. The industry was never that careful about the source but the novelty of the CD and the old engineering skill to avoid clipping lasted for quite a few years until todays Master Manglers. With EVERYTHING TURNED UP TO A CONTINUOUS NOISE SO AN ACOUSTIC GUITAR STRING IS THE SAME LEVEL AS A SNARE DRUM PEOPLE DON"T NEED TO BOTHER WITH HIFI OR MUSIC COLLECTIONS, I MEAN WHAT WOULD BE THE POINT WHEN ALL THE MUSIC IS THE SAME MANGLED WALL OF NOISE THAT NO ONE CARES ABOUT ANYMORE?



I hope you get your caps lock fixed. Meanwhile please look at the waveform below. It's a track from a CD opened in Audacity. You probably think it must be a CD from year 1985 or so, because it looks so dynamic and there's no sign of "brickwall compression" at all. This is a CD that was recorded in 2010-11 and released in 2011. It's not the most dynamic "newer" CD I have. It's the latest CD I bought. The track is the shortest one on the CD and perhaps not the most dynamic one. This CD won't sell millions of copies like Norah Jones does, but a few thousands. It's not very commercial. It is Joseph Schwantner's orchestral music on Naxos (8.559678). So, just to show you that there are dynamic new CDs out there if you bother the look beyond the most popular artists/music.


----------



## Darren G

One of the first CD's I bought as an early adopter was the 1812 Overture.  The CD came with a warning sticker, the cannons are loud, be careful.

So I had my horn speakers, > 100db sensitivity rating, and yes they can play loud.   If I had any real gripe with the CD it would have been that most of the recording was mastered down there at the lowest significant bits to leave room for the cannons, and my first generation CD player's DAC was not so great.

Gads, what a mess when the cannons fire.  My speakers, that I believed to be quite competent, distorted to an extreme, and there was nothing enjoyable about it.  My ears literally hurt.  I didn't measure the absolute SPL, but the point is I simply don't listen at those levels anymore.  These days it is extremely rare I listen at even 80db.    Once or twice a year I'll go listen to live music, or a DJ playing music at levels that leave my ears ringing, but rarely.


----------



## TheSonicTruth (Nov 17, 2017)

reginalb said:


> All I have to do is look at how many exclusives on vinyl they get at Urban Outfitters. I am not an engineer, and attached no value judgement to my statement, so calm yourself down. I understand business, and all you have to do is look at how vinyl is marketed to understand where the sales are happening.
> 
> In 2016, vinyl record sales hit a 25 year high, with 3.2 million sold. Spotify alone has 100 million users and 30 million paid users. Apple has another 30 million users for Apple Music, Deezer has over 3 million, Tidal is north of a million users, And Google Play Music is somewhere in there as well, though they don't say how many they have. So, you're talking ~140 million people streaming music, vs. 3.2 million records sold in a year. Since the average owner will buy more than one record, you're looking at a number of people actively using records in the hundreds of thousands.
> 
> ...




As long as you don't confuse me with an audiophile.  The very root of that word is 'lover of sound' as, in love with how something sounds on a piece of equipment, not necessarily with the music or artist themself.   I want the music to sound like the music, not just to sound good on a particular chain.

 As far as both multi-channel and high res go, they are mere gimmicks to me, until artists know what a good album sounds like, not all squashed and regained up to twelve!

Enjoy your millennial streaming habits.  I prefer a physical collection at home, where I can weed out the 'remasters'(both CDs and video) and control my content.  With streaming, I can't control the quality as regularly, and until streaming gets up to the rates of physical digital audio, it will continue to be a second fiddle, an auditioning stage for me, to decide what I want in physical format.


----------



## TheSonicTruth (Nov 17, 2017)

Darren G said:


> One of the first CD's I bought as an early adopter was the 1812 Overture.  The CD came with a warning sticker, the cannons are loud, be careful.
> 
> So I had my horn speakers, > 100db sensitivity rating, and yes they can play loud.   If I had any real gripe with the CD it would have been that most of the recording was mastered down there at the lowest significant bits to leave room for the cannons, and my first generation CD player's DAC was not so great.
> 
> Gads, what a mess when the cannons fire.  My speakers, that I believed to be quite competent, distorted to an extreme, and there was nothing enjoyable about it.  My ears literally hurt.  I didn't measure the absolute SPL, but the point is I simply don't listen at those levels anymore.  These days it is extremely rare I listen at even 80db.    Once or twice a year I'll go listen to live music, or a DJ playing music at levels that leave my ears ringing, but rarely.




Stick a pair of 6 or 12dB Harrison Labs attenuators into the CD/Aux inputs of your amp or receiver, like I did, and you'll wonder why you never thought of doing so.  CD listening for me has never been more objective and enjoyable since buying them, and the attenuators add nothing to change the sound, other than that you need to set the volume a bit higher to achieve your usual listening level.

I am comfortable that nothing, at least inside my receiver, is clipped or distorted.  I cannot, however, speak for the DACs in my player, when presented with a modern hot-mess mastered CD.  I finally settled on 12dBs for the CD and phono, and 6db for audio out of my DVD and Bluray(yes, I still RCA out, and am _damned proud_ of it!)


----------



## Darren G

I will keep going to hear live music, because hey, I enjoy those those low bass notes without small room interference, but I've given up on home audio sounding like live.  Home audio sounds good these days, but there at least three (3) details no home audio experience can re-create (yet) -

1.) A really big room.
2.) I can turn my head and the sound adjusts in real 3-D as it should.
3.) The crowd energy.  Music is more enjoyable when others are enjoying it too.

Home audio is already so good that there is not much else to add, and a few smaller additions, still isn't as good as live.


----------



## bigshot (Nov 17, 2017)

TheSonicTruth said:


> Why?  Because he works in the business, turning what his clients are paying him to.  If I were in bigshot's shoes, I'd be collecting government support because as an engineer I would not be willing to destroy music that way for the sake of sheer loudness!



Have you ever worked in the business, or did you cut to the chase and go straight to the dole?

By the way, I can recommend many CDs that are well engineered and sound great. If your musical tastes can extend beyond pop, there's a whole world of great music and great sound to explore. I think you guys are listening to the wrong stuff. Complaining about pop music because it's not audiophile is like going to McDonalds and insisting on wine with your meal.



TheSonicTruth said:


> Stick a pair of 6 or 12dB Harrison Labs attenuators into the CD/Aux inputs of your amp or receiver, like I did, and you'll wonder why you never thought of doing so.



The problem with the Mercury 1812 Overture isn't just level and it isn't clipping either. It's that the cannon puts out peak level sub bass at a volume well above the full orchestra. It was recorded separately from the orchestra and bells, but when they mixed it, they put it in at a very realistic level. You have to have VERY good speakers and a very strong amp to avoid having the speakers spit through the sub bass. It's easier to play if you have a subwoofer designed to reproduce those super low frequencies very loud.



TheSonicTruth said:


> As far as both multi-channel and high res go, they are mere gimmicks to me.



"A mere bag of shells!"

Cutestudio, have you had a chance to check out that article in my sig yet?


----------



## bigshot

Darren G said:


> I will keep going to hear live music, because hey, I enjoy those those low bass notes without small room interference, but I've given up on home audio sounding like live.  Home audio sounds good these days, but there at least three (3) details no home audio experience can re-create (yet) -
> 
> 1.) A really big room.
> 2.) I can turn my head and the sound adjusts in real 3-D as it should.
> 3.) The crowd energy.  Music is more enjoyable when others are enjoying it too.



1 & 2 are perfectly possible if you have a good sized listening room and a 5.1 system. For 3, you have to invite a bunch of friends over!


----------



## Darren G (Nov 17, 2017)

Love your posts bigshot.  It's clear you love music, and that is what it's all about   Looking forward to reading your posts.

p.s. love the deep bass and energy some DJs and musicians pull off, and my home system just cannot do this.


----------



## bigshot

If you ever get to Los Angeles look me up and I'll demo my system for you. I can play you some stuff that makes you think you're at a live show. By the way, if you like Zappa get the new Halloween 77 CD. It's one of the best live recordings I've ever heard.


----------



## reginalb

Cutestudio said:


> Norah Jones's The Fall is jazz/blues and probably has the worst clipping damage of any modern tracks today.
> The 'whole new level' of audio quality on multichannel sound is due to a simple lack of attention (and therefore mangling) of the source.



The Fall is pop, sir. That's not an insult, it's one of my favorite albums, but it's certainly pop. With that said, if you like Norah and want less compressed versions, you can pick up the SACD box set from somewhere like Acoustic Sounds, it's mastered differently:

http://dr.loudness-war.info/album/list?artist=Norah+Jones&album=The+Fall (I provided the third measurement down, btw).


----------



## TheSonicTruth

bigshot said:


> Have you ever worked in the business, or did you cut to the chase and go straight to the dole?
> 
> By the way, I can recommend many CDs that are well engineered and sound great. If your musical tastes can extend beyond pop, there's a whole world of great music and great sound to explore. I think you guys are listening to the wrong stuff. Complaining about pop music because it's not audiophile is like going to McDonalds and insisting on wine with your meal.
> 
> ...



Pop stuff from  the 1980s and prior was dynamically compressed, but on an as-needed basis and certain not squashed and buzz-cut as stuff from after 2000 has been!   No wonder people think old vinyl sounds better!

You're right about a sub or bi-amped solution for that 1812 though.


----------



## TheSonicTruth

bigshot said:


> If you ever get to Los Angeles look me up and I'll demo my system for you. I can play you some stuff that makes you think you're at a live show. By the way, if you like Zappa get the new Halloween 77 CD. It's one of the best live recordings I've ever heard.



Thanks for the suggestions.  In general I avoid(but have almost mistakenly bought!!) 'live albums'.  Kiss 'Alive' and 'Frampton Comes Alive' are my two exceptions.  Otherwise, if I wanted live sound, I'd save up some dough and go to a concert.


----------



## bigshot (Nov 17, 2017)

TheSonicTruth said:


> Pop stuff from  the 1980s and prior was dynamically compressed, but on an as-needed basis and certain not squashed and buzz-cut as stuff from after 2000 has been!



Have you ever heard The Rolling Stones' Beggar's Banquet? There are songs on there that are massively compressed and clipped deliberately.

Kiss? Frampton Comes Alive?

wow. maybe the quality of the sound isn't the problem.


----------



## RRod

reginalb said:


> http://dr.loudness-war.info/album/list?artist=Norah+Jones&album=The+Fall (I provided the third measurement down, btw).



Heh, "Limited edition". This industry, man.


----------



## TheSonicTruth (Nov 17, 2017)

bigshot said:


> Have you ever heard The Rolling Stones' Beggar's Banquet? There are songs on there that are massively compressed and clipped deliberately.
> 
> Kiss? Frampton Comes Alive?
> 
> wow



Well in the case of 'Banquet' it doesn't need additional messing with for a so-called remaster.  Why the heck add compression on top of what's already baked in to the stereo master tapes??   Just rip to 24/192, check for channel balance, make minor EQ *as needed!*, dither down to 16/44.1 and reissue!

And I don't want to hear about how 'originals won't translate properly on modern playback devices' - that's a load'a hooey.  Those things have volume controls you know.


----------



## bigshot

They didn't remaster Beggar's Banquet. They remixed it and took all the life out of the deliberate compression and distortion choices on the original album. They ruined it by making it sound too clean.


----------



## TheSonicTruth (Nov 17, 2017)

bigshot said:


> They didn't remaster Beggar's Banquet. They remixed it and took all the life out of the deliberate compression and distortion choices on the original album. They ruined it by making it sound too clean.




In plain terms, they FU___D IT UP.  Whether by 'remastering' or remixing, that's all they did.  And I couldn't imagine any of the Stones signing off on it.  I'm liking the label or some bone-head producer for that disaster.

That said, people of your stripe always try to justify the post 2000 sausage-fest by pointing out the 'sins of the past' in mixing and mastering terms.  That don't cut the muster with people of my stripe.


----------



## amirm (Nov 17, 2017)

gregorio said:


> 1. It doesn't and that's my point! There is no precise definition of dynamic range and therefore there cannot be a precise measurement of it.


Then why on earth did you say you support the statement Bigshot made that there is no more than 50 db of signal to noise ratio in music?  You should have said the above to him as a minimum.

Instead you went on to say in your experience that is true.  What experience I asked.  DId you ever do the simple test of converting your music to 50 db of effective dynamic range like I did?  Have you done so now that I showed how audible the noise level is in that regard?

You hadn't done any of that, right?  If a subjectivists says they can hear the sound of coasters under their equipment, we immediately demand controlled listening tests and scientific references to back the same.  Yet when it comes to our claims, we want a free pass to take things at face value.  "Oh you don't even need more than 10 bits."   When I ask for data to back that, the answer is "that is my experience."  Heck that is what a subjectivist would say about his experience! 

And no, the math you did in the OP doesn't account for that seeing how I showed direct, peer reviewed research that completely opposes it.  You can't substitute your assumptions about audio science as the real deal.  You need to confirm it.

So once again, neither Bigshot or you are remotely correct that there is only 50 db, 10-bit of information, etc. in our music.  It is a repeated myth by forum objectivists that has no data whatsoever to back it.  It is false and misleading and we shouldn't keep repeating it.

Determining the effect bit depth of music in the face of noise requires statistical analysis.  As it turns out, such work was just done and reported in this write-up against MQA: https://www.xivero.com/downloads/MQA-Technical_Analysis-Hypotheses-Paper.pdf

In there you see them analyzing 24-bit audio and finding that in some cases 23 bits are valid music content! 

"Most uncooled physical systems are able to reach a thermal limited noise floor at around -120dB below full scale. Nevertheless, a theoretical 24Bit system is able to push the quantization noise down to -144dBFS (without dithering) and therefore allows for 24dB = 4Bits of Headroom to place the information of the upper frequency bands. BUT! *We did statistical investigations into existing high resolution recordings and we can confirm that sometimes even the 2nd LSB [Least Significant Bit] already holds information*, most likely due to applied dithering processes..."​
That translates in 138 dB of signal to noise ratio in the recording!  We could de-rate that by a good bit and still be nowhere near 16 bits being the limit let alone 8 bits that was advocated.

So I ask again, what have you done to confirm your assumptions that 50 db is enough?  What listening tests?  What references you can quote?  Because if there is none, then the thesis of this thread is disproven and we are done.


----------



## reginalb

RRod said:


> Heh, "Limited edition". This industry, man.



Now, are you referring to the Limited edition 2CD set, or the limited edition SACD box set, or the limited edition vinyl box set? You have to be specific because there are a lot of limited editions of that album. You might also mean the imported version on Amazon. Not sure where it's imported from, but it's imported. Then there is the most rare of all. The regular version.


----------



## RRod

@amirm I'm missing the part where human beings have to be aware of the informational content. That statistical analysis can show a non-uniform noise spectrum is unsurprising.


----------



## amirm

gregorio said:


> 1. I'm not sure if my system will output 120dBSPL at my listening position, I've never tried and would never want to!


Again, how did you determine that your system isn't capable of that and that you have never listened to that level?  No, you can't use your dumb SPL meter for any of that analysis.  What is important here is instantaneous levels not slow average.

At LA audio show, someone asked Andrew Jones what the SPL levels were that his ELAC speakers were producing in that setting.  He first asked the listeners to give numbers.  People were like you, throwing small numbers like 80 and 90 db around.  His answer was that the peaks were hitting in the neighborhood of 115 db!  The music was dynamic and maybe "loud" by some standards but not at all what you are assuming.

In these discussions people take these SPL numbers as if we are sitting there listening to continuous tone at 120 SPL.  We are not remotely doing that.  We are talking momentary peaks that may last *just a few milliseconds*.  

And no, you don't remotely damage your hearing because two things are needed for that: loudness and exposure time.  Here is the recommendation from US workplace safety standard, OSHA:  https://www.osha.gov/pls/oshaweb/owadisp.show_document?p_table=STANDARDS&p_id=9736






On the left is the SPL number, on the right are the duration in *hours*.  Converting 0.125 for 120 db in minutes we get 7.5.  In other words, you need to listen for 7.5 minutes to the same constant noise, not a few milliseconds as we have in music, to hurt your ears.

So please don't keep talking like these are unheard of numbers.  Can't be done.  We will go deaf, etc., etc.  These are forum objectivists talking points we need to leave behind.

Yes, we are talking about reference level playback.  If you listen to very modest level, your needs will be different.  But again, in the context of what is audible we need to include the full population and their usage of technology.


----------



## amirm

RRod said:


> @amirm I'm missing the part where human beings have to be aware of the informational content. That statistical analysis can show a non-uniform noise spectrum is unsurprising.


We are discussing what is in the content, not what is audible to humans there.  The PR tactic is that if we can show music can never have more than 16 bits, then why talk about > 16 bits?  This data disproves that assertion.


----------



## amirm

gregorio said:


> 2. Really, all your hi-res content cost the same as the 16/44 versions? I don't think that matches the experience of most here.


No, that was in regards to **hardware** that plays it.  We have already paid for the capability to play high-res content in the DACs and systems we own.  So you are not saving anyone money by advocating that they should not need > CD capabilities.  They have the hardware and if they choose to play high-res content, that is a decision they can make on an instant by instant basis.

As I mentioned earlier, my hope is that one day the recording industry gives us access to music prior to final mastering for CD.  That content is already in high-resolution format and should it not be subject to evils of loudness compression, would be very well worth the extra premium charged for it.


----------



## amirm

gregorio said:


> No, it's not! Typically today and for a number of years it's created at 64bit float and before that it was typically created in 32bit float or 56bit fixed. I don't know of a time music was created in 24 bits but it must have been more than about 20 years ago.


No, that is the internal processing format for your DAW.  IThe music was captured at 24-bits and the only output format that is playable by all of us is integer PCM at up to 24 bits or "32 bits float."   That is the format that is handed to the mastering engineer who then processes it with EQ, limiting, etc. and then proceeds to also convert the sample rate down to 44.1 Khz and bit depth to 16 bits.  I want this last step completely eliminated for the music I consume.  I have no need for that "mastering" to formats I don't care about (CD, compressed AAC, MP3, etc.).


----------



## amirm

gregorio said:


> I notice you ignored every single one of the 4 questions I asked, which seems to be a trend in your responses!


I hope you are satisfied now.  I don't feel the obligation to always answer your multi-part questions.  My interest in engaging only go so far and I don't want to bore the membership with such detailed back and forths.  So please don't read much into me not answering everything you say.


----------



## TheSonicTruth

amirm said:


> No, that is the internal processing format for your DAW.  IThe music was captured at 24-bits and the only output format that is playable by all of us is integer PCM at up to 24 bits or "32 bits float."   That is the format that is handed to the mastering engineer who then processes it with EQ, limiting, etc. and then proceeds to also convert the sample rate down to 44.1 Khz and bit depth to 16 bits.  I want this last step completely eliminated for the music I consume.  I have no need for that "mastering" to formats I don't care about (CD, compressed AAC, MP3, etc.).



Why don't you care for CD?


----------



## RRod

amirm said:


> We are discussing what is in the content, not what is audible to humans there.  The PR tactic is that if we can show music can never have more than 16 bits, then why talk about > 16 bits?  This data disproves that assertion.



You've got that backward. The PR tactic is to show information below 16 bits and then to say you need more than 16 bits in your normal living room.


----------



## Cutestudio

gregorio said:


> The problem is that it's impossible to quantify what is over-compression, a perfectly acceptable amount of compression for one song might be ridiculous over-compression in another.



Impossible? In reality it's one of the most trivial things to quantify. A simple histogram shows the distributions and reliably picks out the over compressed/clipped tracks, additionally an entire track with the same consistent level peaks regardless of RMS level or frequency (the classic 'brick' shape) is a classic tell. Clearly you have never studied the subject.

In addition to that one can easily see the flat tops, no 'perfectly acceptable' amount of compression squashes down a 6dB peak into a small range of around -40dB or less. If you'd actually bothered looking at some waveforms you'd not be making these daft statements. Indeed your statements lead me to conclude that you may have heard the phrase 'loudness war' but have no real idea or grasp of the problem; just an observation, because like many you simply have decided your opinions trump any research. I have nothing against ignorance, but it's not working for you here.



gregorio said:


> And incidentally, Mastered for iTunes does NOT stipulate appropriate amounts of compression or directly affects/combats the loudness war.



No specific stipulation no, but it's known for not suffering from clipping/over-compression. Because I've actually researched the Mastered by iTunes masters I knew this which is why I mentioned it as a good example of Apple 'doing something' about the shoddy mastering product record companies pass off, your casual implication that they are poorly mangled is false, sorry.

The number of egos on this forum supporting the total lack of QA by the record companies is sad to see in a HiFi forum and is perhaps the reason why the industry is in such decline. Bigshot even claimed that record companies aim for earbuds and car systems as an excuse for their dreadful output, despite that fact that Pink Floyd's 'The Wall' CD tracks play perfectly both in the car and on any MP3 player I've tried, whereas the modern loud continuous drone pumped out today sounds awful on any playback system.

Other industries have reduced product quality until people stopped buying it, I don't see the record industry bucking the trend. Justifying the destruction of a product may be justifiable to some but for the rest of us it's baffling as it's fraudulent for the buyer, hurts sales and re-sales and would be as easy to avoid as not pressing a few buttons. No one who likes music of sound can support this wilful vandalism.


----------



## Strangelove424

amirm said:


> Again, how did you determine that your system isn't capable of that and that you have never listened to that level?  No, you can't use your dumb SPL meter for any of that analysis.  What is important here is instantaneous levels not slow average.
> 
> At LA audio show, someone asked Andrew Jones what the SPL levels were that his ELAC speakers were producing in that setting.  He first asked the listeners to give numbers.  People were like you, throwing small numbers like 80 and 90 db around.  His answer was that the peaks were hitting in the neighborhood of 115 db!  The music was dynamic and maybe "loud" by some standards but not at all what you are assuming.
> 
> ...



Andrew Jones makes great speakers, I own some that he designed. You never responded, but remember I asked before about the acoustic source and micing? The reason for this is that any kind of amplified venue performance, whether a rock concert or an Andrew Jones demo is an artificial example of dynamic range. The peak level is simply what they decide to crank it to. For very large audiences, the amplification required is immense, and nowhere close to the SPL created by the original instrument, assuming it was acoustic. And again, if it was an electronic guitar or something that needed amplification, the measure would be just as subjective.

Regarding OSHA standards, if you feel it is necessary to push your hearing health to the absolute limits prescribed for industrial conditions for workers who are machining and welding, be my guest. If you still end up with hearing damage, despite limiting yourself to .11 of 120db hourly doses, well... you were warned. Most people find a comfortable music volume, peaks included, to be well under that. And you still haven't provided proof of acoustic sound sources from normal listening positions to show that these people aren't listening loud enough.


----------



## TheSonicTruth

Cutestudio said:


> Impossible? In reality it's one of the most trivial things to quantify. A simple histogram shows the distributions and reliably picks out the over compressed/clipped tracks, additionally an entire track with the same consistent level peaks regardless of RMS level or frequency (the classic 'brick' shape) is a classic tell. Clearly you have never studied the subject.
> 
> In addition to that one can easily see the flat tops, no 'perfectly acceptable' amount of compression squashes down a 6dB peak into a small range of around -40dB or less. If you'd actually bothered looking at some waveforms you'd not be making these daft statements. Indeed your statements lead me to conclude that you may have heard the phrase 'loudness war' but have no real idea or grasp of the problem; just an observation, because like many you simply have decided your opinions trump any research. I have nothing against ignorance, but it's not working for you here.
> 
> ...



"_The number of egos on this forum supporting the 
total lack of QA by the record companies is sad to see 
in a HiFi forum and is perhaps the reason why the 
industry is in such decline. Bigshot even claimed that 
record companies aim for earbuds and car systems as 
an excuse for their dreadful output, despite that fact that 
Pink Floyd's 'The Wall' CD tracks play perfectly both in 
the car and on any MP3 player I've tried, whereas the 
modern loud continuous drone pumped out today sounds 
awful on any playback system._"


Could not have put it better myself, Cutestude!   

The snake oil excuses that bigshot and other industry insiders tout about no longer hold water, because technology has leveled the playing field between the record makers and the music buyers. DAW software is readily available, at cost, as shareware, and for free or donation, for installation and use on desktop or mobile platforms.  We all can hear and see the damage inflicted on our musical legacy, and sold to us as "new and improved", and "remastered at 24bit for your enjoyment."

Well bigshot, gregorio, et al, to paraphrase Peter Finch yelling out that window in the film "Network":  

'_WE'RE MAD AS HE L L, AND WE'RE NOT GOING TO BUY IT ANY MORE!!_'


----------



## castleofargh

Strangelove424 said:


> Andrew Jones makes great speakers, I own some that he designed. You never responded, but remember I asked before about the acoustic source and micing? The reason for this is that any kind of amplified venue performance, whether a rock concert or an Andrew Jones demo is an artificial example of dynamic range. The peak level is simply what they decide to crank it to. For very large audiences, the amplification required is immense, and nowhere close to the SPL created by the original instrument, assuming it was acoustic. And again, if it was an electronic guitar or something that needed amplification, the measure would be just as subjective.
> 
> Regarding OSHA standards, if you feel it is necessary to push your hearing health to the absolute limits prescribed for industrial conditions for workers who are machining and welding, be my guest. If you still end up with hearing damage, despite limiting yourself to .11 of 120db hourly doses, well... you were warned. Most people find a comfortable music volume, peaks included, to be well under that. And you still haven't provided proof of acoustic sound sources from normal listening positions to show that these people aren't listening loud enough.



120dB isn't an issue, most people don't have gear to output those levels anyway, and the few who can only have a handful who can do it without massive noise/distortions. my active speakers are rated at 112 dB SPL C-Weighted, and I use replaygain  . guess I'm not ready to be an audiophile just yet(the last time my speakers played at 90dB, it was to measure them).
also using 120dB output while arguing that the LSB of a file are significant, brings the question about noise and distortions again. or stuff like messed up frequency response and how dramatic it must be in proportion.
it's the issue with specific and extreme situations used to argue a point, we're bound to come face to face with a situation that contradicts it.
we need 120dB because music could contain that much. ok, why not. then we need to listen super loud, else the low amplitude content would get lost below the least noticeable sound we determined while listening to super quiet sounds because that's how contradictory we are. then we need the music signal to be overall super quiet because the instantaneous dynamic range of the ear is much closer to 60dB than it is to 120. I remember reading something suggesting that the best listening level for a test was around 60 to 70dB(depending on when ou tympanic response is triggered). so we're back arguing about super quiet music content encoded super low for no reason while playing it with a lot of gain somehow. yeah it stopped making any sense a while back.
but there is better, if I'm playing stuff at 120dB on most of my gears, I will get well over 1% THD from the transducer. which is crap added above 120-40=80dB. so we're back to questions about masking and fidelity levels. how fun it is to debates the need for 20bit resolution when the output is starts being messed up in the -40 to -60dB area for almost all playback systems. only possible when looking at everything in isolation, else it's absurd, but look at all the fun we can have worrying about irrelevant stuff. ^_^



@Cutestudio and @TheSonicTruth this is more relevant and has helped fighting against the loudness https://tech.ebu.ch/docs/r/r128.pdf
if you want better mastering and no more clipping, go fight for a more global embrace of such standards, as they are a much better deterrent to brick walled compression than bitching on an consumer forum and putting the blame on people who never said they were favorable to the extremities of what was called the loudness war. you're picking a fight with the wrong people here. and when you assume they say the opposite of what you say simply because they come to point out some obvious flaws in your correlation to causation arguments. you're the ones in the wrong.
having a noble goal doesn't justify using nonsense arguments to push it. the day the loudness war comes to an end, we'll all open a cool drink and cheer to the arrival of a better world.

[troll mod ON] @bigshot worked on the Chipmunk albums, so he know all there is to know about high fidelity and high dynamic recordings. checkmate! [/troll mod OFF]


----------



## amirm

Strangelove424 said:


> Andrew Jones makes great speakers, I own some that he designed. You never responded, but remember I asked before about the acoustic source and micing? The reason for this is that any kind of amplified venue performance, whether a rock concert or an Andrew Jones demo is an artificial example of dynamic range. The peak level is simply what they decide to crank it to. For very large audiences, the amplification required is immense, and nowhere close to the SPL created by the original instrument, assuming it was acoustic. And again, if it was an electronic guitar or something that needed amplification, the measure would be just as subjective.


Content doesn't have to be acoustic to be considered content.    For reasons you mention, we can in post production create much louder and much softer signals.  So when considering what the acoustic world looks like, we are being kind to the topic at hand.  Ultimately, "content can be whatever it wants!"

And in the context of playback, I can have very small rooms with high efficiency which readily allows high SPL playback.

As to your question, I must have missed your question.  In what context are you asking about micing and acoustic source?


----------



## amirm

castleofargh said:


> but there is better, if I'm playing stuff at 120dB on most of my gears, I will get well over 1% THD from the transducer. which is crap added above 120-40=80dB. so we're back to questions about masking and fidelity levels. how fun it is to debates the need for 20bit resolution when the output is starts being messed up in the -40 to -60dB area for almost all playback systems. only possible when looking at everything in isolation, else it's absurd, but look at all the fun we can have worrying about irrelevant stuff. ^_^


Again, this is the same issue as using single value SPL numbers for room noise.  THD is not a measure of audibility.  It is a single number and psychoacoustically blind.  

The bulk of the energy in music is in low frequencies.  Harmonic distortion generated there doesn't necessarily rise to 2 to 4 Khz where your ears are most sensitive.  To know that it does, you need to evaluate its full spectrum, and then compare that to threshold of hearing.  Without this kind of psychoacoustic analysis, the conclusion are bound to always be wrong.  You just can't go there if you want audio science to back you.

 Importantly, no one cares if we can hear low level noise during that peak.  That peak can be all distorted for all it wants.  We care that once the peak has passed, do we hear the channel noise during quiet portions of music.

Also, one type of distortion doesn't mask another.  I can readily hear my tape deck background noise at 80 db signal to noise ratio in my system.  It has nowhere to hide no matter how much distortion my speakers have.

Finally I keep having to say this: *isn't this a headphone focused forum???*  Here you can get incredible SPLs without even trying.  You can also assure absolutely low noise levels.  So much of the arguments here are moot.


----------



## Strangelove424

amirm said:


> Content doesn't have to be acoustic to be considered content.    For reasons you mention, we can in post production create much louder and much softer signals.  So when considering what the acoustic world looks like, we are being kind to the topic at hand.  Ultimately, "content can be whatever it wants!"
> 
> And in the context of playback, I can have very small rooms with high efficiency which readily allows high SPL playback.
> 
> As to your question, I must have missed your question.  In what context are you asking about micing and acoustic source?



In that case, dynamic range is whatever you decide it to be. 140db of instant deafness? Sure! Just crank it up higher! If you are willing to be such a subjectivist about how to establish max SPL in a music listening scenario, what are you arguing with everyone about then? Obviously there has to be an objective standard, an original upon which all the simulations must be based. And the accuracy of the reproduced simulations is reliant on standard you set for the original. Without an original standard (acoustic source) you are arguing on a slippery slope, and deciding dynamic range based upon a whim of how to set your volume. 

From post #4324, page 289:

"How is the "loudest real life music" instrument measured? I'm assuming acoustic sources only, then which instrument and how was it mic'd up? Did they stick a mic directly into the drums, or rest it on the skin? These numbers are insanely high for a music listening scenario."


----------



## pila405 (Nov 17, 2017)

Have a crack at this to check your hearing - how far can you go and hear the attenuated sample while still being comfortable with the original:
http://www.audiocheck.net/blindtests_dynamic.php

I cave at around 60dB.

And I can pass this test without a problem:
http://www.audiocheck.net/blindtests_16vs8bit.php


----------



## Darren G (Nov 17, 2017)

bigshot said:


> Frampton Comes Alive?



It was quite the treat when I heard this concert live in the 70's, but hey, I am old 

My music collection is utterly incomprehensible.  So are my tastes in who I'll go see live.  Tech N9ne and E-40 live... talk about energy, bass that you could feel, slam.  Why I honestly don't care about shaving off another 10db of noise at home.  The money is better spent on saving some $'s for another live music experience.


----------



## bfreedma (Nov 17, 2017)

amirm said:


> I hope you are satisfied now.  I don't feel the obligation to always answer your multi-part questions.  My interest in engaging only go so far and* I don't want to bore the membership with such detailed back and forths*.  So please don't read much into me not answering everything you say.




Thanks for the laugh, I needed that on a Friday afternoon.  Amir, I've seen your posting style on multiple forums (AVS, Hydrogenaudio, ....) - for you to post that you aren't interested in detailed back and forth (mostly focused around word parsing and using unlikely extremes as "examples") truly made me LOL.  If you were really interested in the discussion, you would respond to questions asked, even if they force you to consider that you may not be entirely correct.

You can have the last word - I'll have you on ignore as usual going forward.


----------



## amirm

Strangelove424 said:


> In that case, dynamic range is whatever you decide it to be. 140db of instant deafness? Sure! Just crank it up higher! If you are willing to be such a subjectivist about how to establish max SPL in a music listening scenario, what are you arguing with everyone about then? Obviously there has to be an objective standard, an original upon which all the simulations must be based. And the accuracy of the reproduced simulations is reliant on standard you set for the original. Without an original standard (acoustic source) you are arguing on a slippery slope, and deciding dynamic range based upon a whim of how to set your volume.
> 
> From post #4324, page 289:
> 
> "How is the "loudest real life music" instrument measured? I'm assuming acoustic sources only, then which instrument and how was it mic'd up? Did they stick a mic directly into the drums, or rest it on the skin? These numbers are insanely high for a music listening scenario."


No, the dynamic range is not whatever I decide to be.  It is actual measured peak loudness of real instruments playing vs threshold of noise detection.  All the details are in the JAES paper I referenced. Here is a bit of it: 
Dynamic-Range Issues in the Modern Digital Audio Environment*
LOUIS D. FIELDER, *AES Fellow*
Dolby Laboratories Inc., San Francisco, CA 91403, USA






Here is figure 2:





More detail is provided in the reference in that paper, "PRE-& POST-EMPHASIS TECHNIQUES AS APPLIED TO AUDIO RECORDING SYSTEMS By louis D. Fielder"

Here is a bit from that:









As you see, none of this is simulation or arbitrary data.  There is fair bit more in the papers so I recommend reading them if you want more information.  The author is one of the luminaries in this space and is ex-president of AES, an AES fellow, etc.  So please don't be so flippant about it.


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## RRod (Nov 17, 2017)

What were the noise levels in the venues?

Edit: to put it another way, you are basically saying that rock has more dynamic range than classical, which isn't the actual experience of listening to them…


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## Strangelove424 (Nov 17, 2017)

It appears that the people who did this study differentiated between acoustic and “electronically augmented” sources just as I am. The problem is that “augmented” is a distortion of the language and the truth. A rock concert, electronic guitar, electronic violin, or keyboard is not merely “augmented” by electronic amplification, it’s entirely composed of it! To include measurements of ‘electronically augmented’ performances is the snake eating its own tail, for the reasons I mentioned above. And for those reasons, you’ll notice how in figure 2, the average SPL curve for electronically augmented material is shifted about 10db higher. Since one rock concert (Metallica) might be arbitrarily louder than another (Paul Simon) I’m going to focus on acoustic sources like classical music, where SPL has no electronic amplification and comes from acoustic resonance only. Instruments I can’t simply pour more watts into it till everyone is deaf, gorged out on SPL.

I am not sure what the author means by “favored listening position”. That is too vague to be of any use. I have been to many symphonies, often getting to sit in what I personally consider “favored listening positions” within the lower third of the hall. I am also very sensitive to loud noise. 120db is enough to cause me extreme discomfort, ear ringing, and muffled sound for a duration afterwards. At no time at any symphony was I uncomfortable with the loudness. And after the symphony, I am always able to enjoy a drink with friends and conduct quiet conversation with ease. A max of 127db for a "non-augmented" acoustic source sounds crazy high to me, and I've attended concerts and worked with recording equipment for many years. Your numbers sound like they're about 20db too high. And I am not alone, you can check this link to the music program at UCSD: http://musicweb.ucsd.edu/~trsmyth/level175/Example_SPL_Levels.html They peg classical music to peak around 105db. That sounds a whole lot closer to my experience in reality.


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## bigshot (Nov 18, 2017)

TheSonicTruth said:


> That said, people of your stripe always try to justify the post 2000 sausage-fest by pointing out the 'sins of the past' in mixing and mastering terms.  That don't cut the muster with people of my stripe.



You and your ilk!

The truth is that when it comes to intentional distortion and compression, it's a creative issue. I'm going to judge a remaster at how well it expresses the spirit of the music. I'm not going to assume the remaster is better or that the original is better, I'm going to listen and decide for myself. In the case of the Stones, the original is better. In the case of David Bowie, the remasters are better. There's no hard and fast rule to this stuff. You have to open your mind and ears and listen to the music.



amirm said:


> Then why on earth did you say you support the statement Bigshot made that there is no more than 50 db of signal to noise ratio in music?



You're a real piece of work. You know darn well that isn't what I said. I said that in commercial recordings, a 50dB dynamic range is about the most dynamics you're going to find. Dynamic range in music is NOT the same as signal to noise and you know that. You're just prevaricating again.



amirm said:


> I hope you are satisfied now.  I don't feel the obligation to always answer your multi-part questions.



No, you refuse to answer the questions that would make you admit you're fudging. You are a remarkably disingenuous person.



RRod said:


> What were the noise levels in the venues?



If the classical music was recorded in Carnegie Hall, the sound of the subway train passing underground is clearly audible on live recordings!


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## bigshot (Nov 18, 2017)

TheSonicTruth said:


> The snake oil excuses that bigshot and other industry insiders tout about no longer hold water.



If you get to call my comments snake oil, I get to point out that you are a duffer- an armchair quarterback who has never made anything but feel like you're qualified to tell other people how to make it.



castleofargh said:


> [troll mod ON] @bigshot worked on the Chipmunk albums, so he know all there is to know about high fidelity and high dynamic recordings. checkmate! [/troll mod OFF]



Man! You listen to the album I produced! It was the most complex recording/mix I ever worked on. I had to make double speed vocals sound like Michael Jackson, Jimi Hendrix, Bob Dylan, Elton John and Bruce Springsteen- all within the space of a single three minute song. I have the CD here somewhere. It was a royal bitch to make! I was very lucky to have first class musicians, singers and engineers to work with on that. The Chipmunks may not be Mozart, but there are a lot of work to get right. I learned a hell of a lot on that. And of course it'll please all the luddites that it was a glorious AAD analogue recording made on 24 track tape.


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## amirm

bigshot said:


> You're a real piece of work. You know darn well that isn't what I said. I said that in commercial recordings, a 50dB dynamic range is about the most dynamics you're going to find. Dynamic range in music is NOT the same as signal to noise and you know that. You're just prevaricating again.


Once more, whatever the 50 db number is, how did you measure it?  How much music did you test?  And do you have any references to back that in literature?


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## bigshot

It's related to the other spec I posted about the dynamic range of human ears. We can hear about 45dB of dynamic range *at a time without having to take time for our ears to adjust to the different peak level.* Look it up.


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## TheSonicTruth

bigshot said:


> It's related to the other spec I posted about the dynamic range of human ears. We can hear about 45dB of dynamic range *at a time without having to take time for our ears to adjust to the different peak level.* Look it up.




Does that justify squashing the recorded product to compensate?

Does a symphony orchestra or a good acoustic jazz band care about "45db at a time"?

If ya can't stand the heat, get back to the cheap seats!  I'm not a front row attendee myself.


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## gregorio (Nov 18, 2017)

amirm said:


> Again, how did you determine that your system isn't capable of that and that you have never listened to that level?  No, you can't use your dumb SPL meter for any of that analysis.  What is important here is instantaneous levels not slow average. At LA audio show, someone asked Andrew Jones what the SPL levels were that his ELAC speakers were producing in that setting.  He first asked the listeners to give numbers.  People were like you, throwing small numbers like 80 and 90 db around.  His answer was that the peaks were hitting in the neighborhood of 115 db!
> [2] The music was dynamic and maybe "loud" by some standards but not at all what you are assuming. In these discussions people take these SPL numbers as if we are sitting there listening to continuous tone at 120 SPL.  We are not remotely doing that.  We are talking momentary peaks that may last *just a few milliseconds*. ... [2a] Converting 0.125 for 120 db in minutes we get 7.5. In other words, you need to listen for 7.5 minutes to the same constant noise, not a few milliseconds as we have in music, to hurt your ears. ...
> [3] So please don't keep talking like these are unheard of numbers.  Can't be done.  We will go deaf, etc., etc.  These are forum objectivists talking points we need to leave behind.



1. In the commercial dub stages I work at the systems are calibrated to -20dBFS = 85dBSPL(C), max instantaneous peaks at near 0dBFS would therefore be around 115dB. However, in a smaller room, say a good sized living room, that figure needs to be reduced by at least 7dB. And, this is for feature films, for music that figure needs to be reduced by about another 8dB because of the compression and the fact that the "momentary peaks" are very significantly less than the roughly 35dB above normal levels, as they are with films. With the vast majority of commercial music releases momentary peaks of 120dB will result in a significant amount/duration of the music being around or even above 100dB.

2. *Furthermore*, as has been explained to you and as you consistently deliberately ignore, the human ear has a dynamic range of around 60dB (or less) NOT the 120dB you keep quoting. To attain the 120dB figure requires a threshold shift of that 60dB window. In the case of say orchestral musicians, they have a fairly high noise floor to start with, due to the fact there are 90 or so people in the room, all breathing, moving, turning pages and operating mechanical instruments. If the noise floor of the venue plus all these musicians were say 40dB, then the threshold shift required of the 60dB window would be in the region of 20dB. It must also be noted that even with the real life noise floors of symphony orchestras, studies (for example "_Hearing protection and hearing symptoms in Danish symphony orchestras_", 2006) have shown significant hearing damage (permanent threshold shift) for orchestral musicians and incidentally, film re-recording mixers also commonly have work related hearing problems. In fact, quite a few orchestral musicians and some film re-recording mixers routinely wear hearing protection while working! However, you are talking about detectable noise floors around 0dB and therefore a much larger threshold shift! If you want to listen to all your music with a 0dB noise floor *AND* the same high SPLs as the musicians themselves experience, it's your ears and your look out. Additionally of course, all this is just for orchestral and purely acoustic music genres, which generally has a significantly lower RMS than the other, far more popular music genres!
2a. Unsurprisingly, you avoided my point and question about 32 bit releases. What does your disputed OSHA chart say is the acceptable duration for (24bit) 144dBSPL peaks and what about (32bit) 192dB peaks? Again, what is the problem with the 120dB range of 16bit????

3. I object to people cherry picking the most favorable studies, evidence and rarest conditions, while ignoring other evidence and real life conditions, just to support an agenda or win an argument. I object even more strongly when that cherry picked evidence is presented as "real life" *BUT*, when that cherry picked evidence is so far beyond real life that it's potentially dangerous/damaging, I object in the STRONGEST TERMS POSSIBLE!!! This quoted statement of yours is breath-takingly hypocritical amirm and highly irresponsible!



amirm said:


> DId you ever do the simple test of converting your music to 50 db of effective dynamic range like I did?



As I'm sure you well know amirm your simple test does NOT convert to 50dB of effective dynamic range! 8 bit provides for 48dB, the last of those 8 bits is dither noise and we are not defining the dynamic range as loudest peak to digital noise floor, as has ALREADY been explained to you! In many cases it's trivially easy to hear dither noise at -42dB.

I've answered your questions, why do you continue to deflect and refuse to answer mine?

G


----------



## gregorio

Cutestudio said:


> [1] Impossible? In reality it's one of the most trivial things to quantify. A simple histogram shows the distributions and reliably picks out the over compressed/clipped tracks, additionally an entire track with the same consistent level peaks regardless of RMS level or frequency (the classic 'brick' shape) is a classic tell.
> [2] ... no 'perfectly acceptable' amount of compression squashes down a 6dB peak into a small range of around -40dB or less.
> [3] If you'd actually bothered looking at some waveforms you'd not be making these daft statements. Indeed your statements lead me to conclude that you may have heard the phrase 'loudness war' but have no real idea or grasp of the problem; just an observation, because like many you simply have decided your opinions trump any research.
> [3a] I have nothing against ignorance ....



1. Obviously you don't know what the word "quantify" means. Yes, seeing a brick shaped waveform would tend to indicate severe over-compression (although that is not always the case) but that's all it tells us, it does NOT tell us how much compression has been applied and the word "quantify" means; to determine how much!
2. Absolutely it can and on occasion applying large amounts of compression is not only "perfectly acceptable" but actually highly desirable! I'm talking in general though, as your figures don't make sense.
3. That's funny! I look at waveforms for roughly 8 hours almost every single working day and have done for about 20 years. And, as one of those music/sound engineers often required to create mixes/masters which perpetuate the loudness wars then obviously it's NOT just an observation, it's 25 years practical experience of actually doing it! The reason my statements may seem daft to you and why they have led you to a conclusion which is so completely opposite to the truth is because ...
3a. Not only do you have nothing against ignorance but you apparently seem to be very strongly in favour of it!!!

Again, accusing others of what you yourself are guilty of, is going to accomplish nothing other than to make yourself look foolish. Obviously, you have a moderately poor/limited understanding of the mixing/mastering process, of the approval process and of when, how and why compression is used. Either you ask some questions and gain a better understanding or continue to defend your misunderstanding/s and make yourself look ever more foolish, your choice.

G


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## Cutestudio

gregorio said:


> 1. Obviously you don't know what the word "quantify" means.


You are free of course to redefine any word the way you like, we don't need many decimal places to detect the mastering carnage and mangled remasters.
Peak/RMS expressed as dB doesn't have any mystical sense for you to fathom, it does however show the amount of damage between original tracks and the remasters to a reasonable accuracy, backed up by histograms the damage is then plain to see and quantifiable in various ways as you are obviously aware.



gregorio said:


> 1. Obviously you don't know what the word "quantify" means.
> 2. Absolutely it can and on occasion applying large amounts of compression is not only "perfectly acceptable" but actually highly desirable! I'm talking in general though, as your figures don't make sense.


Your definition of 'perfectly acceptable' and 'highly desirable' differs from people forced to listen to their purchases. Shoddy products damage sales, the record industry is not immune.



gregorio said:


> 3. That's funny! I look at waveforms for roughly 8 hours almost every single working day and have done for about 20 years.


Clearly then you are either not looking at the final product of mastering for CD or other distribution, or if you are you have somehow convinced yourself that clipped and over compressed brick shaped waveforms are 'perfectly acceptable' and 'highly desirable'. Again, good for you, but don't expect anyone else to like them, especially on a HiFi forum.



gregorio said:


> Again, accusing others of what you yourself are guilty of


??. Cheer up Greg, what you are experiencing is 'people disagreeing with you', obviously it's a novelty for you but not everything has to be about your ego.


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## bigshot (Nov 18, 2017)

TheSonicTruth said:


> Does that justify squashing the recorded product to compensate?
> Does a symphony orchestra or a good acoustic jazz band care about "45db at a time"?



Yes. Making a great sounding recording usually involves using compression. Especially with vocals.
Yes, the conductor is up there adjusting balances to keep the range within a comfortable level.

More isn't better. Just right is best. A dynamic range greater than 50dB in a single song is uncomfortable to listen to. You'd be running to grab the volume control to turn it up in the quiet parts and turn it down in the loud parts- essentially manually compressing it.

There's no reason to clip a digital recording, but compression is a useful tool. It can be abused for sure, but it is essential nonetheless.


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## gregorio (Nov 18, 2017)

Cutestudio said:


> [1] You are free of course to redefine any word the way you like, we don't need many decimal places to detect the mastering carnage and mangled remasters.
> [2] Peak/RMS expressed as dB doesn't have any mystical sense for you to fathom, it does however show the amount of damage between original tracks and the remasters to a reasonable accuracy, backed up by histograms the damage is then plain to see and quantifiable in various ways as you are obviously aware.
> [3] Your definition of 'perfectly acceptable' and 'highly desirable' differs from people forced to listen to their purchases.
> [4] Clearly then you are either not looking at the final product of mastering for CD or other distribution, or if you are [4a] you have somehow convinced yourself that clipped and over compressed brick shaped waveforms are 'perfectly acceptable' and 'highly desirable'.



1. Oh dear, I see you've gone for the second option, of just keep accusing others of what you're guilty of and making a fool of yourself.
2. I take it you mean an earlier master and a re-master? You don't, I presume, have access to the original tracks and therefore you have no idea how much compression was applied to them during mixing. Likewise, you have no idea how much further compression was added to that compressed mix in the earlier master and no idea of the amount of compression added on the re-master. All you've got is a comparison of the crest factor of one already compressed master with the crest factor of another compressed master (remaster) but you have absolutely no idea how much compression either contains. And, this crest factor comparison of two already compressed masters does not necessarily purely relate to compression, as EQ and simple level changes can affect the crest factor. As you would know, if you were not so ignorant of the mixing and mastering processes!
3. Who is forcing you to purchase and listen? Why don't you report them?
4. Ah, you've got me. That's where I'm going wrong, I'm not looking at the masters which I make, can't believe I've been forgetting to do that for 25 years!
4a. OK, so you've demonstrated you don't know what "quantify" means and now that you can't read, as I stated very clearly that I've been arguing against the loudness wars for a very long time, almost as long as I've been in the business. I do not condone clipping, except in some very exceptional circumstances and I don't condone over-compression. However, you appear to be defining over-compression as 6dB, in which case just about every single commercial rock/pop track in about the last 50 years is shoddy, not worth buying, damaged (or whatever) according to you! All you're really doing with this particular argument is demonstrating your ignorance of the approval process, because regardless of what I consider to be a "perfectly acceptable" amount of compression, the final decision is down to the clients; the artists and producer or record label. And, with no exceptions I can recall over the past 25 years, the amount of compression they want is either the same or more than I would apply if the decision were mine!

While over-compression/the loudness war does reduce dynamic range and therefore require considerably fewer than 16 bits, this conversation/argument is effectively now off-topic and I didn't start this thread just to provide you with an opportunity to demonstrate your ignorance and make a fool of yourself.

G


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## TheSonicTruth

Cutestudio said:


> You are free of course to redefine any word the way you like, we don't need many decimal places to detect the mastering carnage and mangled remasters.
> Peak/RMS expressed as dB doesn't have any mystical sense for you to fathom, it does however show the amount of damage between original tracks and the remasters to a reasonable accuracy, backed up by histograms the damage is then plain to see and quantifiable in various ways as you are obviously aware.
> 
> 
> ...




Cutestudio:

These guys know they're trying to justify bad practices and LO-Fi product, and can't stand being told by relative outsiders, or at least, insiders who haven't reached the same point in the Peter Principle as they, and accordingly throw, as exemplfied in #4432, a bunch of diatribe about how WE'RE in the wrong, how we're the ones who are crazy!  We're just wasting our time.


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## gregorio

bigshot said:


> Yes, the conductor is up there adjusting balances to keep the range within a comfortable level.



Often, that means telling the brass section to play quieter. One notable exception I clearly recall is Leonard Bernstein stopping the orchestra during a rehearsal and telling me that as my part was marked ffffff (forte x 6), I should play it as loudly as possible. We ran through the section again and I did as requested, much to the amusement of the orchestra (whom I obliterated) and on-lookers ... Mr. Bernstein was forced to retract/temper his request! 

G


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## gregorio

TheSonicTruth said:


> We're just wasting our time.



If you don't want to waste your time, then I suggest you learn how to read. What part of the "_I don't condone over-compression_" and "_I've been arguing against the loudness war for nearly 25 years_" did you not understand?

G


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## TheSonicTruth (Nov 18, 2017)

gregorio said:


> If you don't want to waste your time, then I suggest you learn how to read. What part of the "_I don't condone over-compression_" and "_I've been arguing against the loudness war for nearly 25 years_" did you not understand?
> 
> G



Alright, perhaps you're not as bad as bigshot in that regard.

I don't oppose entirely the use of compression, but I do oppose it being used on top of whatever amount was used in creating the final stereo master used for vinyl, CD, whatever 30 years ago, in the remastering process.  Doing so changes the sound from what the original master sounded like, and therefore is no longer authentic.


----------



## amirm

bigshot said:


> It's related to the other spec I posted about the dynamic range of human ears. We can hear about 45dB of dynamic range *at a time without having to take time for our ears to adjust to the different peak level.* Look it up.


You said *content* has that range, not what we hear:



bigshot said:


> Classical music is the opposite side of the spectrum, and it can have as much as a 50dB range.



So I ask again where you got that information that classical music has a "50 db range?"

Or do you want to take that back as having no basis in anything other than a guess on your part?


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## bigshot (Nov 18, 2017)

gregorio said:


> If you don't want to waste your time, then I suggest you learn how to read. What part of the "_I don't condone over-compression_" and "_I've been arguing against the loudness war for nearly 25 years_" did you not understand?



And I suggest that he struggle to understand that the use of compression in a mix is not the same as the loudness war. Compression is a tool that is used in almost all mixes, even the ones from the golden past that he's championing.

Amirm, our ability to hear is directly related to how sound engineers mix. They mix to what the human ear can hear. The goal is clarity and organization of sound. They don't create a mix according to numbers on a page. Commercial mixes stay in a dynamic range of 45 to 50dB because that is what ears can hear without having to adjust. Do some simple googling and you'll figure it out. I'm not going to the trouble to scribble on a book with yellow highlighter and scan it when you really aren't interested in anything but your own words.

I've explained both of these things many times and it doesn't seem to register. Ignorance is OK. As Mark Twain said, "Everyone is ignorant... just on different subjects." I know stuff. You know stuff. We should find a way to communicate, not throw up roadblocks like you're doing. I'm here to communicate, not to go in circles repeating the same thing over and over again. When people do that, I start skimming over the repetitive parts of their posts and speak past them in my replies for the benefit of the lurkers. I'll start talking past you and the witch hunt duo too soon.


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## amirm

RRod said:


> You've got that backward. The PR tactic is to show information below 16 bits and then to say you need more than 16 bits in your normal living room.


What is a "normal living room?"  People listening with headphones which has no rooms.  And they have dedicated listening rooms.  We cannot impose limits on fidelity of music by assuming people sit in their living room to listen.

And to what end anyway?  The music industry has already started to distribute higher resolution music.  What is the point of campaigning against it?  You want them to stop?  To what end?

What is the logic of this discussion anyway?  It is not like we are 20 years ago and fighting over SACD/DVD-A vs CD.  We are in digital age where there is no barrier to distribution of high-resolution music and its playback.  You guys want to turn the tide back?  Why?


----------



## gregorio

TheSonicTruth said:


> I don't oppose entirely the use of compression, but I do oppose it being used on top of whatever amount was used in creating the final stereo master used for vinyl, CD, whatever 30 years ago, in the remastering process.  Doing so changes the sound from what the original master sounded like, and is therefore no longer authentic.



While not condoning over-compression, what would be the point of paying a mastering engineer to create a remaster which sounds exactly the same as an already existing, previous master? The whole point of a remaster is to make it sound different!

You say you don't oppose entirely the use of compression but I'm saying that's a bizarre statement! Compression is a vital tool, it's used at numerous stages/places in the process of creating a commercial song; at the individual channel level, at the sub group level, on the final mix and then again during mastering. If you object to compression, then you object to all rock and pop music from the early mid-1960's to the present day. The question/problem is not and cannot be about whether compression is used, because it must be, the problem is over-compression and that is extremely hard to define because it varies, from track to track and genre to genre. In other words, exactly the same amount of compression on one song which sounds great, may sound like a completely ridiculous amount of over-compression on another song.

G


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## amirm

Strangelove424 said:


> I am not sure what the author means by “favored listening position”. That is too vague to be of any use. I have been to many symphonies, often getting to sit in what I personally consider “favored listening positions” within the lower third of the hall. I am also very sensitive to loud noise. 120db is enough to cause me extreme discomfort, ear ringing, and muffled sound for a duration afterwards. At no time at any symphony was I uncomfortable with the loudness. And after the symphony, I am always able to enjoy a drink with friends and conduct quiet conversation with ease. A max of 127db for a "non-augmented" acoustic source sounds crazy high to me, and I've attended concerts and worked with recording equipment for many years. Your numbers sound like they're about 20db too high. And I am not alone, you can check this link to the music program at UCSD: http://musicweb.ucsd.edu/~trsmyth/level175/Example_SPL_Levels.html They peg classical music to peak around 105db. That sounds a whole lot closer to my experience in reality.


I think there is still some misunderstanding about the data.  The numbers I have presented from research are *peak SPLs. *All this other data you see and measurements people use are averaged numbers, not instantaneous peaks.  

With respect to what has to exist in a digital channel, we have no choice but to use peak numbers because that is what the information is that must be stored.  Average SPL numbers are used for other uses but not here.

In general, you have to add 5 to 10 db to go from average SPL to peak.  

It is for this reason I say that many of you have been exposed to much higher peak SPLs than you imagine since you impression of loudness comes from averaging SPL values that are always lower.

The same is true of that link you provided, putting aside the fact that it has no details on how any measurement was performed.


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## gregorio (Nov 18, 2017)

amirm said:


> [1] We cannot impose limits on fidelity of music by assuming people sit in their living room to listen.
> [2] And to what end anyway?  The music industry has already started to distribute higher resolution music.  What is the point of campaigning against it?  You want them to stop?  To what end?
> [3] It is for this reason I say that many of you have been exposed to much higher peak SPLs than you imagine since you impression of loudness comes from averaging SPL values that are always lower.



1. Of course we can, that's what we do, that's what mastering is for!
2. The music industry started to distribute higher res over 15 years ago and continues to distribute higher and higher res, we've got DACs out there capable of 768/32 and 368/32 or DSD64 content. It will never end and audiophiles will continue to be scammed on equipment and content ad infinitum because a higher res is always assumed to be better than a lower res. That's why we argue against it. I ask you AGAIN, what do you need that's beyond the 120dB limitation of 16bit?
3. No, we haven't been exposed to much higher SPLs than we imagine. Taking your own figures from the previous sentence, we've been exposed to peak levels 5-10dB higher than our average levels. So, if our average levels are say 80dB, which would be high in a near 0dB noise floor environment, the peak SPLs are at say 90dBSPL, about thirty times lower than 16bit can provide!

G


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## TheSonicTruth (Nov 18, 2017)

gregorio said:


> While not condoning over-compression, what would be the point of paying a mastering engineer to create a remaster which sounds exactly the same as an already existing, previous master? The whole point of a remaster is to make it sound different!
> 
> You say you don't oppose entirely the use of compression but I'm saying that's a bizarre statement! Compression is a vital tool, it's used at numerous stages/places in the process of creating a commercial song; at the individual channel level, at the sub group level, on the final mix and then again during mastering. If you object to compression, then you object to all rock and pop music from the early mid-1960's to the present day. The question/problem is not and cannot be about whether compression is used, because it must be, the problem is over-compression and that is extremely hard to define because it varies, from track to track and genre to genre. In other words, exactly the same amount of compression on one song which sounds great, may sound like a completely ridiculous amount of over-compression on another song.
> 
> G



I'm going to post the following on audio forums, pages, and newsgroups all over the internet:

_While not condoning over-compression, what 
would be the point of paying a mastering engineer 
to create a remaster which sounds exactly the 
same as an already existing, previous master? 
The whole point of a remaster is to make it sound 
different![i/]
_
You just torpedoed the ENTIRE music remastering business with tha statement, and don't know it yet!  You confirmed everything I already knew about 'remastering': that what is actually done is just a ploy to get people to think they are paying for and owing a better version when what they have is good enough.

I was trained to believe remastering was part of a restoration and preservation process, not one akin to adding 40 stories to the Empire State and selling it as new and improved.  Be careful what you type next time.


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## Cutestudio

bigshot said:


> And I suggest that he struggle to understand that the use of compression in a mix is not the same as the loudness war. Compression is a tool that is used in almost all mixes, even the ones from the golden past that he's championing.
> 
> Amirm, our ability to hear is directly related to how sound engineers mix. They mix to what the human ear can hear. The goal is clarity and organization of sound. They don't create a mix according to numbers on a page. Commercial mixes stay in a dynamic range of 45 to 50dB because that is what ears can hear without having to adjust. Do some simple googling and you'll figure it out. I'm not going to the trouble to scribble on a book with yellow highlighter and scan it when you really aren't interested in anything but your own words.
> 
> I've explained both of these things many times and it doesn't seem to register. Ignorance is OK. As Mark Twain said, "Everyone is ignorant... just on different subjects." I know stuff. You know stuff. We should find a way to communicate, not throw up roadblocks like you're doing. I'm here to communicate, not to go in circles repeating the same thing over and over again. When people do that, I start skimming over the repetitive parts of their posts and speak past them in my replies for the benefit of the lurkers. I'll start talking past you and the witch hunt duo too soon.



You and Greg are the ones who are deliberately confusing studio compression with over-compression and clipping used to ruin masters for todays Loudness War.

Only you two. Everyone else here know perfectly well what we are discussing. 

Your defence of the loudness war and 16 bit is contradicted by both of you are various times, none of us have ANY idea what you are trying to achieve by this and I suspect neither have you, the only thing I can think of is a sad compulsion to argue for the sake of it or that that somehow you both rely on the sales of CDs and CD players, which are the only possible use for obsolete formats that you spend even your Saturdays to vigorously defend. Grow up. Embrace the future. Your views are outdated, irrelevant and frankly wrong. Badly wrong.

SonicTruth: You are correct of course, but various people including you Amirm have done some fantastic posts which outweigh the tiresome luddite element here.


----------



## gregorio

TheSonicTruth said:


> [1] _You just torpedoed the ENTIRE music remastering business with tha statement ...
> [2] You confirmed everything I already knew about 'remastering': that what is actually done is just a ploy to get people to think they are paying for and owing a better version when what they have is good enough._



1. No I didn't. You didn't answer the question, what's the point of paying a mastering engineer to create a master which sounds exactly the same as a previous master they've already paid for?
2. Some remasters are better, not all of them are severely over-compressed. And, in addition to what seems obvious there's the question of better for whom, better for the audiophile in his silent room and 40k speakers or better for the average Joe listening on his ear buds on the way to work?

G


----------



## RRod

amirm said:


> What is a "normal living room?"  People listening with headphones which has no rooms.  And they have dedicated listening rooms.  We cannot impose limits on fidelity of music by assuming people sit in their living room to listen.
> 
> And to what end anyway?  The music industry has already started to distribute higher resolution music.  What is the point of campaigning against it?  You want them to stop?  To what end?
> 
> What is the logic of this discussion anyway?  It is not like we are 20 years ago and fighting over SACD/DVD-A vs CD.  We are in digital age where there is no barrier to distribution of high-resolution music and its playback.  You guys want to turn the tide back?  Why?



Most headphones do not have the isolation of the Etymotics you've referenced, and even then, I have yet to find an amp that powers mine with a noise floor as low as you seem to be expecting. And yes, many people consider speakers to be the actual gold standard for playback, and those speakers tend to be in houses that aren't Skywalker.

I am fine with hi-res, and some sites even sell it as reasonable prices which I happily buy, because why not? Other sites want to charge double price for *each* of the stereo and surround mix!

Funny you bring up SACD an DVD-A, two formats designed to provide multichannel audio and to need special hardware that disallowed digital copying: not exactly headphone friendly material, that.


----------



## Strangelove424 (Nov 18, 2017)

amirm said:


> I think there is still some misunderstanding about the data.  The numbers I have presented from research are *peak SPLs. *All this other data you see and measurements people use are averaged numbers, not instantaneous peaks.
> 
> With respect to what has to exist in a digital channel, we have no choice but to use peak numbers because that is what the information is that must be stored.  Average SPL numbers are used for other uses but not here.
> 
> ...



Bigshot is right, you are disingenuous. I never mentioned the word average, I always said peak or max. This entire discussion, and the paper you’ve scanned snippets from, are all in context to peak level. You know full well that I was referring to peak levels. And you know full well the UCSD link I gave you referred to peak levels too. I hate to backtrack in conversation - redefining the obvious - because people play dumb.

correction: upon checking my post, I did use the word average once, but was referring to the average peak level from multiple performances measured, not average SPL. You are twisting my use of the term as it applies to the data.


----------



## TheSonicTruth

Cutestudio said:


> You and Greg are the ones who are deliberately confusing studio compression with over-compression and clipping used to ruin masters for todays Loudness War.
> 
> Only you two. Everyone else here know perfectly well what we are discussing.
> 
> ...




Thanks, CS:

I might be a luddite too: I have and enjoy my aforementioned mostly pre-remaster CD collection, and have no issues with 16bit 44.1k audio fidelity.  And for gegorio's and bigshot's information: there is no need to over-compress or peak-limit dynamic material to make it *fit* within the redbook palette.

Some folks on here need to visit this site regarding the 16 VS 24bit deliverable debate:

https://people.xiph.org/~xiphmont/demo/neil-young.html

And _watch the video!_


----------



## Arpiben

amirm said:


> I think there is still some misunderstanding about the data.  The numbers I have presented from research are *peak SPLs. *All this other data you see and measurements people use are averaged numbers, not instantaneous peaks.
> 
> With respect to what has to exist in a digital channel, we have no choice but to use peak numbers because that is what the information is that must be stored.  Average SPL numbers are used for other uses but not here.
> 
> ...



By the way, those with no access (free) to the *peak SPLs` *research papers you partially presented keep having no idea at all about SPLs peak measurement resolution for instance !
Add the out of context sentence you partially mentioned from a Xivero MQA`s white paper, it doesn`t help either: 
#*Nevertheless, during our statistical analysis we have learned that sometimes well dithered native high resolution audio files still have audio information within the 2nd LSB and it would be only viable to throw away one bit to avoid losing any critical bit-depth*.#
Why should one lose time reading documents you are presenting if like the one above the statistical analysis is limited to the sentence?


----------



## amirm

gregorio said:


> 2. The music industry started to distribute higher res over 15 years ago and continues to distribute higher and higher res, we've got DACs out there capable of 768/32 and 368/32 or DSD64 content. It will never end and audiophiles will continue to be scammed on equipment and content ad infinitum because a higher res is always assumed to be better than a lower res. That's why we argue against it.


There is no "scamming" on equipment.  I have $120 DACs that play DSD64. 

What we should all be bothered by is what your industry is delivering to us in that larger container.  So far it looks like a ton of garbage in a number of instances.  See my videos here: https://www.youtube.com/channel/UCVRhwKlWccYyoCcS3r0TZ0Q?view_as=subscriber

[watch this full screen]


You need to get your house in order as far as I am concerned.  I sure as heck don't expect to pay more for high-res and then get handed tones from computer monitors as such inside it.

Why not start a campaign in your industry to perform QC on what is produced?  That is a real problem that needs sorting out rather than telling audiophiles what is good for them. 



> I ask you AGAIN, what do you need that's beyond the 120dB limitation of 16bit?


There is no "120 db" limitation of 16 bit.  That only happens with noise shaping.  Maybe you come back and tell me what percentage of the music you own has noise shaping in it.  But I will put my money on you not having that data and just throwing this number out.

Second, to take proper advantage of noise shaping, you need somewhere to push the noise up to.  Higher sampling rates allow that because we can park the noise in ultrasonics.

But yes, if I had any confidence that the people creating music understood noise shaping and utilized it correctly, you would kind of, sort of, have an argument.  But I don't.

So better to not rely on them making any conversion from 24 bits for me.  Give me the 24 bits, and I can convert it to 16 bits with noise shaping if I want.  Or play as is since every equipment these days plays that just fine.  Thank you very much.

You see the message here?  We want less finger in the soup that are not washed.


----------



## gregorio (Nov 18, 2017)

Cutestudio said:


> [1] You and Greg are the ones who are deliberately confusing studio compression with over-compression and clipping used to ruin masters for todays Loudness War.
> [2] Your defence of the loudness war ...
> [3] Your views are outdated, irrelevant and frankly wrong. Badly wrong.



1. What is the difference? Go on, explain it to me ... This should be amusing! 
2. Still can't read then, sad.
3. Of course they are, what else can you say when arguing from a position of complete ignorance?

G


----------



## RRod

bigshot said:


> Yes. Making a great sounding recording usually involves using compression. Especially with vocals.
> Yes, the conductor is up there adjusting balances to keep the range within a comfortable level.
> 
> More isn't better. Just right is best. A dynamic range greater than 50dB in a single song is uncomfortable to listen to. You'd be running to grab the volume control to turn it up in the quiet parts and turn it down in the loud parts- essentially manually compressing it.



Yes, for instance, something like this:



really doesn't give you much leeway for where you set the pot. If you try to take an edge off that blast, the soft part sounds insipid.


----------



## TheSonicTruth

RRod said:


> Yes, for instance, something like this:
> 
> 
> really doesn't give you much leeway for where you set the pot. If you try to take an edge off that blast, the soft part sounds insipid.




Could that be from a certain Telarc Tchaikovsky release?


----------



## Strangelove424

TheSonicTruth said:


>




Is that not exactly what you've been clamoring for? Do you really expect that to sound good?


----------



## Strangelove424

While you're at it, maybe you can knock some sense back into yours. You see the 20db worth of dynamics before the peak? That's *some* dynamics. The peak is *some* headache. There's such a thing as too much. Too much, too little, and just right.


----------



## TheSonicTruth (Nov 18, 2017)

Strangelove424 said:


> Is that not exactly what you've been clamoring for? Do you really expect that to sound good?




That emoji is the closest I could find to 'OOH LA-LAA'  

It is indicative of what digital can more than accommodate, if it were allowed to.

And I am not the only one "clamoring" for better fidelity in mass produced recordings lately.  This is not my 'lone crusade', as many on here, in Gearslutz, Steve Hoffman forums, and rec.audio.pro(Usenet) have insinuated.

Now am I more vocal about it than others?


Someone has to be. This cause is worth it!


----------



## RRod

TheSonicTruth said:


> Could that be from a certain Telarc Tchaikovsky release?



No, the context of the cannons isn't so cruel. This is Jón Leifs' Saga Symphony. Look it up; eClassical gives free previews of everything. It was recorded on one of these. There's some extra percussion in that hit that really give it a strong metallic punch. Good times! Leifs is a composer who wreaks havoc with volume normalization algorithms.


Strangelove424 said:


> Is that not exactly what you've been clamoring for? Do you really expect that to sound good?



It sounds great if you're willing to take the hit…


----------



## Strangelove424

TheSonicTruth said:


> That emoji is the closest I could find to 'OOH LA-LAA'
> 
> It is indicative of what digital can more than accommodate, if it were allowed to.
> 
> ...



One of your posts was just removed, probably because of your expression of violence. You have crossed the line from 'vocal' to irrational temper tantrum. I've found that the louder and angrier a person gets, the more shallow the substance of their argument, otherwise they'd fall back on that instead.  

I also noticed you ignored my statement regarding the waveform in question. You're back to speaking in pure hypotheticals. You will not admit that there is such a thing as too much dynamic range, and that makes you an absolutist, a zealot, or an ideologue... not a creator, or even a connoisseur.


----------



## TheSonicTruth (Nov 18, 2017)

Strangelove424 said:


> One of your posts was just removed, probably because of your expression of violence. You have crossed the line from 'vocal' to irrational temper tantrum. I've found that the louder and angrier a person gets, the more shallow the substance of their argument, otherwise they'd fall back on that instead.
> 
> I also noticed you ignored my statement regarding the waveform in question. You're back to speaking in pure hypotheticals. You will not admit that there is such a thing as too much dynamic range, and that makes you an absolutist, a zealot, or an ideologue... not a creator, or even a connoisseur.




Never heard your mom say that to you and a sibling?  She'd love to "knock your heads together" for something or other you might've done in your spare time together? lol  Older than me, that expression, for sure! lol

Now as far as what you said about that waveform?  Sure!  If one has equipment with the chutzpah to reproduce it faithfully.


----------



## Strangelove424

RRod said:


> It sounds great if you're willing to take the hit…



Everything before Mt. Everest looks great.


----------



## Strangelove424

TheSonicTruth said:


> Never heard your mom say that to you and a sibling?  She'd love to "knock your heads together" for something or other you might've done in your spare time together? lol  Older than me, that expression, for sure! lol



You're not my mom!


----------



## castleofargh

if when playing a song you feel the urge to reach and change the volume halfway through, in my book that doesn't show good mastering. on the contrary, I will think the guy was incompetent for not understanding that variations were too wild for proper enjoyment of both passages. 
just trying to think about stuff I have with wild changes, I fall on classical music where I don't see the issue because the albums are usually not overly compressed so there isn't much to cry about in the first place. and I find mostly old albums where several aspects of the mastering were just different from what they are today. it was fine to have a full song recorded with peaks at -20dB, it was fine to have voices panned 100% on one side, it was fine to have long and very slow start... for better or for worst the world is changing, our consumer's habits have changed too.
anyway if I take something like Locomotive Breath, the decay of the piano notes at the beginning seems to go about 60dB below some of the true peaks in the song. to me that's already an extreme, if I set the volume level for the intro, then the rest ends up loud for my taste, and if I just keep the volume like I have it most of the time, then the piano is really quiet and the decay is something I won't really notice, well recorded or not.  this here is a very personal opinion, but I don't desire wider variations on a song, most of the time I'll wish for less. 

on the event that some album requires more than 16bit do properly record various parts at various sound levels, then I'm tempted to suggest to release it in 24bit only and good luck to the listeners. I don't see how even those weirdo albums make a case for not using 16bit for what clearly doesn't need more bits for the music content when even the quiet passages are background hiss at -60 or -70dB. 

about Don Quixote and Sancho Panza courageously fighting the loudness war by attacking anything that moves, again, try to have a real look at what you're fighting. dynamic compression is a welcome part of mastering, not some evil giant. how it's abused is sad but you guys have such a single minded look on things that even while being against the loudness war, people end up arguing against you. 



Strangelove424 said:


> One of your posts was just removed, probably because of your expression of violence. You have crossed the line from 'vocal' to irrational temper tantrum. I've found that the louder and angrier a person gets, the more shallow the substance of their argument, otherwise they'd fall back on that instead.
> 
> I also noticed you ignored my statement regarding the waveform in question. You're back to speaking in pure hypotheticals. You will not admit that there is such a thing as too much dynamic range, and that makes you an absolutist, a zealot, or an ideologue... not a creator, or even a connoisseur.


"the dignity of truth is lost with much protesting". ^_^
and yes I just happened to look at the topic when the regrettable post happened. we all get mad when we're passionate about things, going often too far myself, I try to forgive others so that they'll forgive me later. I hope you can see it for such a momentary lapse in self control, and forgive. of course if things have to stay that heated, I'll go get a grown up to sort things out for good. 

now if we could go back to discussing audio instead of people, that would make me happy.


----------



## Strangelove424

castleofargh said:


> "the dignity of truth is lost with much protesting". ^_^
> and yes I just happened to look at the topic when the regrettable post happened. we all get mad when we're passionate about things, going often too far myself, I try to forgive others so that they'll forgive me later. I hope you can see it for such a momentary lapse in self control, and forgive. of course if things have to stay that heated, I'll go get a grown up to sort things out for good.
> 
> now if we could go back to discussing audio instead of people, that would make me happy.



Agreed. Good quote. I am over it. Water under the bridge. Let's get back to substance.


----------



## TheSonicTruth (Nov 18, 2017)

castleofargh said:


> if when playing a song you feel the urge to reach and change the volume halfway through, in my book that doesn't show good mastering. on the contrary, I will think the guy was incompetent for not understanding that variations were too wild for proper enjoyment of both passages.
> just trying to think about stuff I have with wild changes, I fall on classical music where I don't see the issue because the albums are usually not overly compressed so there isn't much to cry about in the first place. and I find mostly old albums where several aspects of the mastering were just different from what they are today. it was fine to have a full song recorded with peaks at -20dB, it was fine to have voices panned 100% on one side, it was fine to have long and very slow start... for better or for worst the world is changing, our consumer's habits have changed too.
> anyway if I take something like Locomotive Breath, the decay of the piano notes at the beginning seems to go about 60dB below some of the true peaks in the song. to me that's already an extreme, if I set the volume level for the intro, then the rest ends up loud for my taste, and if I just keep the volume like I have it most of the time, then the piano is really quiet and the decay is something I won't really notice, well recorded or not.  this here is a very personal opinion, but I don't desire wider variations on a song, most of the time I'll wish for less.
> 
> ...





"_if when playing a song you feel the urge 
to reach and change the volume halfway through, 
in my book that doesn't show good mastering. on 
the contrary, I will think the guy was incompetent 
for not understanding that variations were too wild 
for proper enjoyment of both passages._"


That has more to do with inconsistent average levels(which we judge loudness mostly by) not with transients.  A great recording has a steady average level, yet retains most of the transient attacks that help lend life to it, if left in.  Slightly louder choruses, in the refrains between verses in a song, also give it life, and should not send someone scurrying for the volume control all the time.


----------



## danadam

TheSonicTruth said:


> And for gegorio's and bigshot's information: there is no need to over-compress or peak-limit dynamic material to make it *fit* within the redbook palette.



Where did they ever say or imply that compression is to fit into 16 bit or that 16 bit is not enough for anything?

@gregorio , @bigshot , you have my sympathies. If I were you I'd probably hurt myself banging my head against the desk


----------



## TheSonicTruth

danadam said:


> Where did they ever say or imply that compression is to fit into 16 bit or that 16 bit is not enough for anything?
> 
> @gregorio , @bigshot , you have my sympathies. If I were you I'd probably hurt myself banging my head against the desk




Throughout this entire thread.


----------



## TheSonicTruth

Strangelove424 said:


> You're not my mom!



Never implied at I was, and my use of that ancient phrase was in gest.   I really do feel alone in forums like these when it comes to championing a balance between dynamics and overall loudness.


----------



## gregorio

amirm said:


> [1] There is no "scamming" on equipment.  I have $120 DACs that play DSD64.
> [2] What we should all be bothered by is what your industry is delivering to us in that larger container.  So far it looks like a ton of garbage in a number of instances. I sure as heck don't expect to pay more for high-res and then get handed tones from computer monitors as such inside it.
> [3] Why not start a campaign in your industry to perform QC on what is produced?
> [4] But yes, if I had any confidence that the people creating music understood noise shaping and utilized it correctly, you would kind of, sort of, have an argument.  But I don't.
> ...



1. Let's say your $120 DAC plays audibly perfectly, noise and artefacts below -120dB. What about another DAC which also performs audibly perfectly, with artefacts below -120dB but costs $2k instead of $120? It's pretty much guaranteed that $2k DAC is being sold as having audibly superior performance but unless you're happy to pay an extra $1,880 just for a prettier case, how is that not an equipment scam?

2. Of course it's a ton of garbage, what did you expect? You wanted better fidelity, you're getting it, all of it; computer monitors, other electrical interference, mechanical noise from the musical instruments, a ton of garbage plus some small amount of probably the twentieth and higher harmonics of actual musical notes. I stated all this on this very thread more than 5 years ago! None of us engineers can hear anything up there, so how on earth can we process it? With 96/24, at least three quarters of the data we've recorded we can't hear. Heck, even with 16/44 we probably can't hear at least a quarter of it! There's little we can do about it, recording sessions have a flow, they are run around the talent, to encourage the best performance. Making the talent wait because of some serious technical issue can cause the talent to loose the vibe and if the talent happens to be a 100 piece orch, then it's disastrous but making them wait for a technical issue which isn't even remotely audible would be absolutely ridiculous!

3. Because I can't QC what I can't hear. There's more than enough to be concerned about in the audible range, I don't have time to worry about what's way, way outside of human hearing. If you don't like or want a ton of garbage, my advice is to try a low pass filter, if you set it at around 20kHz you'll get rid of most of the garbage and cause little/no damage to audible freqs!

4. Well, you've kind of/sort of answered one of my questions, yea! Now what about the next one, where noise-shaped dither is not used, why do you think it has not been used?

5. Fine, what about 32bit, you obviously want that because you quoted that you've got some? What about 64bit, that's what most commercial music is created in these days, don't you want that too? ...
5a. You see the message here? When does it end? Answer: It will never end! As long as audiophile manufacturers and content distributors can come up with marketing which audiophiles will believe, sample rates and bit depths will continue to increase, regardless of the fact that the music performance they think they're buying will occupy a smaller and smaller percentage of what they're buying and more and more of it will just be garbage, which fortunately they won't be able to hear anyway, unless it causes IMD of course but then that could be marketed as analogue sounding, yea!

G


----------



## Strangelove424

TheSonicTruth said:


> Never implied at I was, and my use of that ancient phrase was in gest.   I really do feel alone in forums like these when it comes to championing a balance between dynamics and overall loudness.



You are not alone, the problem is you are taking an absolutist perspective on a subject of balance, and it is frustrating the people who have to make decisions of balance every day. You think I am opposing you, but I champion quality mastering and dynamics too. I once started thread to analyze the quality of mastering (dynamics included) because I was frustrated with the quality out there. In retrospect, measuring or testing for mastering quality was too difficult, and looking at spectrograms, waveforms, and DR values (as well meaning as it was) didn't fully capture the quality of the mastering. Does it help? Sure, but if you aren't willing to listen to your ears first, there's no hope the numbers will guide you anywhere on their own. Look around. You think we all like compressed pop? No, but we don't like running for the remote either when a movie or opera gets way too loud. If you gave a more honest look at the people in these communities (all of us included), as well as taking a less ideological and more balanced perspective on your own ideals, you probably wouldn't feel so alienated and outcast. Quite the opposite, you'd realize this is a haven of people who truly love and respect audio.


----------



## RRod

Strangelove424 said:


> Everything before Mt. Everest looks great.



He's got the whole Himalaya in there:


----------



## gregorio

TheSonicTruth said:


> And I am not the only one "clamoring" for better fidelity in mass produced recordings lately.



No, you're not the only one but unfortunately you are one of only a very tiny minority. The vast majority do not want better fidelity, they'd far rather have lower prices, instant access and convenience, fidelity is a distant fourth place! Long gone are the days of record labels giving hundreds of thousands of dollars for a band to go and experiment in world class recording studio for 6 months. In fact, most of those world class studios themselves have gone in the last 15-20 years and many/most of those which remain barely break even or operate at a loss. Along with them has gone the knowledge and experience of the engineers. Forget 6 months to record, 6 weeks to record, mix and master an album is the norm these days! Not only do consumers expect to pay far less for their music than they once did, we've now also got the big aggregators such as Apple, taking their 30% of what's left. As bad and dodgy as the record labels were, at least they reinvested a fair amount of their earnings in actually creating music recordings, Apple and other aggregators take their billions and invest absolutely nothing in creating music. That's the way it is, that's the landscape in which we have to survive! I'd love to be given the time to chase that last few percent of fidelity but there's not the money reaching the artists to enable any of us to do that. Even the very top artists only earn a small fraction of their income from recordings these days. It used to be that tours were organised as a tool to market the recordings, today it's the other way around, albums are now little more than promotional material for the tours! Maybe, one day, fidelity will come back into fashion and consumers will be prepared to pay en-masse for it but that's not in the foreseeable future.

G


----------



## JaeYoon

gregorio said:


> No, you're not the only one but unfortunately you are one of only a very tiny minority. The vast majority do not want better fidelity, they'd far rather have lower prices, instant access and convenience, fidelity is a distant fourth place! Long gone are the days of record labels giving hundreds of thousands of dollars for a band to go and experiment in world class recording studio for 6 months. In fact, most of those world class studios themselves have gone in the last 15-20 years and many/most of those which remain barely break even or operate at a loss. Along with them has gone the knowledge and experience of the engineers. Forget 6 months to record, 6 weeks to record, mix and master an album is the norm these days! Not only do consumers expect to pay far less for their music than they once did, we've now also got the big aggregators such as Apple, taking their 30% of what's left. As bad and dodgy as the record labels were, at least they reinvested a fair amount of their earnings in actually creating music recordings, Apple and other aggregators take their billions and invest absolutely nothing in creating music. That's the way it is, that's the landscape in which we have to survive! I'd love to be given the time to chase that last few percent of fidelity but there's not the money reaching the artists to enable any of us to do that. Even the very top artists only earn a small fraction of their income from recordings these days. It used to be that tours were organised as a tool to market the recordings, today it's the other way around, albums are now little more than promotional material for the tours! Maybe, one day, fidelity will come back into fashion and consumers will be prepared to pay en-masse for it but that's not in the foreseeable future.
> 
> G


 That's the sad thing, the artists who create the music should make way more money than they are now!


----------



## TheSonicTruth (Nov 18, 2017)

gregorio said:


> No, you're not the only one but unfortunately you are one of only a very tiny minority. The vast majority do not want better fidelity, they'd far rather have lower prices, instant access and convenience, fidelity is a distant fourth place! Long gone are the days of record labels giving hundreds of thousands of dollars for a band to go and experiment in world class recording studio for 6 months. In fact, most of those world class studios themselves have gone in the last 15-20 years and many/most of those which remain barely break even or operate at a loss. Along with them has gone the knowledge and experience of the engineers. Forget 6 months to record, 6 weeks to record, mix and master an album is the norm these days! Not only do consumers expect to pay far less for their music than they once did, we've now also got the big aggregators such as Apple, taking their 30% of what's left. As bad and dodgy as the record labels were, at least they reinvested a fair amount of their earnings in actually creating music recordings, Apple and other aggregators take their billions and invest absolutely nothing in creating music. That's the way it is, that's the landscape in which we have to survive! I'd love to be given the time to chase that last few percent of fidelity but there's not the money reaching the artists to enable any of us to do that. Even the very top artists only earn a small fraction of their income from recordings these days. It used to be that tours were organised as a tool to market the recordings, today it's the other way around, albums are now little more than promotional material for the tours! Maybe, one day, fidelity will come back into fashion and consumers will be prepared to pay en-masse for it but that's not in the foreseeable future.
> 
> G



And for me anyway, the biggest casualty of this loudness trend is the PAST:  Remasters of 1960s and '70s R&B, Classic Rock, Pop, Country, etc have been subject to the same loudifying processes as modern releases(2000 to present). 

And I understand the reasoning: It started with 5-50 disc CD changers, when consumers shuffled older '80s era CD issues with late 1990s(early loudness war era) CDs, and contiued with file-based versions of legacy artists mixed in with louder modern files.  Those listeners found themselves periodically adjusting the volume - at home, behind the wheel, when exercising, etc.  To get the legacy stuff even *close* to the perceived loudness of the modern stuff, a combination of compression and peak limiting had to be performed, plus anywhere from 4-8dB of makeup gain! And hence were born the first round of 'Digitally Remastered!!' classic rock, etc. CDs.

So I could care less what happens to stuff made more recently, or if artists finally come to their senses, but please, leave our classic hits alone!!


----------



## TheSonicTruth (Nov 18, 2017)

JaeYoon said:


> That's the sad thing, the artists who create the music should make way more money than they are now!



They start making more money when they reject loudness as the sole criterion, in favor of fidelity, and start to sell more albums and downloads.

Nothing wrong with their art, just the way it is packaged/mastered.


----------



## sonitus mirus

JaeYoon said:


> That's the sad thing, the artists who create the music should make way more money than they are now!



They should!

Now to play the devil's advocate.  

I generally agree, but distributing the art and finding a method to market it in such a way that it generates a profit is as much a part of the success in this industry as anything else.  I fully understand that artists are oftentimes taken advantage of and those with little to no talent to create the "product" can make a great living off the efforts of many of these artists.   I do a lot of technical stuff that my bosses and their bosses could probably never do, yet they make more money than I do.  I also think that I could do their jobs as well or better than they do.  But here I am, doing what I do, and there they are, doing what they do.  Musicians are probably no different then most other people that work a job of one type or another.

In the end, it comes down to the notion that if the artists want a bigger cut of the pie, then why don't they just make it happen themselves?  It's not easy.  I love the artist's work, but I also love the people that are responsible for bringing it to me.  I mean, think about what is involved to get a song in a format that I can play and enjoy where I want to listen.  How much of that is because of the artist?  You can be the greatest talent in the world, but if you only sing to yourself in the shower, it doesn't do anything for me or other people that might enjoy it.


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## old tech (Nov 18, 2017)

JaeYoon said:


> That's the sad thing, the artists who create the music should make way more money than they are now!


There are some positives though...  The entry point for artists, particularly with the indie labels, is a lot lower now than what it used to be.  In the past, artists were reliant on getting a record contract or the whims of radio stations for exposure.  Us baby boomers (mainly) think of the glorious days of rock in the 70s but think of how much good music would have been around but never seen the light of the day.

While it is true that most of the mainstream artists make less money from record sales than in the past, there are now a lot more opportunities for those on the periphery. Even so, those mainstream artists who are clever enough to use modern technology are able to make far more money than in the days when they were totally reliant on the whims of the labels.  Think of Taylor Swift, regardless what you might think of her music, she as been quite deft in maneuvering around Apple and other distributors to make them work for her rather than the other way round, for example forcing Apple to back down on not paying royalties on the first three months of a release (a win that benefited all artists) and only sells her music on Itunes and CDs (where the margins are higher).  She uses media channels unrelated to the music industry to communicate directly with her fans, a trend will spread more widely with modern technology.

The music industry is just one example of a business model that has been disrupted by technology.  As with all disruptions it brings benefits as well as costs.  One of the costs, as Gregorio has pointed out, is the human input to the recording is being diminished.  It is ironic that the music producing technology has progressed leaps and bounds but the human element has declined meaning we don't get the benefits on the production side (apart from lower costs).  The fact that many vinyl records and early analog tape recordings can sound better than modern digital because of the lack of human time in the process should be shameful and of serious concern to audiophiles rather than something to be celebrated, as is the case on some forums like the Hoffman site.


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## sonitus mirus

Now I am curious to hear this guy Taylor Swift.  That group must be good.  Which one is Taylor? 

To be honest though, I could not name a single song that Taylor Swift swings or even recognize a song by her if I heard it played.  Too bad that I fast-forwarded her sessions on Saturday Night Live recently.  I have discovered more than a few artists from SNL over the years.  There is plenty of room in this industry for someone like me that listens to over 40 songs each day, and those fans of Taylor Swift, to exist together.


----------



## old tech

sonitus mirus said:


> Now I am curious to hear this guy Taylor Swift.  That group must be good.  Which one is Taylor?
> 
> To be honest though, I could not name a single song that Taylor Swift swings or even recognize a song by her if I heard it played.  Too bad that I fast-forwarded her sessions on Saturday Night Live recently.  I have discovered more than a few artists from SNL over the years.  There is plenty of room in this industry for someone like me that listens to over 40 songs each day, and those fans of Taylor Swift, to exist together.


That's part of the point, we have easy access to a lot more music of different variety these days.


----------



## sonitus mirus

old tech said:


> That's part of the point, we have easy access to a lot more music of different variety these days.



Maybe there are too many artists creating great music?  Is the market saturated?  I guess a big problem for the industry is that great music from the past is always available to consumers at a reasonable cost, and there is only a limited amount of music that consumers can "use".


----------



## JaeYoon

old tech said:


> There are some positives though...  The entry point for artists, particularly with the indie labels, is a lot lower now than what it used to be.  In the past, artists were reliant on getting a record contract or the whims of radio stations for exposure.  Us baby boomers (mainly) think of the glorious days of rock in the 70s but think of how much good music would have been around but never seen the light of the day.
> 
> While it is true that most of the mainstream artists make less money from record sales than in the past, there are now a lot more opportunities for those on the periphery. Even so, those mainstream artists who are clever enough to use modern technology are able to make far more money than in the days when they were totally reliant on the whims of the labels.  Think of Taylor Swift, regardless what you might think of her music, she as been quite deft in maneuvering around Apple and other distributors to make them work for her rather than the other way round, for example forcing Apple to back down on not paying royalties on the first three months of a release (a win that benefited all artists) and only sells her music on Itunes and CDs (where the margins are higher).  She uses media channels unrelated to the music industry to communicate directly with her fans, a trend will spread more widely with modern technology.
> 
> The music industry is just one example of a business model that has been disrupted by technology.  As with all disruptions it brings benefits as well as costs.  One of the costs, as Gregorio has pointed out, is the human input to the recording is being diminished.  It is ironic that the music producing technology has progressed leaps and bounds but the human element has declined meaning we don't get the benefits on the production side (apart from lower costs).  The fact that many vinyl records and early analog tape recordings can sound better than modern digital because of the lack of human time in the process should be shameful and of serious concern to audiophiles rather than something to be celebrated, as is the case on some forums like the Hoffman site.


That's true, nowadays we have sites like bandcamp and many many more, soundcloud, beatport where artists can upload their material, youtube and gain exposure!

Posting of live events where fans can gather!


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## old tech

sonitus mirus said:


> Maybe there are too many artists creating great music?  Is the market saturated?  I guess a big problem for the industry is that great music from the past is always available to consumers at a reasonable cost, and there is only a limited amount of music that consumers can "use".


Interesting question, is the market saturated?  I don't think so unless one is set in their ways like only listening to music they grew up with.

It is not like a market saturation of commoditised products, as songs/albums are different and appeal to different people.  The issue perhaps is whether there is too much choice but again, that should not be a problem if there are various ways of accessing new music.  Although I don't subscribe to any streaming services, I think the algorithms many of them have which can play new music depending on your past tastes is one way of being exposed to a broader market.


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## Strangelove424 (Nov 19, 2017)

TheSonicTruth said:


> And for me anyway, the biggest casualty of this loudness trend is the PAST:  Remasters of 1960s and '70s R&B, Classic Rock, Pop, Country, etc have been subject to the same loudifying processes as modern releases(2000 to present).
> 
> And I understand the reasoning: It started with 5-50 disc CD changers, when consumers shuffled older '80s era CD issues with late 1990s(early loudness war era) CDs, and contiued with file-based versions of legacy artists mixed in with louder modern files.  Those listeners found themselves periodically adjusting the volume - at home, behind the wheel, when exercising, etc.  To get the legacy stuff even *close* to the perceived loudness of the modern stuff, a combination of compression and peak limiting had to be performed, plus anywhere from 4-8dB of makeup gain! And hence were born the first round of 'Digitally Remastered!!' classic rock, etc. CDs.
> 
> So I could care less what happens to stuff made more recently, or if artists finally come to their senses, but please, leave our classic hits alone!!



I agree that it’s a shame what’s happening to remasters of those genres. If you have holes in your collection and want to fill them with decent masters, you have to go looking for used CDs from the late 80s or early 90s. I don’t mind 320kbps streaming quality, and MOG used to have the entire catalog of releases, so you could go all the way back to the original version. Unfortunately, neither Spotify nor Tidal have that capability, and MOG is gone.

Don’t give up on modern music. I believe Daft Punk’s Random Access Memories is one of the best mastered albums ever made. The mastering on the vinyl version is incredible, but the CD release sounds good too. They’re rare, but crafted works are out there. Whether or not you like EDM or Daft Punk, this is an excellent article about the qualities of the mastering, and will give you a stronger idea of the decisions that go into the process, the compromises that have to be made, and why good work is a matter of balance:

http://productionadvice.co.uk/daft-punk-mastering/

It might not hurt to read these either...

http://productionadvice.co.uk/its-all-about-great-sound/

http://productionadvice.co.uk/how-loud-is-too-loud-when-dr-values-just-arent-enough/


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## TheSonicTruth (Nov 19, 2017)

Strangelove424 said:


> I agree that it’s a shame what’s happening to remasters of those genres. If you have holes in your collection and want to fill them with decent masters, you have to go looking for used CDs from the late 80s or early 90s. I don’t mind 320kbps streaming quality, and MOG used to have the entire catalog of releases, so you could go all the way back to the original version. Unfortunately, neither Spotify nor Tidal have that capability, and MOG is gone.
> 
> Don’t give up on modern music. I believe Daft Punk’s Random Access Memories is one of the best mastered albums ever made. The mastering on the vinyl version is incredible, but the CD release sounds good too. They’re rare, but crafted works are out there. Whether or not you like EDM or Daft Punk, this is an excellent article about the qualities of the mastering, and will give you a stronger idea of the decisions that go into the process, the compromises that have to be made, and why good work is a matter of balance:
> 
> ...



Thanks for the links! But I know who they're from and I often find he talks a good game but doesn't always push hard enough.  He seems to think DR8 on the meter plugins is enough for commercial releases, when thirty-forty years ago DR12-15 was the norm for chart stuff! He and I agree respectfully to disagree.

I agee with you about RAM, and own that CD.  Charlie Puth's "Attention"(off his forthcoming self-produced mind you 'Voice Notes' album!) and some of Bruno's recent material also has my 'attention', not just in how good it sounds but the style and arrangement.  It actually takes a b r e a t h! lol


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## Cutestudio

gregorio said:


> No, you're not the only one but unfortunately you are one of only a very tiny minority. The vast majority do not want better fidelity, they'd far rather have lower prices, instant access and convenience, fidelity is a distant fourth place!



Can you point to the research that indicates only a tiny minority care about the sound quality?
Additionally can you qualify that research with the method of asking, because when I play a well mastered track to anyone young they suddenly realise what they were missing and become interested in better quality. Is this one reason why the record industry suppresses decent masters today?

Your statement is also illogical because you reframe the argument as a choice between cost and quality, whereas in reality not over-compressing and clipping costs exactly the same as mastering something properly. I was looking at 'Dirty Mind' from Shakespeare's Sister, the histogram is almost perfect and the dynamic range (Peak/RMS) is around 16dB and it sounds excellent. Compare that to pretty much any similar track from 'Girls Aloud' and theres have all the peaks squashed - resulting in a DR hovering around 10dB and sound dreadful unless heard very quietly on a very cheap band limited radio. 

There is no need to reframe the argument Greg, we KNOW that good mastering costs the same as a Mangled master. Probably cheaper in fact as it's a lot less work. AC/DC's Black Ice and some of A Fine Frenzy's tracks must have cost more to mangle because you can see they were mangled, then EQ'd, and then mangled again which took a special kind of moron at the mastering desk.

Between the two examples between Shakespeare's Sister and Girls Aloud there is no stylistic reason for the Mangling Engineer to be 'creative' and vandalise the sound, the Shakespeare's Sister track sounds quite realistic - as if the instruments are _real_ and the louder it's played the better it sounds. The modern mangling has no place in HiFi, Car or MP3 player and is turning people away from music in droves. Once you have removed all dynamics and any reason to turn it up or play it again what is left? Just a loud noise with some vocals and lyrics that only a tiny minority will enjoy listening to.

I.e. you appear to confuse your role as a creative one where we are supposed to appreciate the mangling, where in reality is should be an invisible one that allows us to hear the musicians.

I think the patronising attitude of the Mangling Engineers and their industry moved into the realm of fraud a long time ago, in fact I've returned quite a few CDs now because of poor quality mastering, people buying CDs or downloads have a reasonable expectation of quality and only a tiny - but growing - minority have any idea how bad the problem is. Too much of the HiFi industry is complicit in enabling this, CD players for example are designed very carefully to avoid overloading on clips - plug in a pro-audio DAC however and the overload light and clips are obvious, a much more honest system.

These overload lights are the same ones the witless cretins at final mastering treat as a badge of honor rather than the stark warnings they are. It's simple audio engineering 101: don't overload and don't clip, which appears to have been forgotten in the Great Dumbing Down of the past 30 years.

HiFi is a shadow of what it was in the 1970s and 1980s, an industry full of charlatans pushing snake oil digital wires, in 40 years it's _still_ using unbalanced interconnects and the lack of anything decent to play has turned many people away. I did find Chlara's version of 'Hotel California' recently though: very well mastered, so there are some people still who either care enough or simply don't know how to mangle music so the odd one slips through. The sound of that compared to the 99% of MangleMusic is of course night and day - and the extra marketing and processing cost of $0.00. 

The 'still sounds good as no one cared enough to mangle it' still applied to many DVDs, often a DVD sound track will be far hight quality simply because the sound engineers are just doing their job, not the 'special' creative types who's aim in life to to reduce all sound to a solid brick shaped wall of noise. I visited a consumer electronics show a few years back and it was revealing that the DVD soundtrack to King Kong was the best sounding thing there, only comparable to some old vinyl, the 'HiFi' was just an unpleasant wall of noise that needed turning down - it seems this was thanks to people like you and Bigshot being creative and knowing 'best'.

Amirm's idea is the smartest, sell the 24bit un-mangled versions for a premium. But the record industry is terminally stupid and has constantly strived to avoid this. When SACD came out the internet was slow enough that a simple DVD density disk of 96k/24 audio would have been worth buying on silver discs, leaving the mangled 16bit for MP3 which people were downloading anyway even over dial-up. They missed that money making opportunity so here we are 30+ years later with mangled 16bit silver discs that people simply bypass in favour of mangled MP3s because no one can tell the difference after the mangling.
It's still revealing that music is on sale at Apple and Amazon. Where is the RIAA or the record companies? Go to Virgin Records today (http://www.virginrecords.com/releases/) and you'll see they've just about worked out how to make a slow clunky webpage, but you can't even buy their product direct. Doh.   

With the Greg and Bigshot attitude I've been watching the 'tiny minority' of audiophiles become a self fulfilling prophecy as the 'experts' dance around the steaming pile of 'product' to justify the production of mangled 16bit for all. For what reason is a mystery besides the overarching need to be 'right'. It's not a good enough reason.


----------



## RRod

If I ask a random person at work "what do you do for music?", the answer is likely to be "I just wear whatever earbuds come with my iPhone, but I only really listen on the subway in to work, and I usually just stream whatever is on." Music that requires dedicated, low-background listening isn't the norm any more. Are you saying you have a real reason to deny this as reality?


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## TheSonicTruth

Cutestudio said:


> Can you point to the research that indicates only a tiny minority care about the sound quality?
> Additionally can you qualify that research with the method of asking, because when I play a well mastered track to anyone young they suddenly realise what they were missing and become interested in better quality. Is this one reason why the record industry suppresses decent masters today?
> 
> Your statement is also illogical because you reframe the argument as a choice between cost and quality, whereas in reality not over-compressing and clipping costs exactly the same as mastering something properly. I was looking at 'Dirty Mind' from Shakespeare's Sister, the histogram is almost perfect and the dynamic range (Peak/RMS) is around 16dB and it sounds excellent. Compare that to pretty much any similar track from 'Girls Aloud' and theres have all the peaks squashed - resulting in a DR hovering around 10dB and sound dreadful unless heard very quietly on a very cheap band limited radio.
> ...



Be careful Cute not to get into habit of blaming the engineers exclusively.  I'd like to believe that the majority of them are just fulfilling, as best as they can, some much better than ofhers, the demands of the ones paying for their services.  This includes the artists themselves, theie producers, and even the labels thaf are trying to build identities via a certain sound across all genres and artists they sign.

That is the segment that needs educating. The engineers know what over-compression and excessive limiting can do to a sound, and the good ones will demonstrate that to their paying clientele.

The best those that buy music can do right now is fight it with their wallets.


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## JaeYoon (Nov 19, 2017)

RRod said:


> If I ask a random person at work "what do you do for music?", the answer is likely to be "I just wear whatever earbuds come with my iPhone, but I only really listen on the subway in to work, and I usually just stream whatever is on." Music that requires dedicated, low-background listening isn't the norm any more. Are you saying you have a real reason to deny this as reality?


Yeah, when I joined this site I also believed in hi-fi and wanting to get great gear in order to enjoy music.
I also always see people with just Apple Earbuds or included ones for Samsung phones, etc or any phone manufacturer and just use those.

I will see people with like ATH M40/50 or Sony headphones, etc. But most people tend to just listen to the music that they have out of their phone, and pay less attention to the equipment.

I kinda hate generalizing/stereotyping this, but most modern folks in my city (Will definitely change on location and geography and trends). But vast majority of young adults and teens do listen to Electronic Genres and Hip Hop/R&B and a lot of Rock/Metal, like listening at loud volumes.
I don't think any of them care about dynamic range or how the music is packaged. Just that it's a reproduction of what they want to hear, most of these people are too busy with their lives to care about how their music is produced in studio, etc.


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## gregorio (Nov 19, 2017)

TheSonicTruth said:


> [1] And I understand the reasoning: It started with 5-50 disc CD changers, when consumers shuffled older '80s era CD issues with late 1990s(early loudness war era) CDs, and contiued with file-based versions of legacy artists mixed in with louder modern files.
> [2] To get the legacy stuff even *close* to the perceived loudness of the modern stuff, a combination of compression and peak limiting had to be performed, plus anywhere from 4-8dB of makeup gain!
> [3] So I could care less what happens to stuff made more recently, or if artists finally come to their senses, but please, leave our classic hits alone!!
> [4] They start making more money when they reject loudness as the sole criterion, in favor of fidelity, and start to sell more albums and downloads.



1. Not really. The loudness war started in the 1960s mainly due to radio, although one could argue it stated in the 40s with duke boxes. The record labels noticed that generally consumers preferred and bought the louder tracks. The artists and the labels wanted their new release to at least be as loud as everything else and hopefully a little louder. Then there were various bumps in the road where some artist/label released something much louder than anything previously, made a fortune and that level of loudness became the new standard which other artists felt they must match. The most famous example is probably Oasis' Morning Glory album, the problem being that it's extremely difficult to match that loudness, just smashing a track with a compressor/limiter is not enough, the track has to be composed, orchestrated and mixed a certain way, plus EQ'ed a certain way. Then came EDM, where you start with highly compressed samples in the first place, compose, orchestrate, successively build up layers of of compression and by the time it gets to mastering you sometimes can't add even the tiniest bit more compression. However, EDM needs to have a relatively small DR because it's designed for playback in extremely noisy environments (clubs, etc). Even going back to the late 60's and 70's layers of compression was common though. Maybe 8 or so dB compression on the kick drum, maybe slightly less on the snare, then another few dB on the drum sub-group, then more on the mix bus and then off to the mastering engineer for a bit more. The lead vox is definitely going to need some, probably 4dB at least, pretty much all of the instruments will probably need at least some, with the possible exception of synths/samplers/keyboards because they are effectively already highly compressed. And these rough figures are before Morning Glory! 20dB or more of compression/make up gain on some of the sounds/instruments is not uncommon. So, there's already a great deal of compression on a mix before it ever gets to the mastering engineer, typically way more than than the mastering engineer is going to add! In fact, a fair number of the mixes I'm given to master these days are a case of spending all my time trying to undo some of the most serious damage already done to the mix and then trying to get back to same loudness as the mix I was given to start with (but hopefully without as much damage)!

2. There's two issues here. What we want to end up with and what we are given to start with. If for example we're given a mix which peaks below 0dBFS and/or there are very few peaks and those peaks can be reduced with no audible damage (or even, made to sound aesthetically better), then 8dB of make up gain during mastering could be not only perfectly acceptable but very possibly preferable. On another track, that same 8dB of compression/make up gain could sound like ridiculous over-compression. There are a lot of variables at play here but quantifying the compression on a master is impossible, not least because most of the compression has already been applied before it's even sent to the mastering engineer! Likewise it's impossible to say how much make up gain/compression applied by the mastering engineer is acceptable.

3. See my response to sonitus mirus below: The labels cannot leave the classic hits alone! Their survival depends on leveraging their back catalogues, which are valued in the billions, and the shouts of a few (or even lots of) audiophiles means squat to global conglomerates fighting declining sales, lower profits and falling share prices!

4. Again, see my responses below. Plus; maybe they do sell more albums and downloads as a result of higher fidelity but probably not much more, as the vast majority of consumers don't really care about high fidelity. In all likelihood, those extra copies they sell wouldn't even cover the cost of making the recording high fidelity in the first place, let alone be profitable!



sonitus mirus said:


> In the end, it comes down to the notion that if the artists want a bigger cut of the pie, then why don't they just make it happen themselves?



Costs! Making it "happen themselves" means paying up front for the cost of making an album, you can only go to Apple/Spotify/Tidal/Youtube with a completed product, you can't go to them and ask for money to make or complete your product, and the record labels are far more risk averse than they once were because there are far fewer big paydays and those big paydays are nowhere near as big as they used to be. The big record labels, the ones with the resources to hire the best and give them the time to do their best, are today more interested in their back catalogues and corporate deals with the aggregators/hosters than they are in developing talent. In other words, to get the up front costs of producing an album from likely the only people who can afford it (the labels) means you already have to have a finished product/s and an established serious social media following/fan base to even stand a chance.



old tech said:


> While it is true that most of the mainstream artists make less money from record sales than in the past, there are now a lot more opportunities for those on the periphery.



Not only is that pie itself much smaller, more corporate entities have their fingers in it and more artists are after a slice of it! Until you get to the point of someone (a label for example) fronting the costs, you're effectively limited to pretty much home-made products, without the knowledge, experience and quality of resources of the good commercial studios and, you're in competition with 50 million other Youtubers who've got $60 worth of great audio software, a mic and are convinced they're the next Beyonce or One Direction. The sad truth is that many of the truly great bands/artists and albums of the past could not be made today.

Sure, the top artists still make many tens of millions a year but unlike in the past, only a small fraction of it actually comes from recordings, they make it from touring, celebrity endorsements, etc. If even the very top artists generally don't make much from recordings, what commercial incentive is there for investing a lot of time/money in making them? This hasn't just hit the headline acts and corporations, it's had an even more serious impact further down the food chain. There used to be professional "session musicians" and other artists who made a modest but entirely livable income from the recording industry. They've all but gone now, typically their income has dropped to just hundreds or a couple of thousand a year and they have to survive by other means. I would agree that there's a lot more opportunity for those on the periphery but it's only more opportunity to make a few grand, I'd say there's a lot less opportunity to make a living from it! There's only so many times you can slice a pie. The professional recording industry is on it's knees, it's likely that it won't really be an industry in the future, just a limited number of small boutique establishments, a whole bunch of home/semi-pro project studios and maybe a handful of commercial studios (or none at all). It looks at the moment that the best albums which will ever be made, have already been made! This is why cries from the likes of TheSonicTruth are being blanked or met with rolling eyes.

Lastly (and not directed specifically to old tech), the impression some audiophiles seem to have is that the mastering engineer's job is effectively to take good mixes and turn them into trash and that the rest of the recording industry is twiddling it's thumbs, waiting for the good old days to return. Obviously that's nonsense, the industry is trying everything it can think of and mastering engineers want to make the mixes better, not worse! But at the end of the day, consumers don't want to pay more than a cent or so to listen to a track and for convenience, they want to pay that cent to a streaming or download service which ends up with a hundredth or a thousandth of a cent being paid to the artists. I understand that and its still better than the nothing at all they got from the likes of Napster but if consumers are only willing to pay peanuts then they're only going to get products made by monkeys. It's going to take a while for the full impact of consumers' collective decision to be felt, as it takes time for the studios and talent to try different options until the only one left is giving up and there's little/no incentive for new studios or talent to take their place.

G


----------



## OddE

Fun (?) anecdote - while reading up on this thread, it suddenly dawned on me that I was listening to Motorhead's "Everything louder than everything else!"


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## sonitus mirus

OddE said:


> Fun (?) anecdote - while reading up on this thread, it suddenly dawned on me that I was listening to Motorhead's "Everything louder than everything else!"



Back in the days before CDs, I remember a few albums that had "warnings" letting everyone know that the record was meant to be played loud.  Mountain's "Climbing!" was one I recall.  They were right, "Mississippi Queen" sounded great cranked up to 11.


----------



## castleofargh

Cutestudio said:


> Can you point to the research that indicates only a tiny minority care about the sound quality?
> Additionally can you qualify that research with the method of asking, because when I play a well mastered track to anyone young they suddenly realise what they were missing and become interested in better quality. Is this one reason why the record industry suppresses decent masters today


ah the ideal quest for quality. that thing making every human on the planet achieve excellence in every domains and then push it all to 11 for the love of all that is good... oh sorry no, I was thinking about that Disney story.
the armchair approach sure is good looking, and when asked in the street, most people will answer that they prefer the better stuff so long as they have a vague even erroneous idea of what "better" means. talk is cheap. people like good food, just ask them. then look at the success of fast food chains and artificial crap in stores. could it be that people are full of good intentions only so long as it comes with absolutely zero drawback compared to doing the easy stuff?
people want the truth, just ask them and they'll tell you. but if the truth means spending more than 5 minutes on google and actually testing stuff and learning stuff to do it properly, all of a sudden we go from pretty much all humanity, to a handful of actually curious people. and the rest will satisfy itself with oversimplified ideas and empty claims.
it doesn't take a genius to notice how people are full of crap when it comes to doing the right thing. oh they believe in what they say for sure, they just won't do any of it when it's a bother.

people prefer good dynamic and good sound at a philosophical level, sure. when put under test without telling them anything, you would already see why the loudness war became a thing in the first place. because it works. and if dynamic was really that highly significant variable that you want it to be, then how comes people buy all the compressed stuff and love it? at some point you have to come down of your high horse and look at the real world where not everybody is you.

and from a practical approach, I agree with @RRod that a lot of music is "consumed" on the go nowadays. making highly dynamic stuff simply not the most appropriate. I've been a classical music lover for as long as I can remember, but I have almost no classical music on my DAPs and wouldn't listen to classical in a car(ok maybe if I had one of those luxurious cars where you can hear yourself breath while on the highway, then I would). mostly because of those damn quiet passages where you can't hear crap. the world is not that good vs evil painting you're trying to force on us. I imagine a lot of albums are discussed and vetted by entire rooms of people, and as several of those people are going to make more money based on how much they sell, if it was as simple as not ruining the tracks with added compression, I got a feeling the loudness war wouldn't have lasted more than a week.
it's easy to look from where we sit and decide that we know better, I even think that way about politics despite knowing absolutely nothing about politic. we're just wired that way, we have opinions on everything and we assume that what we know is always enough to make the right choice. sometimes things are just more complicated and one guy keeps asking for more cowbell. 
if you make me general of the world, I'm fine passing a law to put a taser on the balls of anybody clipping a digital track. I'll sign it on day one of my legit coup. but anything beyond that, anything is fair game and part of the artistic process. not good vs evil, just I like vs I don't.


----------



## 71 dB

****  Loudness war has nothing to do with 24 bit vs 16 bit debate.  ****​


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## bigshot

^ this. And it has nothing to do with the use of compression in mixes either.



amirm said:


> What is a "normal living room?"



Music is engineered to suit the listening conditions of the audience for a particular type of music. That's why a Wagner opera is engineered different than Justin Bieber. If you'd like to know what a normal living room looks and sounds like, go out in the world and look at some. Any person in the entertainment business knows that if you don't know your audience, you risk losing them. We pay attention to how the stuff is going to be played by our target audience.

I really don't understand the people who say that sound quality is terrible across the board now. I hear great sounding recordings all the time. My shelves are packed with current releases that sound better than anything that came before. There are a bunch of lousy recordings too. It's a crap shoot and it's always been a crap shoot. For every remaster that sounds like crap, there's one that sounds much better than previous releases. The other day I got Fleetwood Mac's Tusk in 5.1. I remember when the album came out I bought it on vinyl. It sounded as thick as sludge. The CD didn't sound much better. This new 5.1 remaster sounds fantastic, like a totally different album. You roll the dice and you find out where they land. If you are concerned about sound quality, read reviews before you buy. If a remaster sounds horrible, return it to the store for a refund. I buy at Amazon. I have done that in the past. They refund your money with no questions.

I wouldn't expect k-pop to be mixed and mastered the same as jazz. There's a functionality that they're aiming for in their choices. I've always been puzzled by audiophiles that listen to lousy music. What's the point of buying a $50 gold plated CD of a crappy 70s dinosaur rock album? Why do people buy pristine sounding Mannheim Steamroller albums? No one actually *likes* Mannheim Steamroller! It's extremely irritating music.

The goal is a good mix that presents good music well. Those two go hand in hand. The best mixes are the ones that achieve a synthesis with the style of the music and the functionality of how it will be listened to. An approach that works for one type of music doesn't necessarily suit another. But complaining about mainstream pop not having dynamics is like ordering lake trout and complaining that it tastes like fish.


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## amirm (Nov 19, 2017)

gregorio said:


> 1. Let's say your $120 DAC plays audibly perfectly, noise and artefacts below -120dB. What about another DAC which also performs audibly perfectly, with artefacts below -120dB but costs $2k instead of $120? It's pretty much guaranteed that $2k DAC is being sold as having audibly superior performance but unless you're happy to pay an extra $1,880 just for a prettier case, how is that not an equipment scam?


What something retails for is orthogonal to this technical discussion.  There are CD players that costs thousands of dollars.  How is limiting audio formats to CD rates going to help that?  There will always be luxury products in audio market.  That is no basis to use as any kind of argument.

Your point initially was that support for high resolution audio costs extra for consumers.  That just isn't true.  The silicon chip used to perform the DAC functionality is going to have support for higher sample rates and bit depths and the cost of that is in single digit dollars or at best low tens of dollars.  In other words, the cost of one or two CDs.  This is not some exotic technology that we are talking about.

Heck, here are the formats that my laptop supports natively with its built-in audio:






It goes up to 192Khz and 24 bits (green numbers).  And it comes right out of the headphone jack!   And it is darn clean too:






Heck even our cell phones come with excellent audio support these days.  Here is my Samsung S8+:






As I said to you earlier, *you are fighting an obsolete war. *  High resolution audio support comes for "free" these days just like power locks come on cars.  The latter used to be a luxury but is no more.  You won't advocate that we got back to manual locks as to save money.  You should not advocate the same here either.


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## amirm

gregorio said:


> 1. Of course we can, that's what we do, that's what mastering is for!


Mastering is for a format.  As I have said over and over again, we no longer have a format called "CD" that requires you to crunch things for that.  We don't need you to reduce the bit depth to 16 and sample rate to 44.1.  Give us the choice of what is upstream about that.  Why would insist on doing that conversion for us???  There will always be the option of 16/44.1 for many people who want that.  Just provide the true "masters" that are above that specific rate.


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## amirm

gregorio said:


> 3. No, we haven't been exposed to much higher SPLs than we imagine. Taking your own figures from the previous sentence, we've been exposed to peak levels 5-10dB higher than our average levels. So, if our average levels are say 80dB, which would be high in a near 0dB noise floor environment, the peak SPLs are at say 90dBSPL, about thirty times lower than 16bit can provide!


What do you mean "say?"  Do you have research data on this or we are going to continue to go by made up numbers like that?

When we discuss formats, it should be for all people and all situations.  You can't keep talking numbers out of a hat and throw it out that way.  The research I provided does exactly what I say: surveying real situations like sitting in a concert hall in a real seat.

Really, so many posts and not one, not one bit of research or reference data is presented.  It is all gut feeling, argumentative posts and guesses.  What does anyone learn from this?  Don't you believe in backing what you say with some kind of published reference?  Am I the only one that believes in that in a "science" forum?  People come here to read someone's specific opinion over and over again and not learn something from published research?

This is why I don't like your multi-part questionnaire.   Challenges after challenges is put forward without adding any data to the conversation.  Just creating work for everyone, and making threads long as to lose interest of anyone but the few people fighting.


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## gregorio

amirm said:


> There are CD players that costs thousands of dollars. ...
> Your point initially was that support for high resolution audio costs extra for consumers. That just isn't true.



So you're saying that I could have traded in my CD player and got support for hi-res at no cost? That I can go to HD Tracks and pay the same for the 16bit version as for the 192/24 version or that if I want support for 786/32 I can upgrade my 24/96 player for free? Maybe you can tell me where you're getting all this free hi-res support? If I had a functioning 16/44.1/48 player with artefacts below -120dB why would I need to go and buy a new player?

G


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## gregorio

amirm said:


> 1. We don't need you to reduce the bit depth to 16 and sample rate to 44.1. Give us the choice of what is upstream about that. Why would insist on doing that conversion for us???
> 2. The research I provided does exactly what I say: surveying real situations like sitting in a concert hall in a real seat.



1. Are you playing dumb or what? I have to convert it, I cannot give you 44.1/64 and prior to that, I couldn't give you 44.1/56!!!
2. A real situation of no audience and no musicians. Maybe you get your entertainment from being the only person in a concert hall and listening to the noise floor. Call me crazy but I prefer to go to a concert hall to experience a bunch of musicians performing a concert. You got research for that?

G


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## amirm

Strangelove424 said:


> Bigshot is right, you are disingenuous. I never mentioned the word average, I always said peak or max. This entire discussion, and the paper you’ve scanned snippets from, are all in context to peak level. You know full well that I was referring to peak levels. And you know full well the UCSD link I gave you referred to peak levels too. I hate to backtrack in conversation - redefining the obvious - because people play dumb.
> 
> correction: upon checking my post, I did use the word average once, but was referring to the average peak level from multiple performances measured, not average SPL. You are twisting my use of the term as it applies to the data.


You provided a link to back up what you were saying: http://musicweb.ucsd.edu/~trsmyth/level175/Example_SPL_Levels.html





As you see there, there is no mention if the SPL value is peak or average.  Unless the word "peak" is stated, these numbers are all averages.  

Importantly as I have highlighted, this is not any kind of proper research or original data either.  It is just a set of numbers lifted from Dan Levitin's book.  If we go there, we see this:





As you see, this is just like countless such lists you find online and in books.  It is just rule of the thumb, average values.  In no way can these be compared to a technical paper that I quoted to you where its aim was to find actual peak values.  For 99% of the research out there for other purposes average numbers are fine and that is what is used.  It just happens that we are talking about a topic that requires us knowing the actual peaks because we have to store those numbers in our audio files.  We can't just store "average numbers" as that won't be the music.

So instead of getting upset, do you have a reference that disputed the JAES paper data that I quoted that says otherwise?  If not, that is that.  No reason to get upset or worse yet, throw those personal accusations at me.


----------



## amirm

gregorio said:


> 1. Are you playing dumb or what? I have to convert it, I cannot give you 44.1/64 and prior to that, I couldn't give you 44.1/56!!!


First of all, you can leave the sample rate alone.  Why are you converting that to 44.1? 

Second, what is the format of the stereo mix you are getting to "master?"


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## gregorio

amirm said:


> [1] First of all, you can leave the sample rate alone.  Why are you converting that to 44.1?
> [2] Second, what is the format of the stereo mix you are getting to "master?"



1. Why am I converting 44.1 to 44.1?
2. Typically 4 stems truncated to 24 or 32 bit float.

G


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## RRod

castleofargh said:


> and from a practical approach, I agree with @RRod that a lot of music is "consumed" on the go nowadays. making highly dynamic stuff simply not the most appropriate. I've been a classical music lover for as long as I can remember, but I have almost no classical music on my DAPs and wouldn't listen to classical in a car(ok maybe if I had one of those luxurious cars where you can hear yourself breath while on the highway, then I would). mostly because of those damn quiet passages where you can't hear ****. the world is not that good vs evil painting you're trying to force on us. I imagine a lot of albums are discussed and vetted by entire rooms of people, and as several of those people are going to make more money based on how much they sell, if it was as simple as not ruining the tracks with added compression, I got a feeling the loudness war wouldn't have lasted more than a week.
> it's easy to look from where we sit and decide that we know better, I even think that way about politics despite knowing absolutely nothing about politic. we're just wired that way, we have opinions on everything and we assume that what we know is always enough to make the right choice. sometimes things are just more complicated and one guy keeps asking for more cowbell.
> if you make me general of the world, I'm fine passing a law to put a taser on the balls of anybody clipping a digital track. I'll sign it on day one of my legit coup. but anything beyond that, anything is fair game and part of the artistic process. not good vs evil, just I like vs I don't.



I listen to tons of classical music in the car… using a compressor! "But but" some people on this thread might say, "you are destroying the die-namiks!" Well, yes, I am, because sometimes I still have interest in the melodies, harmonies, and rhythms of the music, despite being in an automobile. Once at work I gladly hook the closed cans up to an amp and destroy some of my remaining cilia. Unfortunately, many people aren't quite keen enough on audio to mess around with compression on their own, but they want it to exist, because they too have to commute to jobs they hate! So that means baking some compression in, especially to material that your average Josephine will listen to on the R line. Now add to this the fact that much music isn't dynamic to begin with, and you can see there hasn't been a huge push against sausage music. Companies spitting out classical music have to contend with the fact that their material has a readily-available real-world baseline. People actually attend concerts of Mahler's symphonies, played by real instruments that make real SPL levels. You can only push things so far before even the deafest classical fan will know you are selling the dynamics short. What the heck is the baseline for Taylor Swift?


----------



## amirm

gregorio said:


> 2. Typically 4 stems truncated to 24 or 32 bit float.


There you go.  You are handed 24/32 bits, give us the same thing.  We don't get any benefit out of you getting that down to 16 bits.

As to sample rate being 44.1 in what you get, then I like to understand why that was converted down.

We need a new integrated chain for high fidelity music that gives us data at the resolution and sample rate that was used to capture the instruments.

BTW, you use higher bit depth in your audio workstation software to allow manipulation of PCM data without loss.  We need the same thing here as we also post process the data in the form of Room EQ, additional digital filtering, etc.  We don't need to keep dithering down, convert back up, dither down, etc.


----------



## RRod

amirm said:


> There you go.  You are handed 24/32 bits, give us the same thing.  We don't get any benefit out of you getting that down to 16 bits.
> 
> As to sample rate being 44.1 in what you get, then I like to understand why that was converted down.
> 
> ...



So he'll charge us less then for not putting in the extra time to covert the file? Oh wait, no, that's not how the music industry is handling hi-res, is it? Also, the processing chain argument isn't quite on: you are perfectly free to take a 16bit input file and never take it back down to 16bits again if the rest of your chain operates at >= 24bits.


----------



## gregorio

amirm said:


> 1. Really, so many posts and not one, not one bit of research or reference data is presented. It is all gut feeling, argumentative posts and guesses. What does anyone learn from this? Don't you believe in backing what you say with some kind of published reference? Am I the only one that believes in that in a "science" forum? People come here to read someone's specific opinion over and over again and not learn something from published research?
> 2. There you go. You are handed 24/32 bits, give us the same thing.
> 3. As to sample rate being 44.1 in what you get, then I like to understand why that was converted down.
> 4. We need a new integrated chain for high fidelity music that gives us data at the resolution and sample rate that was used to capture the instruments.
> 5. BTW, you use higher bit depth in your audio workstation software to allow manipulation of PCM data without loss. We need the same thing here as we also post process the data in the form of Room EQ, additional digital filtering, etc.



1. Let me get this straight. You can talk about symphony orchestra recordings and quote noise floors of empty concert halls because there's published scientific data but I can't talk about noise floors and dynamic ranges of a hall with a symphony orchestra actually in it because there's no published scientific data? Where does that leave us, completely misleading unreal noise figures or just not talking about anything related to the recording and reproduction of classical music?
2. What, 4 stems of 24bits, how you going to play that? What I am handed is not what is distributed, if it were, why would I be hired in the first place? Don't you want mixed or mastered music, you just want a hard disk full of RAW takes?
3. It wasn't converted down, that's often what I'm given, although more commonly 48kHz, in which case I leave it at 48kHz unless my client requests 44.1kHz. I have had clients give me 44.1kHz and want 96kHz, only twice though.
4. Which instruments? The softsynths operating at 64bit, the 16bit samples from the sampler or the 8bits of the bass guitar in a 24bit container? And what sample rate do you want? The 44.1kHz of the lead vox, the 96kHz which the reverb unit resamples at or the 192kHz which my compressor oversamples to? Or maybe you want the 22mHz/5bit (or whatever) my ADC digitizes at before decimation? Much of this hi-res thing is nonsense, don't you get that, it's marketing, there's not even an agreement of what hi-res is! In practice and for some years, sample rates are all over the place, some processors up/down sample and we don't even know about it, other times we are given the option but on any particular popular genre song there may have been a dozen or more conversions to various sample rates and that's before it even gets to the mastering engineer. It was like the ridiculous SACD vs PCM marketing thing, SACDs had to be converted to PCM to be edited, mixed and mastered and then converted back to DSD again, same with vinyl which starting in the mid 70's was digitized before being cut, digital is bad, analogue is great ... suckers!
5. That's a good argument for a DACs which applies EQ having a 64bit processing environment, just like our DAWs. That's got nothing to do with the distribution format though, just as our 64bit DAWs have nothing to do with the 16, 24 or 32 bit files we load into them.

G


----------



## pinnahertz

Cutestudio said:


> Can you point to the research that indicates only a tiny minority care about the sound quality?


No problem: commercial radio.  Listener statistics are taken continuously, ratings published periodically.  The stations with the biggest numbers don't correlate at all with the best quality, and never have.


Cutestudio said:


> Additionally can you qualify that research with the method of asking, because when I play a well mastered track to anyone young they suddenly realise what they were missing and become interested in better quality. Is this one reason why the record industry suppresses decent masters today?


Yup.  Historically there have been a couple of methods, including hand-written diaries distributed to a sampling of 1000 or so listeners who tracked their listening habits.  Today's high-tech solution is the People Meter, a device that you keep with you that listens to whatever you do.  Stations signals are encoded so they are easily recognized and counted, data is uploaded to a server when the device is placed in its charging dock.  The data can include internet streams, satellite radio, and TV.  Listeners have a wide selection of choices of free entertainment.  The collected data includes time of listening, time spent listening, and station choice.  Ratings and demographics drive advertising rates. 

A couple of notes on broadcast audio.  Even though audio quality has never been well correlated with listenership, stations obsess on individualizing their on-air processing.  Think of broadcast processing as the loudness war gone nuts.  It's been an armed conflict for over 60 years, and todays DSP processing units are brutal, though perhaps slightly more gentle than the most brutal analog processors.  Each station is different, but stations with competing formats tend to be generally similar.  Classical stations, what there are left of them, are the lightest processed, but still aggressive compared to anything on record or CD.  AM radio is severely band-limited, and with a high noise floor, and deliberate clipping is commonplace.  Yet with the worst quality on air, AM stations still win ratings. 

Sorry to report, audio quality is worse than secondary for most listeners of radio.  And that translates to recorded material as well.  It's all about the content.  People listen to what they want to hear, and audio quality is tertiary behind convenience.


Cutestudio said:


> Your statement is also illogical because you reframe the argument as a choice between cost and quality, whereas in reality not over-compressing and clipping costs exactly the same as mastering something properly.


Actually, today cost is reducing to almost a non-factor either way.  The largest listenership of any on-line music source is YouTube, for free.


Cutestudio said:


> I.e. you appear to confuse your role as a creative one where we are supposed to appreciate the mangling, where in reality is should be an invisible one that allows us to hear the musicians.


We've all heard this before: blame the engineer.  You don't understand the industry, though.  If someone comes to me asking for maximum loudness to the destruction of quality, and I refuse to do it and attempt to educate him, he'll leave and go somewhere else.  The only thing accomplished is my loss of business.  That means I can't pay bills and feed my family.  What would you do?


Cutestudio said:


> I think the patronising attitude of the Mangling Engineers and their industry moved into the realm of fraud a long time ago, in fact I've returned quite a few CDs now because of poor quality mastering, people buying CDs or downloads have a reasonable expectation of quality and only a tiny - but growing - minority have any idea how bad the problem is. Too much of the HiFi industry is complicit in enabling this, CD players for example are designed very carefully to avoid overloading on clips - plug in a pro-audio DAC however and the overload light and clips are obvious, a much more honest system.


Actually, you returning CDs is the absolute best thing you can do - vote with your wallet!  I support your efforts, though they are futile in reality.  I take strong exception to your labelling engineers as fraudulent.  Nothing they or the industry does is criminal, and loudness processing is wrongful in some eyes, great in others. 


Cutestudio said:


> These overload lights are the same ones the witless cretins at final mastering treat as a badge of honor rather than the stark warnings they are. It's simple audio engineering 101: don't overload and don't clip, which appears to have been forgotten in the Great Dumbing Down of the past 30 years.


Well, if you actually took Simple Audio Engineering 101 you'd know that audio can be fully loudness processed, and clipped without any overload lights ever lighting at all.  It's easy, and done all the time.  You're focussing on the wrong thing.


Cutestudio said:


> HiFi is a shadow of what it was in the 1970s and 1980s, an industry full of charlatans pushing snake oil digital wires, in 40 years it's _still_ using unbalanced interconnects and the lack of anything decent to play has turned many people away.


Focussing on the wrong thing again.  What does using unbalanced interconnects have to do with sound quality?  Are you aware that the process of creating a differential drive signal to drive a balanced interconnect and then receiving it with an instrumentation amplifier configuration more than doubles the active devices, noise and distortion sources?  How is that better?  Balanced and unbalanced interconnects are topologies that each have benefit _when properly applied._


Cutestudio said:


> The 'still sounds good as no one cared enough to mangle it' still applied to many DVDs, often a DVD sound track will be far hight quality simply because the sound engineers are just doing their job, not the 'special' creative types who's aim in life to to reduce all sound to a solid brick shaped wall of noise. I visited a consumer electronics show a few years back and it was revealing that the DVD soundtrack to King Kong was the best sounding thing there, only comparable to some old vinyl, the 'HiFi' was just an unpleasant wall of noise that needed turning down - it seems this was thanks to people like you and Bigshot being creative and knowing 'best'.


Well, equating today's "best" with "old vinyl"...that would be a highly filtered personal opinion not reflecting the reality of anything important in audio...like reproducing the signal as true to what was heard in the control room as possible.  Vinyl doesn't do that, never has.


Cutestudio said:


> Amirm's idea is the smartest, sell the 24bit un-mangled versions for a premium. But the record industry is terminally stupid and has constantly strived to avoid this. When SACD came out the internet was slow enough that a simple DVD density disk of 96k/24 audio would have been worth buying on silver discs, leaving the mangled 16bit for MP3 which people were downloading anyway even over dial-up. They missed that money making opportunity so here we are 30+ years later with mangled 16bit silver discs that people simply bypass in favour of mangled MP3s because no one can tell the difference after the mangling.
> It's still revealing that music is on sale at Apple and Amazon. Where is the RIAA or the record companies? Go to Virgin Records today (http://www.virginrecords.com/releases/) and you'll see they've just about worked out how to make a slow clunky webpage, but you can't even buy their product direct. Doh.
> 
> With the Greg and Bigshot attitude I've been watching the 'tiny minority' of audiophiles become a self fulfilling prophecy as the 'experts' dance around the steaming pile of 'product' to justify the production of mangled 16bit for all. For what reason is a mystery besides the overarching need to be 'right'. It's not a good enough reason.



Yet another case of misfocused attention.  We have, and have had several easy channels of distribution of higher than red-book audio for quite some time.  I doubt you'd find many record producers or artists that would permit an "un-mangled version" of their work to get out into the wild.  They consider mangling, as you call it, is part of their art.  You'd be asking the artist to release an incomplete early version.  Not going to happen.  If you want unmangled audio, there are a few sources catering to a small but loyal market.  Give Mark Waldrep's AIX records a shot.  Good clean unmangled audio in abundance.  And not expensive, really.  He's all about provenance, so you're getting the real deal.  I have a selection of his stuff, and he does a good job.  Unfortunately, it's not because it's 24/96, he just does a really good job!  And unfortunately, not really much main-stream material.  Still good, and entertaining.  

You have a minority opinion, and while I share your feelings on the music product being overly processed, I don't think the solution lies in griping about it.  Through history of recorded music there have been records ranging from excellent to awful, including the revered vinyl days.  I have some very good vinyl, and a lot of just average stuff.  It was easy to get great sound out of a CD, thus vinyl died.  And yes, I know, it's had a "resurgence", but still is insignificant in the total picture.  It's not better, it's a different total experience.  But bad vinyl was more the order of the day before the CD, now we have bit-perfect copies of masters.  You don't like what's on the master...well, that's a problem, but not an easy one to solve.  Keep voting with your wallet, and get about 50 million of your best friends to do the same, and you might start to steer the ship.


----------



## amirm

RRod said:


> So he'll charge us less then for not putting in the extra time to covert the file? Oh wait, no, that's not how the music industry is handling hi-res, is it? A.


He is not charging us either way.  He is being paid by the record label to do his "mastering" no matter what as they need that kind of down conversion for lossy encoding and CD.  I made no argument that something would be cheaper if left at 24 bits.

What the music industry needs to do is take high-resolution music as a serious offering, not left to random mastering engineer to do whatever.  They need to put quality standards in place and create that as a new deliverable.  After all, in a market where they are struggling to make any profit, they have found an audience that is willing to pay more for high-resolution music.  In the case of smaller independent labels this is happening, sans the hygiene that is lacking at times.  Major record labels need to do the same thing.  Once there, Gregorio wouldn't be making decisions for us to dumb down the dynamic range and bandwidth of audio for us.  He will do as the label tells him to do.  He can even charge more for "mastering for high-res" as another format.  Why he is complaining in that context is beyond me.  Mastering engineers should celebrate plurality of multiple formats!

And oh, the best thing they can do is disallow loudness compression for high-res music.


> lso, the processing chain argument isn't quite on: you are perfectly free to take a 16bit input file and never take it back down to 16bits again if the rest of your chain operates at >= 24bits


No, dither is already added to it in the process of conversion to 16 bits, reducing its dynamic range.  I can't put that genie back in the bottle.  As you say, since my output is 24 bit converter, no one should attempt to convert it down to 16 before I even get my hands on it.  If what you say is what should happen, then he should be getting 16 bits just the same and not: "Typically 4 stems truncated to 24 or 32 bit float."  What is good for him, is good for us.  We perform signal processing on the content and for the same reason he and his software maker need more bit depth, we do too.


----------



## amirm

gregorio said:


> 1. Let me get this straight. You can talk about symphony orchestra recordings and quote noise floors of empty concert halls because there's published scientific data but I can't talk about noise floors and dynamic ranges of a hall with a symphony orchestra actually in it because there's no published scientific data? Where does that leave us, completely misleading unreal noise figures or just not talking about anything related to the recording and reproduction of classical music?


Where it leaves you is to go and buy the paper, read through it, instead of constantly asking me questions about it.  How much research have you read on this topic anyway?  I have yet to see anything quoted from literature from any of you.  There is no learning in that.  Just arguing and arguing and arguing. 

And no, this is not "my" data.  It is data from a luminary in the industry, who is ex president of AES, and AES Fellow and has written multiple detailed papers for the Journal of AES.  Because you haven't read the research you are not understanding the methodology.  Until such time that you actually spend some time and a bit of money to learn about this topic from top signal processing and psychoacoustics experts in our industry, you are going to be in the dark on the core thesis and logic.



> 2. What, 4 stems of 24bits, how you going to play that? What I am handed is not what is distributed, if it were, why would I be hired in the first place? Don't you want mixed or mastered music, you just want a hard disk full of RAW takes?


Now we get to the meat of the discussion: _*"why would I be hired in the first place?"  *_Finally we hear why someone is in the business of creating music for us, is up in arms about this.  They think their job is to take studio masters and reduce it to size and fidelity for us.  Well, with advent of high-resolution music we no longer need that.  You can't keep creating folklore like what started this thread as a form of job security.  The ship has sailed and labels have started to release high resolution music.  You need to figure out what your role is in this new world order.  It is not our job or place to think of your job security especially when you are advocating us getting less rather than more.

Same kind of nonsense is going on with MQA with mastering engineers up in arm that someone may grab "masters" behind their back and release it.  And then we get subjected to bunch more nonsense arguments as to why that is not good for us.  What they really not saying is that it is not good for them.  Sometimes life is not fair.  Typewriters are gone and we shed no tears for them.  So don't ask me how new formats may obviate your role.  That ship has sailed and you either get on board and figure out what you will do in the new world order or find something else to do.


----------



## Cutestudio

RRod said:


> I listen to tons of classical music in the car… using a compressor!
> 
> Unfortunately, many people aren't quite keen enough on audio to mess around with compression on their own, but they want it to exist, because they too have to commute to jobs they hate!
> 
> So that means baking some compression in, especially to material that your average Josephine will listen to



I'm not following why the need for a compressor in the car stereo translates to the need to bake it into the source material. Why ruin it for everyone when not ruining it is easier? Pioneer etc. could just do it in the stereo for the car. In fact the last cheap Pioneer car radio I bought had a special setting to improve the sound of compressed music so it's hardly going to be difficult for them to add compression, they just replace their complex algorithm with a very simple one. 

The problem with baking the damage into the material is that most compression is _extremely_ difficult to reverse, so by selling the same clipped and over compressed rubbish to everyone guarantees that it won't quite sound right on _anything_, whereas the best qualified to compress music for a car is a company like Pioneer, not some talentless hack at the record company ruining the output for everyone.

I note that the narrative is sticking with compression and ignoring _clipping_. Tracks like that in the Elephunk album from the Black Eyed Peas and many others are compressed sure, but also contain many perfectly flat clips, some over 200 samples long. Clipping is common is digital music, some clips are quite rough and there's a gray area between some severe brick wall limiting and clipping but the effect is the same and is a dishonest way of selling a product that is marketed with the pretence of CD quality.

Greg, Pizza, Biggs, fascinating replies but you entirely miss the point, sorry. I thought you would. Why are you arguing for LoFi on a HiFi forum exactly, is there something we should know?


----------



## amirm

Cutestudio said:


> Greg, Pizza, Biggs, fascinating replies but you entirely miss the point, sorry. I thought you would. Why are you arguing for LoFi on a HiFi forum exactly, is there something we should know?


Well said.  That is what really puzzles me.  So much time and energy spent to say, "you guys should get less fidelity."  Other than making a name on forums to champion some cause, I have yet to see anyone articulate any reason why.  To add insult to injury we are then handed a bunch of folklore masquerading as audio science for the reasons why.

Sitting here, I have analog master tapes that sound so much better than CD versions that have been butchered to death in the process of "mastering."  These guys are not listening to the fact that we like better fidelity content.  I was reading Brian Lucey interview on how he was doing his mastering workflow: http://fairhedon.com/2017/11/05/an-interview-with-mastering-engineer-brian-lucey/

"*Preferred work flow*_*: * I have a simple chain with *3 analog pieces in between DA and AD.*  I work back to each single from the overview by skipping around with the cursor to sections of each track for a few seconds.  It’s a *mostly analog chain for the EQ, MS, compression and limiting. * Also at times I use a linear phase EQ in Sequoia 12, or the internal DeEss."_

So he takes the digital mix he is handed, converts it to analog, messes with it in that domain multiple times, and then converts it back to digital!  Now we would be lucky to have 12 bits of signal to noise ratio and anything above 16 is a long lost dream.

I downloaded one of his 24-bit/44.1 Khz "high res" recordings and it is an abomination.  It is loudness compressed to hell and it simply is not listenable.  I will be doing a video on this soon but it is just shameful what they are doing to music and then go brag about how good they are.  They are good for meeting the needs of labels and talent that wants, loud, loud.  Not those of us who want good sounding music.

I mean look at LP listeners.  So many of them enjoy better sound because these offenses are not occurring there because LP can't handle it.  So in an odd turn of events, the limitations of that format allows them to enjoy better sounding music much like my tape masters.

As I said, my wish remains that the industry takes this market seriously and provides quality releases for us.  I am happy to pay $5 or even $10 for that.  Heck, I paid $250 per tape to get that music!!! :eek:


----------



## Darren G

I enjoy your posts too amirm.  You have done a good job of explaining your position.

You know it's just that when I read posts like someone linked from Rob Watts, about the benefits 200-300db of noise level (and why stop there, why not 500db?), I choke. When you do the (basic) math, and realize what is being said/written, well hopefully you can see why some are dubious when it gets into the 120db range.   The numbers really don't convey the extremes being talked about.


----------



## Strangelove424

TheSonicTruth said:


> Thanks for the links! But I know who they're from and I often find he talks a good game but doesn't always push hard enough.  He seems to think DR8 on the meter plugins is enough for commercial releases, when thirty-forty years ago DR12-15 was the norm for chart stuff! He and I agree respectfully to disagree.
> 
> I agee with you about RAM, and own that CD.  Charlie Puth's "Attention"(off his forthcoming self-produced mind you 'Voice Notes' album!) and some of Bruno's recent material also has my 'attention', not just in how good it sounds but the style and arrangement.  It actually takes a b r e a t h! lol



“Pushing hard enough” is something a person with an agenda does. Don’t confuse activism with mastering. And don’t dismiss someone’s point of view because they’re not extreme enough in their expression for your taste. If something sounds good at a DR of 8, it would be insincere of him to say otherwise. I think music with a DR of 8 can still sound good, as RAM on CD demonstrates. That’s the entire point of the second article, and a bit of the first. Atleast read the first one about Daft Punk since you are a listener of theirs too. The quality of the sound is not just about dynamic range.


----------



## Strangelove424

amirm said:


> You provided a link to back up what you were saying: http://musicweb.ucsd.edu/~trsmyth/level175/Example_SPL_Levels.html
> 
> 
> 
> ...



SPL charts like this are all in reference to peak SPL since average SPLs can vary widely based on the variety of quietest sounds. Do you include the parts of the singer talking to the audience when  averaging a rock concert? It's a grey area, so people refer directly to peak. Look closely at the 105db number for classical music, its says during “loud passages”. That means peak. I guarantee you the number for a WHO concert at 130db is peak as well. Can you imagine what the peak would be if that were an average?!? If you weren’t so set on proving yourself right, you might learn some things by looking at those values. Remember you argued with people before, saying that 20db is the ambient for a normal room, not a recording studio? Look at the numbers again. An office with the door closed and computer turned off is 35db, a recording studio is 20db. You are so focused on being right you ignore piles of facts in front of you, dozens of contradictory SPL readings, and even twist context.

BTW, that study is filled with holes, I can’t understand why you treat it as the gold standard besides the selfish reason that its the only document in existence backing you up. For one thing, they made their very own SPL meter! Did they document how they designed it or calibrated it? Nope! This study lacks transparency, and its data is an outlier. It’s simply not valid, and has made very little attempt to establish its own validity.


----------



## 71 dB

Everyone here wishes loudness war never existed, but less "loud" music makes less money for the record companies and money runs the world, not fidelity. Fortunately a lot of music (less commercial stuff) does not suffer from loudness war and the average sound quality of for example classical music releases today is very high. I also have to say the "DR6" pop of today sounds amazingly dynamic considering how brickwalled it is dynamically. Music producers really know how to use such a compressed audio signals!

I am not banning high res audio. I am just telling my opinion how I think it doesn't offer anything compared to 16/44.1 in _consumer_ audio, because in consumer audio the practical dynamic range is totally covered by 16 bits. Nobody listens to concertos for jackhammers and mosquitos and even that is more or less possible (just barely maybe) with 16 bits and shaped dither!


----------



## RRod

Cutestudio said:


> I'm not following why the need for a compressor in the car stereo translates to the need to bake it into the source material. Why ruin it for everyone when not ruining it is easier? Pioneer etc. could just do it in the stereo for the car. In fact the last cheap Pioneer car radio I bought had a special setting to improve the sound of compressed music so it's hardly going to be difficult for them to add compression, they just replace their complex algorithm with a very simple one.
> 
> The problem with baking the damage into the material is that most compression is _extremely_ difficult to reverse, so by selling the same clipped and over compressed rubbish to everyone guarantees that it won't quite sound right on _anything_, whereas the best qualified to compress music for a car is a company like Pioneer, not some talentless hack at the record company ruining the output for everyone.
> 
> ...



You're skipping the part about normal people not wanting to play with a compressor, but that suits your narrative. I am well aware of what clipping is, and indeed such audibly *bad* things are what finally break even the staunchest lover of loudness.

You can cry 'shill' all you want; we all know what it means. The waveforms I posted earlier are indicative of the average level of fidelity I listen to regularly. Sorry if your music doesn't have it. Peace Cutey.


----------



## JaeYoon

Darren G said:


> I enjoy your posts too amirm.  You have done a good job of explaining your position.
> 
> You know it's just that when I read posts like someone linked from Rob Watts, about the benefits 200-300db of noise level (and why stop there, why not 500db?), I choke. When you do the (basic) math, and realize what is being said/written, well hopefully you can see why some are dubious when it gets into the 120db range.   The numbers really don't convey the extremes being talked about.


Hey! if we can do 1,100 DB, we could make a giant black hole!


----------



## RRod

amirm said:


> He is not charging us either way.  He is being paid by the record label to do his "mastering" no matter what as they need that kind of down conversion for lossy encoding and CD.  I made no argument that something would be cheaper if left at 24 bits.
> 
> What the music industry needs to do is take high-resolution music as a serious offering, not left to random mastering engineer to do whatever.  They need to put quality standards in place and create that as a new deliverable.  After all, in a market where they are struggling to make any profit, they have found an audience that is willing to pay more for high-resolution music.  In the case of smaller independent labels this is happening, sans the hygiene that is lacking at times.  Major record labels need to do the same thing.  Once there, Gregorio wouldn't be making decisions for us to dumb down the dynamic range and bandwidth of audio for us.  He will do as the label tells him to do.  He can even charge more for "mastering for high-res" as another format.  Why he is complaining in that context is beyond me.  Mastering engineers should celebrate plurality of multiple formats!
> 
> ...



I'm perfectly fine with a world of various masters, I just don't think one should cost more than the other. Or do you disagree?


----------



## gregorio (Nov 19, 2017)

amirm said:


> [1] And no, this is not "my" data. It is data from a luminary in the industry, who is ex president of AES, and AES Fellow and has written multiple detailed papers for the Journal of AES.
> [2] How much research have you read on this topic anyway?
> [2a] I have yet to see anything quoted from literature from any of you.
> [3] Until such time that you actually spend some time and a bit of money to learn about this topic from top signal processing and psychoacoustics experts in our industry, you are going to be in the dark on the core thesis and logic.
> ...



1. Oh, I didn't realise it was from a luminary ex-president of the AES. In that case, the noise floor must be that of an empty hall with no musicians or audience in it, I take it all back! Going to be a bit tricky to get any dynamic range with no musicians or people in the concert hall but what do I know, I'm not an ex-president of the AES! BTW, do you want that empty hall recording delivered in 16, 24, 32 or 64bit? 768kHz? How about with a TYPICAL anti-imaging filter with a 500Hz transition band and rectangular dither?
2. A fair amount. Also, I once went to a concert hall with some audience and musicians in it, doesn't that count as research? Obviously not for you or an ex-president of the AES but not even for us poor normal people! Question: If I'd switched the HVAC off, thrown everyone out of the concert hall, so it was just me, then would it have counted as research or would I have to become an ex-president of the AES first?
2a. Ah, someone else who can't read.
3. Actually I have but until such time that you actually spend some time contextualising that core thesis and logic you're going to remain in the dark about what is "real life" and what actually happens in the recording industry! First hint (AGAIN), the noise floor of an empty hall with no musicians in it, is not what happens in a real life performance or recording, not even a real life performance of 4:33!!
4. I've obvious got my job wrong, I always thought my job was to take a final mix and make a studio master, not take a studio master and trash it, silly me, I better ring my clients and let them know. The size and fidelity of my masters are dictated by my clients; the artists, producers and/or record labels. Jeez amirm don't you even know the very basics of music creation roles? Maybe an ex-president of the AES needs to publish a paper for you?
5. You no longer need what, mixes, studio masters? Did the ex-president of the AES tell you that?
6. The reason I need that bit depth is because I'm often running 200+ audio channels, 50 or so EQ's, 10 or 20 compressors, 8 or more convolution reverbs and probably 100 plugins in total. That's what you're doing is it? That's why you need what I've got is it? And you need 20dB of headroom do you?

Honestly amirm, I started out having quite a bit of respect for you but now you're painting yourself into the corner of the nuttiest, most ignorant of audiophile extremists!



Cutestudio said:


> Why are you arguing for LoFi on a HiFi forum exactly



Err, for the same reason that you never learned to read.

G


----------



## JaeYoon

I feel like @gregorio is getting fired upon for being an audio engineer/producer/mixer.

I mean, if people put up a slogan that recording industry is doing horrible things to recording, it seems that someone has to be put to blame, and audio engineers are unfortunately the goat chosen to be in full display of the Anti-Loudness War and 16 bit chopped up mangledness.
We'd have to think about what the client/industry and why they want that, which is to sell music the way they wanted it done.

I know it's easy get a mob up against Audio engineers, but before we hang someone, we need to think about this before we spit fire on them.


----------



## Cutestudio

JaeYoon said:


> I feel like @gregorio is getting fired upon for being an audio engineer/producer/mixer.



An interesting view, but I rather think he's simply meeting 'consumer resistance' to his continued assertions that we are an ignorant minority not worth listening to and should be grateful that we pay for over-compressed and clipped music because it's stylistic/artistic, creative and compression is necessary so how can we complain when there is too much?

However it's a very positive thread for me because now I totally understand the attitudes of the people in the industry and the reason that sound quality is so dire, will continue to be dire, and we should really expect no improvement. We give our views, they not only refuse to listen but try to lecture us about how wrong we are, so the situation continues as before. Such is life.

This continues the fun of searching for those early CD masters and classic vinyl which is in many ways a lot more satisfying than buying rubbish with real money. By many orders of magnitude in fact. Which is what I've been doing for years now, my yearly CD bill fell sharply in around 2004 when I realised the CD waveforms were all mangled, and has all been diverted to 2nd hand eBay buys around the world and market stalls where for £2 one can find hidden gold, mastered by people smart enough not to clip the signal too badly so I can call it CD Quality.

The future of music is not with record companies anyway, The Market adapts and the internet enables, artists will start to realise that Simply Red had a point all those years ago: why give record companies all the power and money when you can just sell music yourself? There are various routes opening up that allow artists to keep far more control over the process and maintain far closer links with their customers. The smart bands will then not renew their contracts and start selling direct, as I think is already happening: middle man(gler) not required


----------



## gregorio

Cutestudio said:


> now I totally understand the attitudes of the people in the industry



You couldn't even accurately read (let alone understand) what one person in the industry posted and from that you "totally understand the attitudes of the people in the industry". Ignorance is bliss and delusion seems to be a lot of fun too! Anyway now that you've apparently got whatever it was you trolled this thread for, there's no need for you to post to it any more is there?

G


----------



## TheSonicTruth

RRod said:


> If I ask a random person at work "what do you do for music?", the answer is likely to be "I just wear whatever earbuds come with my iPhone, but I only really listen on the subway in to work, and I usually just stream whatever is on." Music that requires dedicated, low-background listening isn't the norm any more. Are you saying you have a real reason to deny this as reality?



It is the reality.  That's why I keep original issue CDs of most classic artists in my collection.  I'm just fighting to keep furher damage from being inflicted to that pre-1990 legacy material trying to make it as loud as more recent releases.  It's consumers and the artists that are destroying the sound of recorded music, not the engineers - both by demanding loudness and by buying it!  smh


----------



## TheSonicTruth (Nov 19, 2017)

bigshot said:


> ^ this. And it has nothing to do with the use of compression in mixes either.
> 
> 
> 
> ...




"_And it has nothing to do with the use of 
compression in mixes either._"

Above a certain amount of compression, it DOES.  Below that, compression helps tame a shaky vocal, or gives a drum track some 'beef'.  Above that, dynamic compression, especially when combined with peak limiting and makeup gain to boost what's left back up to a tenth below full-scale, becomes a volume control, whether you care to admit it or not.  And the relatively highly dynamic '70s era pop and rock CD transfers of 25-30 years ago sound just fine to me in my car via earbuds on a plane without the help of a squasher or buzzcut, whether or not you care to believe it.



"_...CD of a crappy 70s dinosaur rock album?_"

Born during or after Reagan, eh bigshot?  smFh....


----------



## reginalb

amirm said:


> ...It is data from a luminary in the industry, who is ex president of AES, and AES Fellow and has written multiple detailed papers for the Journal of AES.  Because you haven't read the research you are not understanding the methodology...



As I'm not an audio engineer or researcher, I've read the posts in your discussion without replying. But I'm sick of your constant appeals to authority. We get it, you think you're smarter than everyone else, but you just come across to me as incredibly arrogant. I've read your articles, and they aren't remotely compelling. I get it, I should spend a bunch of money and read all the AES papers they're based on, but there really isn't a point in doing that for me, so other people providing summaries of the information is interesting to me, I do enjoy it. But I wish you could present your arguments with a bit less arrogance and logical fallacy. 

In another thread, you talked about how bad you feel about someone who runs some hi-res audio thingy because of people like bigshot and gregorio. Perhaps you should feel bad for people that get ripped off by snake oil peddlers pushing audio equipment and formats without any practical benefit. Sure, I could train myself to hear artifacts in music that's been compressed in a lossy format, and I could crank up quiet sections of my 16-bit music to hear the noise floor then complain about it. Or I could care about music. You know, the actual thing that these formats and equipment are supposed to be reproducing? 

I get that this is the sound science section, and I actually find your posts interesting, but I really wish you could drop the appeals to authority and act in a bit less arrogant manner.


----------



## TheSonicTruth (Nov 19, 2017)

amirm said:


> First of all, you can leave the sample rate alone.  Why are you converting that to 44.1?
> 
> Second, what is the format of the stereo mix you are getting to "master?"



Why are they converting to 16/44.1 for delivery?  Because those specs more than adequately take care of what humans can hear.

If you can hear a difference between the same identical recording mastered at 24/192 vs 16/44.1, then either 1. you have golden ears or 2. something was *done* to the higher res version to make it sound different.  

And no, I do not believe audio has to be compressed and or limited to 'fit' into 16/44.1, even though I actually have had engineers(on GearSlutz and rec.audio.pro) tell me that that final peak limiting to give the waveform that 'flat top' look is considered 'proper mastering', as opposed to leaving peaks ragged(intact!) to within 2dB of 0dBfs - as seen on waveform representations of most CD audio before the mid-'90s.


----------



## old tech

amirm said:


> Mastering is for a format.  As I have said over and over again, we no longer have a format called "CD" that requires you to crunch things for that.  We don't need you to reduce the bit depth to 16 and sample rate to 44.1.  Give us the choice of what is upstream about that.  Why would insist on doing that conversion for us???  There will always be the option of 16/44.1 for many people who want that.  Just provide the true "masters" that are above that specific rate.


Amirm, some of your comments appear to conflate bit depth/sample rates with mastering choices.  The two are not necessarily related.

I doesn’t appear that big shot, gregorio or others are championing 16/44 over higher res formats but rather the obvious, that the higher res formats in themselves do not offer any improvement in consumer sound quality given that 16/44 already reproduces music within the bounds of human hearing.  Any extension of bit depth or frequency range is moot.

The real issue is the mastering choices with regards to new music or remasters of the old.  It is pointless blaming the engineers for the loudness wars as they need to serve their clients to make a living.  It is their clients you should be directing your frustrations.

While no doubt there are some engineers who actually like producing loud, brickwalled music, the majority don’t.  Many do not sit there idle with regard to this issue either.  For example, Ian Sheppard, the guy you mocked for producing DR8 music, started the loudness day movement.  Bob Ludwig, tries to educate his (artist) clients, usually by providing more than one version of a mastering so they can hear what a more dynamic production sounds like, but if they still choose the brickwalled version what can he do?  Go on strike?  It is better to try an influence the paying clients from within than from the outside.

As far as formats go, no one is suggesting that we should all stick with 16/44 even though that is all we need for accurate reproduction.  I do needle drops of LPs and have done so for decades.  Most of them are 16/44 because I used to make CDs from them.  These days I leave them at 24/96 because I rarely listen to CDs, so it is more around practicality rather than a misplaced belief that 24/96 is going to sound better.

The bottom line is that we should be agitating for better mastering regardless of whether the recording is going to be on CD (remember that CDs still account for the majority of new physical format sales, easily eclipsing LPs and hi res downloads) or another format.  Focusing on whether the file is 16/44 or hi res is irrelevant.


----------



## TheSonicTruth (Nov 19, 2017)

old tech said:


> Amirm, some of your comments appear to conflate bit depth/sample rates with mastering choices.  The two are not necessarily related.
> 
> I doesn’t appear that big shot, gregorio or others are championing 16/44 over higher res formats but rather the obvious, that the higher res formats in themselves do not offer any improvement in consumer sound quality given that 16/44 already reproduces music within the bounds of human hearing.  Any extension of bit depth or frequency range is moot.
> 
> ...




"_Focusing on whether the file is 16/44 
or hi res is irrelevant._"

I mentioned 16/44 vs high res only in the context that the mastering matters more, sonically.


----------



## old tech

Cutestudio said:


> Can you point to the research that indicates only a tiny minority care about the sound quality?
> Additionally can you qualify that research with the method of asking, because when I play a well mastered track to anyone young they suddenly realise what they were missing and become interested in better quality. Is this one reason why the record industry suppresses decent masters today?


Research is unnecessary, the market has spoken.

If interest in audiophile quality masterings were significant there would be a significant market catering to that demand.  After all, music is a competitive business and how many labels wouldn’t participate in a market of high demand for such masterings, particularly if they can increase their margins. It just wouldn't make any commercial sense to "suppress decent masters".

But if you want evidence, consider the state of the market for audiophile releases.  Most specialist labels do not sell large quantities to justify expanding their production.  Indeed many go broke or just don’t see it as worth continuing, such as DCC or MFSL.

Of course some of those specialist labels still exist, eg MFSL, Speaker’s Corner etc but they are bit players with no impact on mass consumers.  I’m willing to bet that if you sampled 1000 music consumers you’d be lucky to find one that has heard of any of these labels.

The music industry is in retreat but the reasons for its decline is far more complicated than “bad” mastering, indeed, “bad” mastering is not even a significant factor given the low demand for audiophile music.  A much bigger issue is that the entertainment market is far broader than what it was back in heyday when choices were limited to the home stereo, TV with limited channels and cinema.  Now we have video on demand, internet, cable TV and so on.

While two channel hi fi is becoming a thing of the past, the part that surprises me a bit is why the music industry hasn’t evolved into high fidelity, multi-channel music formats with video – like blu ray video.  Sure there is some of that available but very little product offerings to have broad appeal.


----------



## TheSonicTruth (Nov 19, 2017)

old tech said:


> Research is unnecessary, the market has spoken.
> 
> If interest in audiophile quality masterings were significant there would be a significant market catering to that demand.  After all, music is a competitive business and how many labels wouldn’t participate in a market of high demand for such masterings, particularly if they can increase their margins. It just wouldn't make any commercial sense to "suppress decent masters".
> 
> ...




Why would I want 5/6/7.1 channels of dynamically bereft, brickwall limited _owl vomit_?


"_ A much bigger issue is that the 
entertainment market is far broader 
than what it was back in heyday when 
choices were limited to the home stereo, 
TV with limited channels and cinema. 
Now we have video on demand, 
internet, cable TV and so on._"

Yes, more sources for information and entertainment, but of compromised production quality.  Sad!


----------



## bigshot (Nov 19, 2017)

TheSonicTruth said:


> Below that, compression helps tame a shaky vocal, or gives a drum track some 'beef'.



OK! There we go. That wasn't so painful was it? By the way, it isn't to tame shaky vocals. It's to make the enunciation of the lyrics clear.



TheSonicTruth said:


> Born during or after Reagan, eh bigshot?  smFh....



I'm 58 years old. My tastes in music have improved and broadened since I was 17.


----------



## TheSonicTruth

bigshot said:


> OK! There we go. That wasn't so painful was it? By the way, it isn't to tame shaky vocals. It's to make the enunciation of the lyrics clear.
> 
> 
> 
> I'm 58 years old. My tastes in music have improved and broadened since I was 17.



SMH!  I'm only 47.  Why are so many folks who are older than me making such millennial statements??  Guess I'm really conservative culturally: 'Put on some Who and Van Halen and put a six-pack in the fridge for when Mike and Jim and their wives come over!'  Politically I'm a different story, LOL not for discussion here.


----------



## pinnahertz

71 dB said:


> Everyone here wishes loudness war never existed, *but less "loud" music makes less money for the record companies* and money runs the world, not fidelity.


I emphatically disagree with that!  Within the commercial pop music business there is a distorted perception that loud music is more desirable, but it has never been proven to sell more music. What has been proven is that people listen to the music they want to with astounding general disregard for quality.  In these days where higher quality than ever is possible, that's the real puzzler.


----------



## pinnahertz (Nov 20, 2017)

Cutestudio said:


> I'm not following why the need for a compressor in the car stereo translates to the need to bake it into the source material. Why ruin it for everyone when not ruining it is easier? Pioneer etc. could just do it in the stereo for the car.


Well, when it comes to classical music, it's NOT baked in, so it has to happen in the car or car-destiined chain somewhere.  Processing classical music for high-noise environment listening is darn difficult, and quite aggressive, so it absolutely would be rejected in that market for any other listening.  So far I have yet to see a car stereo with actual dynamics processing in it.  Link me up to one if you know about it.


Cutestudio said:


> In fact the last cheap Pioneer car radio I bought had a special setting to improve the sound of compressed music so it's hardly going to be difficult for them to add compression, they just replace their complex algorithm with a very simple one.


There's been an attempt by several manufacturers to improve the sound of compressed music with some sort of algorithm-based DSP.  However, that's not dynamic compression they're talking about, that's the kind of lossy compression found in .mp3 files, etc.  Technically, "compression" is the wrong term for that, it's bit-rate reduction with a lossy codec.  But compression is easy and stuck, hence your confusion.


Cutestudio said:


> The problem with baking the damage into the material is that most compression is _extremely_ difficult to reverse, so by selling the same clipped and over compressed rubbish to everyone guarantees that it won't quite sound right on _anything_, whereas the best qualified to compress music for a car is a company like Pioneer, not some talentless hack at the record company ruining the output for everyone.


I get the idea here, and sort of agree.  You can't reverse loudness processing for many reasons, mostly you don't actually know how it's been applied, and if applied on a track level, you can't isolate that track ever again to reprocess it.  Yes, baking it in is a one-way street.  However, what you get out of anything is what the producer wanted you to get, which may or may not be garbage depending on the music and processing.  I don't agree that it was a talentless hack that applied it.  If you had ever attempted loudness processing yourself you'd know how actually difficult it is to do well.  And a certainly don't believe Pioneer is a company capable of properly processing any type of music for all listening environments.  There are companies that have specialized in audio processing for 40 years that I might turn to for that, perhaps with the end result being a chip or something, but otherwise processing is not simple to do well without side effects.


Cutestudio said:


> I note that the narrative is sticking with compression and ignoring _clipping_. Tracks like that in the Elephunk album from the Black Eyed Peas and many others are compressed sure, but also contain many perfectly flat clips, some over 200 samples long. Clipping is common is digital music, some clips are quite rough and there's a gray area between some severe brick wall limiting and clipping but the effect is the same and is a dishonest way of selling a product that is marketed with the pretence of CD quality.


200 samples at 44.kHz is about 4ms long, which frankly, depending on the degree, is just barely audible as being clipped.  You don't have a handle on how clipping can be used, and when it becomes audible or not.  It's time vs frequency vs degree, and not as easy as just peeping at a waveform, blowing it up and saying in horror "See! It's Clipped!".  There actually can be inaudible clipping.  But I'm not advocating clipping, just explaining it. 

I don't see what's dishonest about selling your art.  The CD represents that art flawlessly.  The artistic choices were made, and the CD reproduces it.  If you don't like the choices, return the CD and ask for a refund. 

I agree the loudness war ruins a lot of good audio quality, but you keep pointing fingers at things you don't understand.  I get you're mad at the audio world!  You just don't understand what you're mad at.


Cutestudio said:


> Greg, Pizza, Biggs, fascinating replies but you entirely miss the point, sorry. I thought you would. Why are you arguing for LoFi on a HiFi forum exactly, is there something we should know?


I'm not arguing in favor of LoFi, I'm trying to help you understand what's going on, and why.  You should do yourself a favor and stop zooming in on waveforms, and just listen.  I'm sure you'll still hear loudness war processing, but you're not doing yourself any favors by magnifying the inaudible. 
For the record, I oppose the kind of processing done on pop music today.  I feel it's unnecessary and gets in the way of musical enjoyment.  If I had my choice I'd prefer a whole lot less.  But I do also understand two important things: 1. Artistic purpose - and processing and EQ and a whole lot of other functions come under that heading.  You can't take away the palette!  And 2. The real war is the result of a general distortion of values and reality, but it's not the engineers making the call, it's above them - the guys producing the music and writing the check. You've labeled the engineers as frauds, dishonest, talentless hacks...and on and on.  You just have no idea what a real fraudulent dishonest talentless hack would do.  I promise, the results would be far worse.


----------



## ev13wt

pinnahertz said:


> I emphatically disagree with that!  Within the commercial pop music business there is a distorted perception that loud music is more desirable, but it has never been proven to sell more music. What has been proven is that people listen to the music they want to with astounding general disregard for quality.  In these days where higher quality than ever is possible, that's the real puzzler.



Yup. I listenened to the top 25 yesterday. At least half is recorded really bad. Doesn't become better when you turn it up either. After that session, I thought my system is broken. Then I played "Morph the cat" at 96dB and the world was ok again.


----------



## gregorio

TheSonicTruth said:


> Above a certain amount of compression, it DOES.  Below that, compression helps tame a shaky vocal, or gives a drum track some 'beef'.  Above that, dynamic compression, especially when combined with peak limiting and makeup gain to boost what's left back up to a tenth below full-scale, becomes a volume control, whether you care to admit it or not.



No, that's way over-simplified to the point of being almost totally nonsense. There is no "certain amount of compression", it ALL depends on the track, the genre and other variables. Furthermore, you cannot compare the rock/pop of the '70's with the music today in simple terms of compression, because the music today is NOT 70's pop/rock! Modern R&B, Drum & Bass, EDM, Techno, Hardcore, Trance, Hip Hop, Death Metal, etc., these and others are the music of this millenium and even though pop/rock of the '70's required significant compression to "beef up the drum track" and "tame a shaky vocal", most modern genres require massive amounts of compression to create the drum like and other sounds which actually define these genres/sub-genres. In other words, you can't reduce the use of very heavy compression to the levels of the 1970's without taking away what makes today's genres, those genres in the first place! Therefore, some of the statements by cutestudio and this one of amirm's "_And oh, the best thing they can do is disallow loudness compression for high-res music._" are effectively the statements of Nazi's! They are effectively saying that most/all of the modern popular music genres should be disallowed. Now I don't think they are actually Nazi's, I just think they are so ignorant of how modern popular music genres are created that they don't realise they're effectively being Nazis.

Current professional music engineers and artists are obviously creating music for today, not for the 1970's, so of course they are going to look at the likes of cutestudio and amirm as some sort of nazi dinosaurs, stuck in a 1970's - 1980's time warp. That's OK though, it's always been that way, even Elvis in the 1950's was banned and slammed by nazi dinosaurs, so we almost take it as a compliment! I get that you don't like the modern popular genres and that's fine but by the same token, today's artists have the right to express themselves and their generation's culture however they want and no one is forcing you to buy it. If we take say Adele's "25", it's ranked as "bad" by the DR database, with a score of just DR 6. If it were down to me and if I were mixing or mastering it for my own critical listening pleasure, I'd have applied a fair bit less compression but it's won all the top awards (Grammys, Brits, etc.), garnered great critical acclaim and broken all kinds of historical sales records, which is even more impressive at a time when the album is supposed to be dead!

G


----------



## Cutestudio

old tech said:


> Research is unnecessary, the market has spoken.
> 
> If interest in audiophile quality masterings were significant there would be a significant market catering to that demand.



There is a chicken and egg situation here that you do not account for:
For instance if the average man on the street wants a genuine 96/24khz - or even an unclipped 44.1/16  copy of Sarah McLachlan's 'Solace' CD because they wanted to hear a version without 200 sample long clips in it, what does he do?
He can complain on a forum like this - (this IS a market force BTW known as 'demand') - and be told that he's wrong. 
he can ask the record company and they'll tell him he's a tiny minority and there is no market for non mangled music.

But you are right about the market having spoken, but it has spoken for higher quality _already_:
Instead of complaining to the manglers what he does is go to Apple - not a record company but a large successful tech who _does_ have a mangle free  product known as 'Mastered For iTunes' which will hopefully have the album on that. Then he'll buy that version, helping to turn Apple in to a $1Tn cap company in the process.

Apple has basically found that mangling is unpopular and proven the manglers on here that they are wrong, music sounds better un-mangled and there is a market for it. 
For the ostrich mimics - the manglers here's an explanation of Mastered For iTiunes:

https://www.justmastering.com/article-masteredforitunes.php
https://www.npr.org/sections/therecord/2012/02/24/147379760/what-mastered-for-itunes-really-means

So in a way _we have already won_ the argument against clipped, over-compressed 16bit, _the market has already spoken and been answered_ and one of the world's most successful companies has given us a route to bypass the manglers.
It would have been nice to have the option of buying a CD of Merchantable Quality rather than the fraudulent mangled mess of today. But given the huge amount of hostility to the idea on - _a HiFi forum_ - I can see this will never happen. I hoe the record companies enjoy going bust, their decision to drop quality entirely has already resulting in their output have to shift to 12 year old singers and the whining mating calls of lovesick teenagers with auto-tune but I don't see that as a sustainable business model. 

I still find it amusing how people on a HiFi forum campaign, twist, invent, obfuscate and tirelessly argue for LoFi, it's amazing. 
Perhaps the average mumsnet contributor is enjoying better music on their iPhone that we are, even with our £400 USB cables sending the mangled mess into our poor DACS... LOL


----------



## ev13wt

TheSonicTruth said:


> They start making more money when they reject loudness as the sole criterion, in favor of fidelity, and start to sell more albums and downloads.
> 
> Nothing wrong with their art, just the way it is packaged/mastered.



Nope, they won't make more money - because only a tiny niche market cares about sound quality.

The typical consumer and mixdown are for era buds, BT speakers and Cellphones. Not much choice there for dynamic masterng, really. Users would not hear half the songs...


----------



## Cutestudio

gregorio said:


> No, that's way over-simplified to the point of being almost totally nonsense. There is no "certain amount of compression", it ALL depends on the track, the genre and other variables. Furthermore, you cannot compare the rock/pop of the '70's with the music today in simple terms of compression, because the music today is NOT 70's pop/rock!


Actually we can compare what we like, over-compression and clipping is more related to the year rather than style. The lie of 'it's a different era' is undone by looking at remasters, even Floyd's Final Cut and ABBA have been mangled in all the re-issues. That's the very SAME recording, the original is good quality, the more recent re-mangles are all mangles. So dies the argument about style.



gregorio said:


> today's artists have the right to express themselves and their generation's culture however they want and no one is forcing you to buy it.


The 'no one's forcing you to buy it' argument. What's up with that? What if we want to buy it but without clipping and over-compression? Isn't our money any good?



gregorio said:


> If we take say Adele's "25", it's ranked as "bad" by the DR database, with a score of just DR 6. If it were down to me and if I were mixing or mastering it for my own critical listening pleasure, I'd have applied a fair bit less compression but it's won all the top awards (Grammys, Brits, etc.), garnered great critical acclaim and broken all kinds of historical sales records, which is even more impressive at a time when the album is supposed to be dead!
> G


Yes, lets take Adele.
In the top quote you incorrectly claim that we can't compare stuff (despite the litany of shoddy remangles that are now infamous for mangling and shunned by the HiFi community), so lets compare Adele to Adele. This is easy as she conveniently numbers her albums, take a look at the overall mangling of Adele's 19, Adele's 21 and Adele's 25. You can clearly see that the mangling on 19 is of it's time and that 21 is mangled worse. Then look at 25, its mangled even worse than 21.

So here you have provided a classic example of a single artist and genre who dates and tracks the trend to mangle records worse with each passing year.

Adele's 19: rather badly mangled
Adele's 21: mangled worse than 19
Adele's 25: mangled worse than 21

The mangling of music also coincides with the loss of interest in HiFi by the general public. 
Once HiFi was something to aspire to and everyone I knew wanted a decent HiFi. It was big business and in every shopping center and mall.
Now no one cares because of one very simple fact: modern music sounds just as bad on HiFi as it does in the car, radio and MP3 player. 
That's why the music industry has become a circus of 'The voice', BGT etc, because the mangling has removed all of the dynamic interest from the music.  Without dynamics and with generic instruments and autotune what you are left with is a product of a) a whining teen, b) soppy PC lyrics, c) a youtube video. 
So why would any millennial  want a HiFi today? - there's no use for it.

So yes, Adele: Good example of the rot setting in, a good boast of 'not a single peak left unclipped'.


----------



## ev13wt

RRod said:


> I listen to tons of classical music in the car… using a compressor!



Sorry bud, but we need to take away your audiophile card


----------



## ev13wt

amirm said:


> What the music industry needs to do is take high-resolution music as a serious offering, not left to random mastering engineer to do whatever.  They need to put quality standards in place and create that as a new deliverable.



"Most music sound just as crappy on CD as it would in the DAW". CD IS high resolution, but 95% of songs don't care. 

Maybe we need a standard of "no clipping = high quality" as a start. It won't fix anything, but it sure is a start. I do think that (listening to the stuff my "home studio beat makers make) many of the original samples they use are already cipped. I listenend to the top 25 yesterday - 23 songs where clipped or simply terribly recorded!

Had to listen to morph the cat on 44.1/16 after all that high res CRAP.   Ahhhhh - much better sound quality than all those MQA high res song of Adele...


----------



## 71 dB

Cutestudio said:


> So in a way _we have already won_ the argument against clipped, over-compressed 16bit, _the market has already spoken and been answered_ and one of the world's most successful companies has given us a route to bypass the manglers.
> It would have been nice to have the option of buying a CD of Merchantable Quality rather than the fraudulent mangled mess of today. But given the huge amount of hostility to the idea on - _a HiFi forum_ - I can see this will never happen. I hoe the record companies enjoy going bust, their decision to drop quality entirely has already resulting in their output have to shift to 12 year old singers and the whining mating calls of lovesick teenagers with auto-tune but I don't see that as a sustainable business model.
> 
> I still find it amusing how people on a HiFi forum campaign, twist, invent, obfuscate and tirelessly argue for LoFi, it's amazing.
> Perhaps the average mumsnet contributor is enjoying better music on their iPhone that we are, even with our £400 USB cables sending the mangled mess into our poor DACS... LOL



Why do you think anyone here is for "mangled music"? The number of bits has nothing to do with loudness war and. CDs could have 100 bits and an obscenely high sampling rate and we'd still have DR6 pop. Maybe your favorite music is always mangled on CD, but mine isn't, because only a small part of my favorite music is commercial. The more they play the music on radio, the more it's mangled. If they don't play it on radio (most music is never played on radio) it's probably not mangled. It just means you have to find the music yourself, because you don't hear it on radio. There's tons of un-mangled CDs out there and if you don't find them… …well that's your problem, not my problem.

Also, this is not about _how_ you get your music. This about the number of bits and sampling rates. 16/44.1 is not a synonym of CD and CD is not a synonym of mangling. When I say 16/44.1 is enough, it means 16/44.1 is enough and nothing else. It definitely doesn't mean I support loudness war.


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## TheSonicTruth (Nov 20, 2017)

Cutestudio said:


> Actually we can compare what we like, over-compression and clipping is more related to the year rather than style. The lie of 'it's a different era' is undone by looking at remasters, even Floyd's Final Cut and ABBA have been mangled in all the re-issues. That's the very SAME recording, the original is good quality, the more recent re-mangles are all mangles. So dies the argument about style.
> 
> 
> The 'no one's forcing you to buy it' argument. What's up with that? What if we want to buy it but without clipping and over-compression? Isn't our money any good?
> ...




Cutestudio:  See how both gregorio and bigshot condemn the truth as "nonsense"?  Sounds just like what's coming out of the White House recently!  There's only way to handle the likes of gregorio and bigshot:  It begins with the letter 'I' and contains six letters.  I'm going to do it, and I suggest you, and others fighting the good fight, also take heed.  I'm tired of everything I say on here being called nonsense or ridiculous when those people have no way of justifying something, and will no longer waste any keystrokes on these individuals.  I did it with Ian Shepherd, and will do it with others.  Let them talk to walls on here.


----------



## 71 dB

Cutestudio said:


> The 'no one's forcing you to buy it' argument. What's up with that? What if we want to buy it but without clipping and over-compression?* Isn't our money any good?*



Hate to break it to you Cutestudio, but our money is good only _occationally_. I'd like to upgrade my 1998-2005 J-horror movies DVD collection to Blu-ray, but only a few of these movies are out and some of them with Italian or Polish subtitles or whatever, not even English! Finnish subtitles would be nice, but unrealistic. So, yeah, my money isn't that good in this respect. I'd also like to have Good Wife tv show (one of the most underrated shows of the last decade imo) on Blu-ray, but they offer DVD only! They can keep them, it's not year 2001 anymore. We live in capitalism. If there's not enough demand there is not enough demand. If you want to get rid of your money then I recommend to do some discovering to find new things to spend your money on. Life will be pain in the ass if you assume everything revolves around you. It doesn't. It revolves around the 100 richest people in the world, people who own almost everything on this planet. You can thank capitalism for that.


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## TheSonicTruth (Nov 20, 2017)

ev13wt said:


> Sorry bud, but we need to take away your audiophile card




Rrod is onto something, only I've been suggesting it for the last five years:  Put a barebones, basic compression codec in mobile audio gear(car, boat hi-fi, and as an app available across phone and player platforms.  Better yet, a loudness codec as Apple has already done with SoundCheck for iTunes.

Right next to the 'bass, treble, balance' controls in the menus.  Everything would play back at a predetermined RMS or loudness algorithm, and there would be no justification for artists & labels to demand the 'squashing' or limiting of anything any more!


----------



## ev13wt

TheSonicTruth said:


> Rrod is onto something, only I've been suggesting it for the last five years:  Put a barebones, basic compression codec in mobile audio gear(car, boat hi-fi, and as an app available across phone and player platforms.  Better yet, a loudness codec as Apple has already done with SoundCheck for iTunes.
> 
> Right next to the 'bass, treble, balance' controls in the menus.  Everything would play back at a predetermined RMS or loudness algorithm, and there would be no justification for artists & labels to demand the 'squashing' or limiting of anything any more!




Sure sounds like you are on to something there!

I use a compressor for watching movies - I hate the dynamic range they put in there (vocals vs. music/effects) First you are straining to hear them talk over the crunch of chips, after the next scene the neighbors are whatsapping me (its just the bass)...


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## TheSonicTruth (Nov 20, 2017)

ev13wt said:


> Sure sounds like you are on to something there!
> 
> I use a compressor for watching movies - I hate the dynamic range they put in there (vocals vs. music/effects) First you are straining to hear them talk over the crunch of chips, after the next scene the neighbors are whatsapping me (its just the bass)...



I believe many DVD and HD players(which includes Blu Ray) have a DRC(dynamic range control) or Night Mode setting that also accomplishes the same thing.

Either way, it would free up makers of music and movies to concentrate just on doing that, instead of engaging in - and stuttering to justify - a pointless loudness spat to the bottom of the sonic toiletbowl.  Compress to he|| in the sanctity of your own home or vehicle, while someone else listens to it in all its dynamic glory.  Digital Democracy at last!


----------



## 71 dB

ev13wt said:


> I use a compressor for watching movies - I hate the dynamic range they put in there (vocals vs. music/effects) First you are straining to hear them talk over the crunch of chips, after the next scene the neighbors are whatsapping me (its just the bass)...



I watch movies with headphones (Lt/Rt downmix + wide crossfeed gives pretty nice result). My neighbors are not whatsapping me…


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## gregorio (Nov 20, 2017)

Cutestudio said:


> [1] But you are right about the market having spoken, but it has spoken for higher quality _already_: Apple has basically found that mangling is unpopular and proven the manglers on here that they are wrong, music sounds better un-mangled and there is a market for it. For the ostrich mimics - the manglers here's an explanation of Mastered For iTiunes:



For those who are able to read (which apparently excludes cutestudio): Mastered for iTunes does not affect the amount of compression/mangling, nor has Apple "found that mangling is unpopular". What cutestudio states here is *utter nonsense which he's just completely made up*! Here's what Apple actually states, in it's official specifications for Mastered for iTunes:

"_*Your decision about the volume and loudness of your tracks is a technical and creative choice. You might decide to take the listener on a dynamic journey through an album as a complete work, raising and lowering the volume level across the sequence of tracks to increase the music’s emotional impact. Alternately, you might pursue the loudest possible signal at all times. Whatever you decide—exquisitely overdriven and loud, or exquisitely nuanced and tasteful—we will be sure to encode it and reproduce it accurately.*_"

A quick look at the dynamic range database will also confirm many MFiT albums with ridiculously low DR scores. Here's one example of a MFiT album with a DR of 04 and one of it's track's is just DR 02!



Cutestudio said:


> [1] Actually we can compare what we like.
> [2] The mangling of music also coincides with the loss of interest in HiFi by the general public.
> [2a] What if we want to buy it but without clipping and over-compression? Isn't our money any good?



1. You know, you might be right. I just compared Scarlatti's Lute Sonatas with Motorhead's "Ace of Spades" and I think I can hear what you're talking about!  I want "Ace of Spades" with it's heavy compression and distortion, call me crazy but I absolutely DO NOT want Ace of Spades to sound like a nice gentle Lute Sonata! The ridiculous thing about your lack of comprehension is that it was us engineers who first started complaining about the loudness war, long before consumers even knew what it was and now you're blaming and insulting the very people who've been fighting against it the longest, what is wrong with you?
2. Duh, what have we been telling you and you've been arguing against? Who's going to make more expensive content which the public have lost interest in and don't want to buy?
2a. You know very well your money is not "any good" because there's nowhere near enough of it to make it worth anyone's time!



TheSonicTruth said:


> I'm going to do it, and I suggest you, and others fighting the good fight, also take heed.



Good luck banning all modern pop music genres. Do you remember the nazi dinosaurs who banned Elvis? ... No, neither does anyone else! They were too dumb to realize that acting like nazi dinosaurs actually promoted Elvis. You want to fight the loudness war, then great, join the club but enough with the ignorant nazi dinosaur crap, you're not helping, you're hindering and you're making yourself look like another complete fool in the process!

G


----------



## RRod

TheSonicTruth said:


> Rrod is onto something, only I've been suggesting it for the last five years:  Put a barebones, basic compression codec in mobile audio gear(car, boat hi-fi, and as an app available across phone and player platforms.  Better yet, a loudness codec as Apple has already done with SoundCheck for iTunes.
> 
> Right next to the 'bass, treble, balance' controls in the menus.  Everything would play back at a predetermined RMS or loudness algorithm, and there would be no justification for artists & labels to demand the 'squashing' or limiting of anything any more!



I'm sure some marketing dept. can come up with a shnazzy icon that indicates "if you want to actually hear your music in the train station, press this". I posted in another thread that a focus just on normalizing loudness doesn't really get to the issue of your listening environment supporting only a certain dynamic range.

The issue of course is that it's hard to decide on a default way to do compression writ large. I'm sure a smart enough person could come up with a generally useful loudness-based compression scheme, though. Still, there's bound to be material where a generalized algorithm doesn't quite do it right, which leads to a whole set of issues for artists, engineers, and listeners.


----------



## JaeYoon

I don't think I could boycott any of music industry. I've bought a ton of CDs off bandcamp, some pop, some latin and some upstart post-rock bands. 
I'm still buying them now. I decided instead of spending money on audio equipment, I could've have bought 20 CDs for the price of a $200 piece of gear, etc.

As long as it sounds good, I definitely will get it.


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## TheSonicTruth (Nov 20, 2017)

RRod said:


> I'm sure some marketing dept. can come up with a shnazzy icon that indicates "if you want to actually hear your music in the train station, press this". I posted in another thread that a focus just on normalizing loudness doesn't really get to the issue of your listening environment supporting only a certain dynamic range.
> 
> The issue of course is that it's hard to decide on a default way to do compression writ large. I'm sure a smart enough person could come up with a generally useful loudness-based compression scheme, though. Still, there's bound to be material where a generalized algorithm doesn't quite do it right, which leads to a whole set of issues for artists, engineers, and listeners.



Perhaps a series of values, IE: 'Low, Medium, Max' compress, or a dial-in range, so the average user could just set it to what sounds best, to them, in their local listening environment.

It has to be transparent, though:  When they crank up the compressor feature, no volume adjustment should be required.  Makeup gain is applied as compress feature is dialed up, and is reduced when the feature is turned off.  Seemless.

I know this can be done, in cars and on mobile devices, given the technology and processing power of the last few years.


----------



## bigshot

TheSonicTruth said:


> Cutestudio:  See how both gregorio and bigshot condemn the truth as "nonsense"?  Sounds just like what's coming out of the White House recently!  There's only way to handle the likes of gregorio and bigshot:  It begins with the letter 'I' and contains six letters.  I'm going to do it, and I suggest you, and others fighting the good fight, also take heed.



GREAT ADVICE!


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## bigshot (Nov 20, 2017)

old tech said:


> But if you want evidence, consider the state of the market for audiophile releases.  Most specialist labels do not sell large quantities to justify expanding their production.  Indeed many go broke or just don’t see it as worth continuing, such as DCC or MFSL.



There was an album on MFSL vinyl that impressed me back in the LP era- Waiting For Columbus by Little Feat. Best live concert recording I've ever heard. I decided many years later to buy the CD, so I spent a lot of money and bought the MFSL again because I wanted to get the same mastering. It sounded fantastic. It was such a good album that it spurred me to buy a box set of Little Feat so I could hear what their studio albums sounded like. It turned out that the box included tracks from Waiting For Columbus. When they came on I was surprised to find that they sounded just as good as the MFSL, so I switched back and forth and compared. If there was any difference at all between the regular release and the MFSL version, I sure couldn't hear it. I suspect that sometimes labels will appropriate third party audiophile masterings and release them in their own expensive box sets.



TheSonicTruth said:


> SMH!  I'm only 47.  Why are so many folks who are older than me making such millennial statements??  Guess I'm really conservative culturally: 'Put on some Who and Van Halen and put a six-pack in the fridge for when Mike and Jim and their wives come over!'



That's fine. There's some good music there, but I don't listen to much of what I listened to as a kid any more. For a while I dug down deeper into my "kid music"- looking for similar bands that I hadn't heard of or solo albums or work before they became famous- but it was a dry well. The "greatest hits" really were the greatest hits and that's all there was.

In college I had an epiphany. I was listening to the college radio station and they played Cab Calloway's "Some of these Days" and it blew my mind. The second it came on, I jumped up and cranked the volume. This song had ten times the musicianship and ten times the energy and ten times the fun of anything I had ever heard. I grabbed a piece of paper and a pencil and sat by the radio until the DJ came back to say who it was by. The next day I was at Rhino Records asking the guy for Cab Calloway. I bought an album and devoured it. The next week, I was back at Rhino asking for more stuff like that. He started listing off names... Don Redman, Fats Waller, Fletcher Henderson, McKinney's Cotton Pickers, etc... and I was off and running.

Harlem Jazz led me to trad jazz and be bop. I dived in and tried to understand this stuff that sounded so different to me. Pretty soon I was getting into Cuban mambo and conjunto music. Pop vocals. big band... a boss turned me on to the best of classic country music. The world kept expanding faster than I could keep up following all the breadcrumbs. Classical, opera, ethnic music like Hawaiian slack key guitar and Balinese gamelan, bluegrass. My record collection exploded to over 7,000 LPs. Then I started collecting CDs and it doubled in size again. And I realized that not everything from the past has been released on LP and CD, so I got an acoustic phonograph that plays 78s and my collection doubled again.

At this point I have more music than I can listen to in my lifetime. I listen to music that is new to me every day, and every new kind of music I dive into helps me to widen my appreciation and understanding of all the other kinds of music. It's like learning a language. If I had just stuck with 70s album rock, I would still have the musical vocabulary of a 12 year old kid. That's fine for a 12 year old, but life is short and the world of the arts is vast. I want to experience as much of it as I can before I croak.

Getting back to engineering- I'll tell you something you might not be aware of since you are zoomed in on just one kind of music... bad engineering is not pervasive. It's primarily a problem just in "kid music". Classical music has always had great sound. I have an opera recording on 78s from 1935 that sound amazing. Studio jazz has always had very natural recording and engineering. There are amazing sounding records in the genres of ethnic music, country and easy listening as well.

The problem isn't a general one of engineering and the music business, it's specific to the genre of music you've chosen to focus on. Music aimed at kids are engineered for playing on cell phones and ear buds because that is what kids listen to. You have a nice sound system and you like this kind of music, but you aren't typical. The reason audiophile recordings of dinosaur rock exists is to serve your demographic. It's a small market, but you are being served. But the vast majority of the customers for pop/rock aren't like you.

In other genres, like classical, it's assumed that the listener is going to have a good sound system. The vast majority of classical recordings are remarkably well recorded. There is a pretty small market for audiophile recordings of classical music, but it's much smaller than in pop/rock because most classical recordings since the 60s could be considered audiophile.

You're extrapolating a problem in your particular niche to the entire music industry but it isn't really a valid complaint. It only applies to the type of music you've chosen to listen to. If I was you, I'd just focus on the audiophile labels that release pop/rock. Trying to get the industry to cater to your particular needs when you aren't representative of the overall audience for that music isn't going to get you far. That would be like raging at the ocean.


----------



## pinnahertz (Nov 20, 2017)

Cutestudio said:


> The 'no one's forcing you to buy it' argument. What's up with that? What if we want to buy it but without clipping and over-compression? Isn't our money any good?


That's a meaningless argument.  What if you want to buy a car with 5 wheels and two engines?  They don't make it? "Isn't my money any good?"


Cutestudio said:


> Yes, lets take Adele.
> In the top quote you incorrectly claim that we can't compare stuff (despite the litany of shoddy remangles that are now infamous for mangling and shunned by the HiFi community), so lets compare Adele to Adele. This is easy as she conveniently numbers her albums, take a look at the overall mangling of Adele's 19, Adele's 21 and Adele's 25. *You can clearly see that the mangling on 19 is of it's time and that 21 is mangled worse. *Then look at 25, its mangled even worse than 21.


Do you listen with your eyes? You should stop looking at start listening.


Cutestudio said:


> The mangling of music also coincides with the loss of interest in HiFi by the general public.


No, sadly, it does't.  The "HiFi" market...high(er) end two-channel stereos...has always been a niche.  It's been diminished by the popularity of home A/V systems, often multi-channel. Just look at the number of  multi-channel AVRs vs stereo devices in the market, even if you don't want to bother with actual statistics.


Cutestudio said:


> Once HiFi was something to aspire to and everyone I knew wanted a decent HiFi. It was big business and in every shopping center and mall.
> Now no one cares because of one very simple fact: modern music sounds just as bad on HiFi as it does in the car, radio and MP3 player.


You talk as if everyone aspired to high-end audio!  That's NEVER been true.  The popularity of audio systems was always low-middle quality.  Those awful "rack systems" of plastic chassis components all hooked together in a big black stack driving big black boom/tinkle speakers. That's where the bulk of audio was, and it satisfied the bulk of the market just fine.  And those buyers/owners didn't aspire any higher than that.  The same buying demographic is still in existence and still strong.  They buy HTIB systems for $800, and if they want to impress their friends they buy Bose Lifestyle systems.  Ever heard one of those?  Your ears would fold over and seal themselves shut!  And they spend $3K on them, not because of sound quality! 


Cutestudio said:


> That's why the music industry has become a circus of 'The voice', BGT etc, because the mangling has removed all of the dynamic interest from the music.  Without dynamics and with generic instruments and autotune what you are left with is a product of a) a whining teen, b) soppy PC lyrics, c) a youtube video.


I don't know, I still listen to music recorded recently, and I find a few genres that are smashed to death, but certainly not all.  High quality Jazz recordings seem alive and well, classical is untouched by mangling hands, and there are others.  You talk as if every recording made in the last 20 years were run through a hard clipper.  Just not true.


Cutestudio said:


> So why would any millennial  want a HiFi today? - there's no use for it.


Millennials seem to like vintage gear and vintage vinyl.  That would expose them to less audio mangled in mastering, and more mangled by vinyl and the vintage pornographs with knitting needle stylii in misadjusted tone arms.  (Hows that?  Trying to imitate your style there...)


Cutestudio said:


> So yes, Adele: Good example of the rot setting in, a good boast of 'not a single peak left unclipped'.


I don't think I'd have to search very long for a recording made in 2017 that was not excessively clipped...just not in that genre.


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## pinnahertz

ev13wt said:


> Maybe we need a standard of "no clipping = high quality" as a start. It won't fix anything, but it sure is a start.


The standard for redbook was, at one time, no clipped samples.  However, the standard defined clipping as digital peaks over 0dBFS.  You can achieve that and set your clipping threashold at -.5dBFS, and clip like crazy...if you want.  

But as much as I hate loudness processing, I'd also fight against that sort of standard.  Clipping can be done without audibility, and it is a good and valid means of processing.  Like any tool, it can be subjectively abused.  You can't standardize subjectivity or artistic tools.


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## Strangelove424

RRod said:


> I'm sure some marketing dept. can come up with a shnazzy icon that indicates "if you want to actually hear your music in the train station, press this". I posted in another thread that a focus just on normalizing loudness doesn't really get to the issue of your listening environment supporting only a certain dynamic range.
> 
> The issue of course is that it's hard to decide on a default way to do compression writ large. I'm sure a smart enough person could come up with a generally useful loudness-based compression scheme, though. Still, there's bound to be material where a generalized algorithm doesn't quite do it right, which leads to a whole set of issues for artists, engineers, and listeners.



Hm, you got my wheels turning. Maybe they could make the process semi-automated by adding a microphone to the system (on the wire to the headphones or on the player itself) that could measure ambient noise, and constantly adjust the amount of compression to meet both the minimum value and range value of ambient noise in the environment. It could be customized with a light, medium, and strong setting but the algorithms would do most of the work using mic data. In fact, I wonder how easily this could be designed into the pre-existing hardware of smartphones, which many people use as music players nowadays.


----------



## Strangelove424

JaeYoon said:


> I don't think I could boycott any of music industry. I've bought a ton of CDs off bandcamp, some pop, some latin and some upstart post-rock bands.
> I'm still buying them now. I decided instead of spending money on audio equipment, I could've have bought 20 CDs for the price of a $200 piece of gear, etc.
> 
> As long as it sounds good, I definitely will get it.



Voting with your wallet does not mean total boycott of music, it just means choosing to buy well mastered music only. If it's rock or jazz, buy used albums not new remasters. If a new album comes out, listen to it on a music streaming service to see if its well made before you buy it. These changes in habits are not a boycott, they are merely decisions which reflect developed taste, and if enough people follow this trend the sales impact would be enormous.


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## Cutestudio

gregorio said:


> Mastered for iTunes does not affect the amount of compression/mangling



Your continued ranting about a subject of which you are clearly ignorant is puzzling. Other people will research it, read the forums and reviews and _decide for themselves_.
My message for audiophiles is simply this: don't listen to the disinformation and FUD, read and speak to people who have actually downloaded, checked and listened to the *Mastered For iTunes* music. You will not be disappointed. At least no one so far has been 
E.g: http://www.forums.stevehoffman.tv/threads/the-official-mastered-for-itunes-thread.350068/



gregorio said:


> ignorant nazi dinosaur crap
> G



Ah - Godwins Law strikes again LOL 
https://en.wikipedia.org/wiki/Godwin's_law


----------



## Cutestudio

pinnahertz said:


> Clipping can be done without audibility, and it is a good and valid means of processing.  Like any tool, it can be subjectively abused.  You can't standardize subjectivity or artistic tools.



The issue with clipping the medium (rather than clipped content like fuzz boxes) is twofold:
1. Each DAC's overload character will be different, some get very upset with audible clicks while others hide them better. I.e. the result is now out of your control.
2. Each flat top tends to add 3rd/5th/7th/9th etc. odd harmonics which are the most audible distortion products.

Additionally the most basic limiter and compressor should be able to round off the tops, clipping can never be either necessary or desirable.

Today I walked past a street musician with a microphone and a guitar, the sound balance was actually pretty good and dynamics were well under control. I smiled as I recalled the 'art' and 'skill' which mastering engineers claim is required, the guy I heard must have been a genius .
Maybe he's like the people on here who buy a compressor for film soundtracks, getting a good sound without clipping is hardly rocket science particularly when operating on a fixed track rather than a live stream.


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## gregorio

Strangelove424 said:


> If a new album comes out, listen to it on a music streaming service to see if its well made before you buy it. These changes in habits are not a boycott, they are merely decisions which reflect developed taste, and if enough people follow this trend the sales impact would be enormous.



Absolutely! Instead of stamping their little feet, insulting the very people trying to stop the loudness war and screaming "no more compression" like lunatics, when they don't even seem to know what compression is or how it's used, they should simply vote with their wallets and buy the well mixed and mastered stuff. We've been telling artists and labels not to apply silly levels of compression for years, they nod wisely, totally agree and come back a day or two later and say "sounds great, good job but it needs to be louder". There's only two ways to get them to actually heed the advice, standardise and enforce loudness normalisation, as they have in the TV broadcast world, or hit them in the only place they'll feel it, in the wallet!

G


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## gregorio

Cutestudio said:


> Well, you may wish to impart your wisdom to Apple Inc., you may not have heard of them but they struggle by.



I quoted and gave you a link to the actual document Apple themselves publish and distribute to content creators, providing all their guidelines and tools to create "Mastered for iTunes" certified content and then you say, "you're wrong, go impart your wisdom to Apple". It wasn't "my wisdom", it was Apple's wisdom, I was directly quoting APPLE!!! Honestly, what is wrong with you?

G


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## pinnahertz

Cutestudio said:


> The issue with clipping the medium (rather than clipped content like fuzz boxes) is twofold:
> 1. Each DAC's overload character will be different, some get very upset with audible clicks while others hide them better. I.e. the result is now out of your control.


True, but production clipping should take that into account, and they don't use a fuzz box. 


Cutestudio said:


> 2. Each flat top tends to add 3rd/5th/7th/9th etc. odd harmonics which are the most audible distortion products.


True too, but the harmonics of clipping are masked by other content, and the audibility of distortion in general is a function of time, degree and masking.  Your view is correct, but far to shallow to be related to reality.


Cutestudio said:


> Additionally the most basic limiter and compressor should be able to round off the tops, clipping can never be either necessary or desirable.


Let me introduce you to the wonderful world of audio processing!  SOOO much you don't know here.  A basic limiter modulates the entire channel gain, and therefore its action becomes audible in several ways.  If it attacks quickly and recovers quickly it creates a high level of distortion a lot of the time.  If it attacks quickly and recovers slowly, a high peak causes an overall gain duck.  What you can do with a clipper is to clip a short high level peak without changing the channel gain.  If the peak that is clipped is short, and surrounded with other music,_ the clip is inaudible._  The result is far cleaner than using just a basic limiter. 

But who uses a basic limiter? Nobody!  Not even radio stations!  In recording, processing is done first on the track level, second on the mixing level, and finally in mastering.  All of those possibilities are entirely variable from none at all to lots.  Processing is complex, multiple processors with multiple attack/release characteristics and curves, even multiple bands are handled separately.   That's an _extremely_ basic description, it's far, far more complex in reality.  You're way out of your depth here, arguing things about which you have no idea. 


Cutestudio said:


> Today I walked past a street musician with a microphone and a guitar, the sound balance was actually pretty good and dynamics were well under control. I smiled as I recalled the 'art' and 'skill' which mastering engineers claim is required, the guy I heard must have been a genius .
> Maybe he's like the people on here who buy a compressor for film soundtracks, getting a good sound without clipping is hardly rocket science particularly when operating on a fixed track rather than a live stream.


Sorry, your level of understanding of these issues is almost non-existent.  Helping you out on that would take a totally different thread.


----------



## gregorio

pinnahertz said:


> Processing is complex, multiple processors with multiple attack/release characteristics and curves, even multiple bands are handled separately. That's an _extremely_ basic description, it's far, far more complex in reality. You're way out of your depth here, arguing things about which you have no idea.



I can't even get him to read and understand simple sentences and quotes, so I think you're completely wasting your time trying to explain some technical basics of the equipment he's crying about. Good luck!!!

G


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## Strangelove424

gregorio said:


> Absolutely! Instead of stamping their little feet, insulting the very people trying to stop the loudness war and screaming "no more compression" like lunatics, when they don't even seem to know what compression is or how it's used, they should simply vote with their wallets and buy the well mixed and mastered stuff. We've been telling artists and labels not to apply silly levels of compression for years, they nod wisely, totally agree and come back a day or two later and say "sounds great, good job but it needs to be louder". There's only two ways to get them to actually heed the advice, standardise and enforce loudness normalisation, as they have in the TV broadcast world, or hit them in the only place they'll feel it, in the wallet!
> 
> G



Yes, they need to hit the music industry where it hurts! The power to stop this is not in any single person’s hands. It’s a structure built on sales, and even the CEOs of the labels aren’t positioned to reverse that kind of tide. But the listeners are equipped to do so. They are the collective power. Where they move, the music is certain to follow. If the majority desire free, pirated mp3s then the system will cater to that, and that is what it is currently doing. To compete with 99 cent songs and pirated mp3s, the labels have now turned in desperation to free music streaming services that rely heavily on advertising to recover profits. It’s a sad situation for art and expression, but consumers bear responsibility for the current situation too. They demanded easy and cheap, and that is exactly what they got. Getting angry at producers and engineers who probably value a well mastered album more than anyone does not help the situation, it just creates more battle lines we don’t need drawn, that are unfair to draw in the first place.


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## RRod (Nov 20, 2017)

Strangelove424 said:


> Hm, you got my wheels turning. Maybe they could make the process semi-automated by adding a microphone to the system (on the wire to the headphones or on the player itself) that could measure ambient noise, and constantly adjust the amount of compression to meet both the minimum value and range value of ambient noise in the environment. It could be customized with a light, medium, and strong setting but the algorithms would do most of the work using mic data. In fact, I wonder how easily this could be designed into the pre-existing hardware of smartphones, which many people use as music players nowadays.



I wonder how many people ever played with AC-3 compression options when watching movies at home. Somehow I think anything above 'on/off' would be too much for the top-most interface. Using the mic to make things sound better is a great idea, though I'm sure there is still some beepy-bloopy music that would horribly screw things up.

Essentially we're making a mastering-engineer-in-the-box. How much we could expect such an algorithm, blessed by the industry, to avoid the same pressures that the actual engineers are facing now, I'm not sure. You would need a simultaneous release of un-sausaged masters, which we can't even get now.



Strangelove424 said:


> Voting with your wallet does not mean total boycott of music, it just means choosing to buy well mastered music only.



The problem is that my vote is weighted by my relative rarity as an audiophile.


----------



## pinnahertz

gregorio said:


> I can't even get him to read and understand simple sentences and quotes, so I think you're completely wasting your time trying to explain some technical basics of the equipment he's crying about. Good luck!!!
> 
> G


Yeah, I know, but I woke up this morning and said "Today I'm going to frustrate myself!", so I felt obligated.


----------



## Cutestudio

gregorio said:


> I quoted and gave you a link to the actual document Apple themselves publish



I suggest _you_ trying reading the document yourself

http://images.apple.com/euro/itunes/mastered-for-itunes/docs/mastered_for_itunes.pdf



			
				Apple said:
			
		

> *Be Aware of Dynamic Range and Clipping*
> :
> Many artists and producers feel that louder is better. The trend for louder music has
> resulted in both ardent fans of high volumes and backlash from audiophiles, a
> ...



It's not my fault you don't like Apple's _Mastered For iTunes_ initiative - I guess it bypasses your creative mangling - again, nothing to do with me. I haven't heard people complaining about clips on there - perhaps that statement is why. 

Go attack Apple if you don't like it, stop shooting the messenger and learn to live with people who disagree with you for there are many


----------



## Strangelove424

RRod said:


> I wonder how many people ever played with AC-3 compression options when watching movies at home. Somehow I think anything above 'on/off' would be too much for the top-most interface. Using the mic to make things sound better is a great idea, though I'm sure there is still some beepy-bloopy music that would horribly screw things up.
> 
> Essentially we're making a mastering-engineer-in-the-box. How much we could expect such an algorithm, blessed by the industry, to avoid the same pressures that the actual engineers are facing now, I'm not sure. You would need a simultaneous release of un-sausaged masters, which we can't even get now.
> 
> ...



I use night mode on almost all of my movies. My receiver has 3 compression settings: day, night, and midnight. Midnight probably has a high pass filter of sorts though, it castrates the bass for the sake of my relationship with neighbors. I've also messed with the compressor in VLC, which gives complete control over elements like peak, sustain release, threshold, etc. The problem is I am in no mood to sit down and remaster every movie when I start watching it, adjusting and testing the audio in scenes I've never even watched yet. That's no way to start a film, so I just use my receiver because it seems to work well enough. Ease of use really is important for DSPs I think. When I switch headphones sometimes I let out kind of a sigh, because it means I have to switch around a bunch of stuff. 

You bring up a lot of good points. This is well above my level of ingenuity, but I suppose the algorithm would also need to analyze the track itself, as well as the ambient levels, in order to apply the most intelligent kind of compression. It get's more and more complicated, doesn't it? Until we're doing exactly what you said... making an engineer in a box. With artistic decisions being made by robots. 

I dunno.


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## pinnahertz

Cutestudio said:


> I suggest _you_ trying reading the document yourself
> 
> http://images.apple.com/euro/itunes/mastered-for-itunes/docs/mastered_for_itunes.pdf
> 
> ...


If you read that document too (and comprehended it) you'll notice how they define "clipping":
_"With digital files, there’s a limit to how loud you can make a track: 0dBFS. *Trying to increase a track’s overall loudness beyond this point results in distortion caused by clipping* and a loss in dynamic range. "_

Now, if you can open your mind to this possibility:  What if I deliberately clipped audio, but did, not by exceeding the digital max of 0dBFS, but instead it with a processor at a level just below 0dBFS?  The audio would still be clipped, but wouldn't show up as clipped using Apple's tool, and would likely be accepted as "Mastered for iTunes" too!  

You are unaware of the actual tools available to process audio.  There are actually things called "smart clippers" that hard limit peaks on a dynamic basis, then undo some of the distortion effects.  Surprised?  I knew you would be.  The technology has been around for decades, just not in your world. What would "Mastered for iTunes" do with that?  Nothing.  

I applaud Apple's attempt at improving things, but it's a weak attempt with marginal results at best.


----------



## TheSonicTruth

RRod said:


> I wonder how many people ever played with AC-3 compression options when watching movies at home. Somehow I think anything above 'on/off' would be too much for the top-most interface. Using the mic to make things sound better is a great idea, though I'm sure there is still some beepy-bloopy music that would horribly screw things up.
> 
> Essentially we're making a mastering-engineer-in-the-box. How much we could expect such an algorithm, blessed by the industry, to avoid the same pressures that the actual engineers are facing now, I'm not sure. You would need a simultaneous release of un-sausaged masters, which we can't even get now.
> 
> ...




I hope you know that AC3 compression and dynamic compression are two different things.  AC3 might contain provisions for less dynamic night time listening, but the effect is one of amplitude and not data size reduction.  Just making sure.


----------



## gregorio

Cutestudio said:


> I suggest _you_ trying reading the document yourself



You really do like making a fool of your self don't you! How do you think I quoted from it, if I didn't read it. And, thanks for posting a link to the document I already linked to. At least you're not just picking on my quotes, you apparently can't even read YOUR OWN quotes. Where does it say in your quote (or anywhere else in the document) that they will reject a track or album for over-compression????

And, just for the benefit of others, cutestudio is trying to be clever by stopping his quote at a particular point. The paragraph actually ends:
"_While some feel that overly loud mastering ruins music by not giving it room to breathe, others feel that the aesthetic of loudness can be an appropriate artistic choice for particular songs or 
albums_." Of course, trying to be "clever" is relative and his starting point wasn't exactly high to start with! 

G


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## bigshot

gregorio said:


> I quoted and gave you a link to the actual document Apple themselves publish



OH! Thanks for reminding me! CUTESTUDIO! Have you had a chance to read the article in my sig yet? Your reading list is piling up. Maybe it's time to cut back on your talking and get to work on learning!


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## bigshot (Nov 20, 2017)

Cutestudio said:


> learn to live with people who disagree with you for there are many



There are lots of different opinions... everyone's got one just like you-know-whats. The thing is, some opinions are built on the sand of logical fallacies and others are built on a foundation of facts. If you get so invested in an opinion that you refuse to listen to and consider the facts other people share with you, and you resort to fallacies to try to bully your argument through instead, you aren't part of the latter group.

Now why you're reading this... let your cursor mosey on down to that blue link at the bottom of this post and click on it. You just asked us to click on your link. It's only fair.


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## RRod

TheSonicTruth said:


> I hope you know that AC3 compression and dynamic compression are two different things.  AC3 might contain provisions for less dynamic night time listening, but the effect is one of amplitude and not data size reduction.  Just making sure.



Yes I mean dynamic range compression in AC-3 (p.87).


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## castleofargh

1


Cutestudio said:


> I suggest _you_ trying reading the document yourself
> 
> http://images.apple.com/euro/itunes/mastered-for-itunes/docs/mastered_for_itunes.pdf
> 
> ...


look, you turn every issue into a right vs wrong thing no matter how complex it actually is, so of course what you say is basically false every time. if only for failing to mention all the grey areas left out of your black or white narrative.
if you can't stand it when people disagree with you, be more cautious with what you post. and if you can't help but make everything a personal issue where you're the justice warrior against the evil idiots, maybe it would be better not to post at all. antagonizing posts can only ever lead to personal attacks and this forum has clear rules against those. dispute ideas all day long, but the all "we vs them" rhetoric has to stop. you're not your ideas. Greg or big shot aren't secretly destroying all which is beautiful in life for reasons only evil masterminds understand(at least I don't have solid evidence yet). they simply don't see the world in a binary way and can disagree with your massive exaggerations, while still being saddened when some cool album is remastered by an incompetent noob. they can be against the loudness war and use compressors all day long. things are not as simple as you want to make us see them;

now if everybody else could refrain from responding to provocation with more personal stuff, that would be great. it takes 2 to tango. report personal attacks, don't respond to them by making some more.

now back on the subject even though it has nothing to do with this topic...
the very stuff you quote is showing how mastered for itune is mainly a suggestion guideline and the requirements are actually pretty limited. their main motive aside from money for putting their name on things, was to limit audible clipping. not to fight the loudness war like you seem to believe. maybe you think they're the same thing, but they're not. any idiot can cripple an album and leave a 0.5dB headroom at the top.  mastered for itune mostly tell to do that TBH, they don't have a take in how much compression you think is artistically relevant or how close to the original live event the album must be.


----------



## Don Hills

Another reason for iTunes not wanting clipped input is that the encoding and eventual decoding process can result in "digital overs", where the decoded output can exceed 0dBFS. If the DAC digital output filter and/or the analogue stage don't have the headrooom to cope with this, additional clipping distortion can be generated.


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## TheSonicTruth (Nov 21, 2017)

Don Hills said:


> Another reason for iTunes not wanting clipped input is that the encoding and eventual decoding process can result in "digital overs", where the decoded output can exceed 0dBFS. If the DAC digital output filter and/or the analogue stage don't have the headrooom to cope with this, additional clipping distortion can be generated.




Which should be common sense!

I can't believe the production of digital audio has DEVOLVED to this state, where constant overs and clipping are condoned in the industry! All in the name of having the loudest CD or downloads.  There are a lot of good and talentd acts out there today, but there's a difference between being talented and just plain DUMB.  Have artistst and labels forgotten that volume controls still exist??

And engineers are equally complicit - for violating basic principles of digital audio production just to keep a paycheck coming in!  We need more Bob Katz in this industry and less dogs rolling over.

This abuse of the best audio format of the last 100 years is no wonder that listeners are again seeking out vinyl records and tables to play them on.  And you engineers can going saying the vinyl movement is just a "hipster phase", or a "nostalgia thing" for middle-aged and older music fans.  Yes, there are those factors, but DON'T completely dismiss sound quality as a significant factor in the re-emergence of analog formats.  Even cassette albums of select current releases are becoming available.

There's a reason you are called ENGINEERS.  You are disciplined in a specific field or set of fields, with training on fundamental techniques to turn out a quality finished product.  What if you were building highway bridges?  Would you state that it's ok to regularly exceed the maximum design load even if you know the consequences?  Same principle applies to digital media production.

As for me I appreciate the jagged peaks on my mid-1990s and earlier CDs.  At least I know everything's there!  The way digital audio was intended to be, and why early CD players had as much as 2V maximum outputs.  They were loud enough, as long as basic digital production principles were adhered to when making albums or transfering legacy ones.


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## gregorio (Nov 21, 2017)

RRod said:


> Essentially we're making a mastering-engineer-in-the-box. How much we could expect such an algorithm, blessed by the industry, to avoid the same pressures that the actual engineers are facing now, I'm not sure.



That's not going to happen. In practice, mastering engineers do not just slap on a compressor and dial up the make-up gain. Typically there's at least one compressor, often more than one, plus a limiter and, the response of those compressors and limiters are very carefully set for each track. And, by "response" I don't just mean make-up gain but attack shape and duration, hold, ratios, release shape and duration, etc., there are loads of potential parameters and additionally, loads more parameters for dynamic EQs and multi-band compressions. These parameters are adjusted to create a "character" and creating a bespoke character for each track/album is what makes mastering an art rather than a simple turn the knob to 11 exercise. This is why some mastering engineers can't charge more than $100 to master an album and others can charge many thousands. Some mastering engineers still use vintage analogue compressors or limiters specifically because of their character, in fact, in top class condition, a vintage compressor can go for as much as $40,000! This is a specialist job, requiring specialist equipment, specialist skills and specialist monitoring environments, this is why there are mastering engineers in the first place and why artists making commercial releases always use them! Artist's are not going to sell a piece of unfinished art, and if finishing their art were just a case of choosing some generic compressor preset, why wouldn't the artists or producers be doing that themselves, instead of paying many thousands to a mastering engineer to do it?

Furthermore, the above ignores the fact that much/most of the compression as already been applied during mixing. Note that I specifically said DURING mixing and NOT AFTER mixing and that means that there is no mix without compression! The artists couldn't give/sell you a mix without compression even if they wanted to (and they definitely don't!). To go back to the stage before compression was added, effectively means to go back to the stage before the mix was started, in other words, effectively a hard disk of raw takes rather than an actual coherent piece of music!

BTW, my use of "you" is more in the sense of the plural, to those erroneously clamoring for compression free recordings, rather RRod specifically. Where it's appropriate, most classical and other purely acoustic genres, you're already getting compression free recordings or at least, so lightly compressed you wouldn't notice. And lastly, the analogy with TV and/or films doesn't work!! Films are not compressed (or use very light compression) and TV is compressed but compression in TV and film is relatively straight forward, just some basic settings with a compressor/limiter which is as transparent as possible. There's no creating compression/limiting "character", certainly no expensive "characterful" vintage compressors and there are no mastering engineers in TV or film! So, no big artistic issue with slapping on a generic built-in compressor for a particular listening situation in TV/Film, although we'd prefer that it wasn't employed as it can have some unwanted/unintended impacts.

G


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## 71 dB

TheSonicTruth said:


> And engineers are equally complicit - for violating basic principles of digital audio production just to keep a paycheck coming in!  We need more Bob Katz in this industry and less dogs rolling over.



I don't know about your priorities, but I think paychecks are far more important than dynamic CDs. Why should audio engineers sacrifice their livelihood for you to have your dinosaur rock dynamic? We need an anti-dumbing-down society where high quality dynamic sound is appreciated by masses and audio engineers are _asked_ to produce that to their best capabilities.


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## TheSonicTruth (Nov 21, 2017)

71 dB said:


> I don't know about your priorities, but I think paychecks are far more important than dynamic CDs. Why should audio engineers sacrifice their livelihood for you to have your dinosaur rock dynamic? We need an anti-dumbing-down society where high quality dynamic sound is appreciated by masses and audio engineers are _asked_ to produce that to their best capabilities.



"_Why should audio engineers sacrifice 
their livelihood for you to have your dinosaur 
rock dynamic?_"

If you want to earn my respect, keep statements like ^that out of the dialogue.

As far as 'dumb-down society' goes, that's a thread unto its own.  In the musical sense, those limited and squashed late-90s remasters of legacy albums were advertised as 'new & improved', dangled in front of consumers' noses like carrots, and they bought into it.   So there are two parties complicit in the dumbing down of any society:  A seller with a product, and a body of consumers who failed to THINK of the pluses and minuses before adopting it.  You appear to be from another country, surely an outside observer of how the American government managed to convince its citizens that Saddam Hussen and Iraq were somehow responsible for the 9/11 attacks on NY and Washington - even though most of those terrorists themselves were Saudi-born!   And at that time, a majority(not including me!)  believed it, and a sovereign nation was destroyed based upon pure quackery!  People. Just. Don't. Think.


 It's the same principle with selling music: Convince the public that they need remastered versions of music and movies, or studio-grade resolution to hear them back.  Then crank up the color of the movies, and smash the CD sound super loud, and sell as "better than before".  BS!


Knowledge(most anyway) in this internet age, is free and readily available, even on a Sunday in a town where the libraries are all closed.  Right on that internet tablet.  It's how, ten years ago, I learned the real nature of what was being sold as 'remastered', and began the process of rejecting it and making changes accordingly to my music collection.  And I get comments all the time from people who visit us:  "Hey Sonic, how come your mp3s of the same songs sound better than mine?"  "What's your secret?"  And then they see my CD collection: few or no remastered versions, mostly original, first generation releases.  Knowledge passes on.  People learn to reject that carrot in front of them.


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## gregorio

TheSonicTruth said:


> So there are two parties complicit in the dumbing down of any society: A seller with a product, and a body of consumers who failed to THINK of the pluses and minuses before adopting it.



Exactly, so why are you blaming a third party? The engineer who built that product to the seller's specifications?!!!

G


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## OddE

gregorio said:


> The paragraph actually ends:
> "_While some feel that overly loud mastering ruins music by not giving it room to breathe, others feel that the aesthetic of loudness can be an appropriate artistic choice for particular songs or
> albums_."



And, of course, the prime example would quite possibly be Iggy and the Stooges' "Raw Power". Loudness Wars database entry for one of the releases attached below.


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## RRod

gregorio said:


> This is a specialist job, requiring specialist equipment, specialist skills and specialist monitoring environments, this is why there are mastering engineers in the first place and why artists making commercial releases always use them! Artist's are not going to sell a piece of unfinished art, and if finishing their art were just a case of choosing some generic compressor preset, why wouldn't the artists or producers be doing that themselves, instead of paying many thousands to a mastering engineer to do it?



Right, one rationale behind my posts was to get people thinking about just what it means to 'end the loudness wars' from various possible angles of attack. The 'automatic compression' angle is rife with potential problems, perhaps none bigger than what artists and their handlers expect from their own music these days.


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## bigshot (Nov 21, 2017)

TheSonicTruth said:


> I can't believe the production of digital audio has DEVOLVED to this state, where constant overs and clipping are condoned in the industry!



It's a double edged sword. It's possible now for just about anyone to put together a home studio and record a record. That is good because musicians who probably wouldn't have had an opportunity to record in the past can make their own album in their living room or garage studio. The bad part is that DIY freedom also means that they aren't hiring professional recording engineers. They're grabbing performances on the fly and sometimes they aren't getting everything down perfectly clean. It's pretty much only an issue with pop music, but it can also affect big name bands who want to go back to their roots and make a down and dirty record themselves. This isn't an issue in other genres of music. The quality of engineering in classical music and jazz is higher than it's ever been.



gregorio said:


> Where it's appropriate, most classical and other purely acoustic genres, you're already getting compression free recordings or at least, so lightly compressed you wouldn't notice.



That's because hall ambience and mike placement are accomplishing natural compression. The closer the mike the more compression is required.

By the way, does anyone really think that Iggy Pop didn't want his music heavily compressed? That's the aesthetic they were going for.


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## castleofargh

bigshot said:


> It's a double edged sword. It's possible now for just about anyone to put together a home studio and record a record. That is good because musicians who probably wouldn't have had an opportunity to record in the past can make their own album in their living room or garage studio. The bad part is that DIY freedom also means that they aren't hiring professional recording engineers. They're grabbing performances on the fly and sometimes they aren't getting everything down perfectly clean. It's pretty much only an issue with pop music, but it can also affect big name bands who want to go back to their roots and make a down and dirty record themselves. This isn't an issue in other genres of music. The quality of engineering in classical music and jazz is higher than it's ever been.
> 
> 
> 
> ...


 Iggy is an advocate of the untouched sound like you're with the artist, he explained it clearly years ago:


> So messed up I want you here
> In my room I want you here
> Now we're gonna be face to face
> And I'll lay right down In my favorite place


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## bigshot

He said he was a fan of brick wall compression applied by those dum dum engineers!

Now I'm looking for
The dum dum boys
The walls close in and
I need some noise


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## OddE

bigshot said:


> By the way, does anyone really think that Iggy Pop didn't want his music heavily compressed? That's the aesthetic they were going for.



-That was kind of, sort of my point. 

It is a stellar album in which compression-to-death,  dynamics-be-damned, everything-goes-to-eleven is utilised to great artistic effect, clipping pretty much from the first chord to the last, even.

And yet, I doubt anybody who've ever listened to the album would have wanted it any other way.

Heck, I'll put it on right now. Just got to go get a beer first. (Past 8PM in my neck of the woods)


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## 71 dB

TheSonicTruth said:


> You appear to be from another country, surely an outside observer of how the American government managed to convince its citizens that Saddam Hussen and Iraq were somehow responsible for the 9/11 attacks on NY and Washington - even though most of those terrorists themselves were Saudi-born!   And at that time, a majority(not including me!)  believed it, and a sovereign nation was destroyed based upon pure quackery!  People. Just. Don't. Think.



Yeah, it may seem surprising, but there are people living outside your country, more than 20 times the population of your country in fact. You can clearly see where I am from and sometimes we "outside observers" can see better than the average Joe of your country what's going on, because we aren't brainwashed by bought corporate media only.

However, dumbing-down culture isn't a problem only in your country. Loudness war is _international_. Finnish pop is just as clipped as American pop and Finnish teenagers use the same iPhones to listen to music. People don't think enough anywhere in the world!


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## amirm

Darren G said:


> I enjoy your posts too amirm.  You have done a good job of explaining your position.
> 
> You know it's just that when I read posts like someone linked from Rob Watts, about the benefits 200-300db of noise level (and why stop there, why not 500db?), I choke. When you do the (basic) math, and realize what is being said/written, well hopefully you can see why some are dubious when it gets into the 120db range.   The numbers really don't convey the extremes being talked about.


Thanks Darren.  As to Rob Watts, he gives a new meaning to the word "overboard."    At RMAF I sat in his presentation and it was mostly good until he got into number of taps, distortions down in -200 db and such.  Sadly he says he doesn't believe in blind listening tests so I can see how he has followed that path unimpeded.


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## amirm (Nov 21, 2017)

Strangelove424 said:


> SPL charts like this are all in reference to peak SPL since average SPLs can vary widely based on the variety of quietest sounds.


No.  They may think they are telling you "peak" numbers but that is not what they are.   True peak values may last a PCM sample or two.  As such, you need to capture the audio and have a microphone that you know has no limiting.
Random pointing of an SPL meter at sound may not apply I am afraid.

I mean really.  You don't even know who came up with those numbers yet you are defending their exact nature???

Here is Fielder talking about how majority of such numbers come from average SPLs:





Rest of the paragraph below.



> Do you include the parts of the singer talking to the audience when  averaging a rock concert?


What?  Of course not.  For determination of peak SPL, you want peak loudness.  What does your question has to do with that?  Peak is peak.  It is the absolute loudest something gets because that is what we need to store in our digital samples.  We then need to allow for quietest sound we can hear and that sets the lowest level.  Take the ratio of these two, express it in log db numbers and you have  your dynamic range.



> It's a grey area, so people refer directly to peak. Look closely at the 105db number for classical music, its says during “loud passages”. That means peak. I


No, it means nothing because it is not a proper study.  A random survey does not make for scientific data.  You are just going by anecdotal data that serves your point of view.  I understand that but for heaven's sake, you can't dismiss authoritative, peer reviewed study for this specific purpose and chase random charts like that.  We would all be dead if the medical industry researched drugs that way.  



> BTW, that study is filled with holes, I can’t understand why you treat it as the gold standard besides the selfish reason that its the only document in existence backing you up. For one thing, they made their very own SPL meter! Did they document how they designed it or calibrated it? Nope! This study lacks transparency, and its data is an outlier. It’s simply not valid, and has made very little attempt to establish its own validity.


Oh really?  Where is there calibration data for the list you provided?

You haven't even read the paper yet have all of these objections?  Here is the paragraph before what I quoted to you on the work that others had done prior to Fielder doing his own work:





This is a 17 page paper has a whopping 67 references at the end.  Here is the last ones:





It ends with his bio at the time:





I have not only read all of his papers but a bunch of the underlying research he references.  You have read what on this topic?  Some random things you googled???

Answering your question anyway, that data is in another peer reviewed paper in J. AES by Fielder references in the above paper:









There is tons and tons of detail in the papers.

Really, you have not post one line of authoritative research like this.  Nor has anyone else who is complai*ning.  Now that you have the real data, it is time to show that you care about learning about audio science.  Isn't that what we ask subjectivists to do?  Yet when it is our turn we cling to anecdotal onliners online instead over real data???*


----------



## amirm

RRod said:


> I'm perfectly fine with a world of various masters, I just don't think one should cost more than the other. Or do you disagree?


Depends on what hat I put on.  As a cheap consumer, sure, I like everything for less money.    But knowing how digital music industry has become a money losing proposition at prices Apple and Amazon charge, I don't mind paying more so that there is a healthy and competitive marketplace for digital music distribution.  I can't ask the general public to pay more.  But I think we can ask the audiophiles to do so if in return they get a clearly superior product to CD and lower mass distributed formats.


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## bigshot (Nov 21, 2017)

Apple and Amazon have given consumers a vast ocean of great sounding music at very low prices. They are making the music business better because they are creating a direct conduit for distribution of content. No more warehouses full of unsold inventory and retailers and distributors adding mark ups all the way down the chain. It's the greatest advancement that has happened to music distribution since the advent of radio. More music. Great sounding music. Lower prices. Nothing to complain about there.



amirm said:


> I sat in his presentation and it was mostly good until he got into number of taps, distortions down in -200 db and such.



Yeah. We had a joker around here the other day saying that the highest acceptable noise floor level was 120dB. Can you imagine that?!


----------



## TheSonicTruth

amirm said:


> Depends on what hat I put on.  As a cheap consumer, sure, I like everything for less money.    But knowing how digital music industry has become a money losing proposition at prices Apple and Amazon charge, I don't mind paying more so that there is a healthy and competitive marketplace for digital music distribution.  I can't ask the general public to pay more.  But I think we can ask the audiophiles to do so if in return they get a clearly superior product to CD and lower mass distributed formats.



I don't see 'audiophiles', or the 'general public.  I see music fans - period - and think they should all be able to afford quality, dynamic music.

It's the content, not the format, that matters, and the less processed a song is in mastering, the cheaper it is to produce!


----------



## TheSonicTruth (Nov 21, 2017)

71 dB said:


> Yeah, it may seem surprising, but there are people living outside your country, more than 20 times the population of your country in fact. You can clearly see where I am from and sometimes we "outside observers" can see better than the average Joe of your country what's going on, because we aren't brainwashed by bought corporate media only.
> 
> However, dumbing-down culture isn't a problem only in your country. Loudness war is _international_. Finnish pop is just as clipped as American pop and Finnish teenagers use the same iPhones to listen to music. People don't think enough anywhere in the world!



And I appreciate the insight of those living outside of my birth nation, perhaps more than the vast majority of MYOBs I share it with.

Most of the music on my devices is not clipped, ripped from CDs at least twenty years old.  That which is clipped or over-compressed is from the few recent CDs in my collection, and downloads of the newest entries.

It's all loudness-normalized, meaning just about all of the last twenty years of material saw anywhere from 6-10dB negative replay gain applied, to match the legacy stuff loudness-wise.  I just have to turn my volume controls up a little more than most, but, that's why there'a a range of volume, right?


----------



## Strangelove424

amirm said:


> No.  They may think they are telling you "peak" numbers but that is not what they are.   True peak values may last a PCM sample or two.  As such, you need to capture the audio and have a microphone that you know has no limiting.
> Random pointing of an SPL meter at sound may not apply I am afraid.
> 
> I mean really.  You don't even know who came up with those numbers yet you are defending their exact nature???



A single PCM sample? Here we go, we’re back to transients again. I bet if I gave you enough time, you’d wind this trail through all of audiophilia. We go from needing more bits to needing more samples now. You deliberately ignore or twist the facts to suit your narrative. I have never seen an SPL chart such as this refer to average when speaking of music. Maybe it’s easy to average a jackhammer, but music is always mentioned in terms of loudest passage or peak. I’m defending these numbers because they match my own experiences attending many concerts, and recording live music performances. Experiences that keep my understanding of these numbers in context to the REAL WORLD. 



amirm said:


> What?  Of course not.  For determination of peak SPL, you want peak loudness.  What does your question has to do with that?  Peak is peak.  It is the absolute loudest something gets because that is what we need to store in our digital samples.



Wow, Amiram, you are quite a piece of work. You originally told me all the numbers are average SPLs, and once I proved the ridiculousness of averaging a rock concert, now you are saying “a peak is a peak”? I was wrong, disingenuous is a euphemism for you. 



amirm said:


> No, it means nothing because it is not a proper study.  A random survey does not make for scientific data.  You are just going by anecdotal data that serves your point of view.  I understand that but for heaven's sake, you can't dismiss authoritative, peer reviewed study for this specific purpose and chase random charts like that.



These "random" charts (sourced from a book by Daniel Levitin, and vouched for by UCSD) are corroborated by the experience of thousands of people who actually work with audio. Some of those people in this thread have been trying to get you to recognize sense. I'm done with all the references to your quack study. It is not authoritative and not peer reviewed. In fact, a basic critique of it in the pages of this thread have brought up all sorts of weak points! Anybody who makes their own SPL meter, doesn’t mention how they tested or calibrated it, and then goes and measures the ambient level of an empty music hall, is not doing valid work. I don’t care what his title is. If you could think for yourself, you wouldn't keep mentioning his title either. You are appealing to authority, not reason.


----------



## amirm

Strangelove424 said:


> A single PCM sample? Here we go, we’re back to transients again.


Again, you don't understand the core science and engineering here.  When you make a recording, you must record *all samples. * You can't record averages.  You must include the full amplitude of the peak, and the quietest moments.



> I have never seen an SPL chart such as this refer to average when speaking of music.


As I explained, this is implied when you don't see the word "peak" and ways the data was gathered.  Here is another reference, this time from Journal of Acoustic Society of America, ABSOLUTE AMPLITUDES AND SPECTRA OF CERTAIN MUSICAL INSTRUMENTS AND ORCHESTRAS • BY L. J. SIVIAN, H. K. DUNN, AND S. D. WHITE:






Notice the integration time between peak (1/8 second) and average (15 seconds).  Almost all data that you see on SPL levels is with respect to noise and hearing damage which occurs in minutes and hours so it is NOT based on peaks.  You need to specifically seek out peak data to find it and that information is only in a handful of research papers as I have quoted.



Strangelove424 said:


> These "random" charts (sourced from a book by Daniel Levitin, and vouched for by UCSD) are corroborated by the experience of thousands of people who actually work with audio.


I already quoted Daniel's book for you.  That is NOT his data.  He is just posting that same information with no attribution on where it came from.  It is no different than any chart you find online, all which are aimed at warning you about damaging your hearing.  They are not intended for defining the dynamic range you need for transparent storage and conveyance of digital music.  You can't use lay data like that to counter specific and targeted research by luminaries in our field.

Come back with at least one paper that is peer reviewed and published in J. ASA, J. AES, IEEE Spectrum, Acoustica, etc. and we can talk. Let's not  be so closed minded to what audio science really teaches us lest we want to look worse than the worst of subjectivist audiophiles.


----------



## amirm

Strangelove424 said:


> I'm done with all the references to your quack study. It is not authoritative and not peer reviewed.


I missed this fantastical statement.  Everything I am posting is from Journal of AES and Journal of ASA.  Both have strict peer review requirements prior to publication:

http://www.aes.org/journal/





Here is the journal paper in question: http://www.aes.org/e-lib/browse.cfm?elib=7948





And J. ASA:





And paper I quoted from it:






In contrast, what was it that you showed us?  Some link online and not to the source itself?  And the source was a book which has no backup whatsoever on where the data came from.

So it is not like you cared about anything authoritative or heaven forbid peer reviewed.  

I mean really, I have had these discussions for years and this is the first time I have seen someone not know that JAES papers are peer reviewed.


----------



## amirm (Nov 21, 2017)

gregorio said:


> Current professional music engineers and artists are obviously creating music for today, not for the 1970's, so of course they are going to look at the likes of cutestudio and amirm as some sort of nazi dinosaurs, stuck in a 1970's - 1980's time warp.


Come again?  Much of what I listen to is modern music.  Let's take a case in point an album of mine which I bought back in 1999 called "UP UP UP UP UP"  by Ani DiFranco.  This is its amplitude response:





Nice avoidance of peak amplitude.  Now let's look at the abomination that our mastering friend whom I quoted earlier, Brian Lucey has created *for the same band* in their album, _binary, _*released in 2017*.  This is the spectrum for the "high-res" 24-bit version that I bought:





You see all those clipped or nearly clipped peaks?  You see how the music never, ever settles down?

Again, this is music from the exact same group.  No genre change. What used to be a proper mastering is now butchered to hell.  And you are calling us the problem?  I don't think so.

The *worst enemy of our system performance is the badly recorded and mastered content.*  It is time you recognized the significant role your industry is playing in destroying the enjoyment we could get out of our music.

So I say again, if you want to do any good, start by cleaning up the noses of your own peers.  They are doing us far more harm than allowing customers to play high resolution music.


----------



## Strangelove424

amirm said:


> Again, you don't understand the core science and engineering here.  When you make a recording, you must record *all samples. * You can't record averages.  You must include the full amplitude of the peak, and the quietest moments.
> 
> 
> As I explained, this is implied when you don't see the word "peak" and ways the data was gathered.  Here is another reference, this time from Journal of Acoustic Society of America, ABSOLUTE AMPLITUDES AND SPECTRA OF CERTAIN MUSICAL INSTRUMENTS AND ORCHESTRAS • BY L. J. SIVIAN, H. K. DUNN, AND S. D. WHITE:
> ...



A recorded file is a record of all samples taken. I have no idea what "recording an average" even means. Recording = taking samples, not averages. And a crescendo doesn't last a single sample! Nor have I ever seen an SPL meter that averaged readings every 15 seconds. You are arguing that more samples are needed in order to capture a peak, and that is ridiculous. Some SPL meters have an averaging capability, some don't. But they all do peak. And you don't need ultra fine sample rates to capture the peaks of a concert. The biggest irony I find in all of this is that many bands carry around an SPL meter so they won't get fined for getting into the db ranges you are advocating for! You ignore standard practice, reason, and even law!



amirm said:


> Everything I am posting is from Journal of AES and Journal of ASA.  Both have strict peer review requirements prior to publication:
> 
> http://www.aes.org/journal/
> 
> Here is the journal paper in question: http://www.aes.org/e-lib/browse.cfm?elib=7948



https://en.wikipedia.org/wiki/Argument_from_authority


----------



## bigshot

amirm said:


> Come again?  Much of what I listen to is modern music.  Let's take a case in point an album of mine which I bought back in 1999 called "UP UP UP UP UP"  by Ani DiFranco.



There is some nice acoustic guitar playing on that album. Does she play that herself or does she just sing? Some of the mix is weird how the acoustic guitar is miked super close and the rest of the band is at a normal distance. They don't sound connected, that's why I asked if she played it.


----------



## reginalb (Nov 21, 2017)

amirm said:


> Again, you don't understand the core science and engineering here.  When you make a recording, you must record *all samples. * You can't record averages.  You must include the full amplitude
> of the peak, and the quietest moments.



Who argues that you shouldn't use 24-bit[1] during recording and mastering[2][3]? The topic of this thread is in reference to delivery format, not recording format.[4] But we all know that DSD[5] is the best format for masters.[6]

Now, back to your repeated[7] logical fallacies[8][9][10][11], you actually tried to prove[12] the veracity of a paper by saying how many citations it has at the end[13]. Man, did you know that my post here has WAY more citations than yours[14], so it's better?[15]

[1] https://en.wikipedia.org/wiki/Audio_mastering
[2] https://www.kvraudio.com/forum/viewtopic.php?p=6212136
[3] https://www.justmastering.com/article-masteredforitunes.php
[4] https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/
[5] https://en.wikipedia.org/wiki/Direct_Stream_Digital
[6] http://www.mojo-audio.com/blog/dsd-vs-pcm-myth-vs-truth/
[7] https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-306
[8] https://www.logicallyfallacious.com/tools/lp/Bo/LogicalFallacies/21/Appeal-to-Authority
[9] https://yourlogicalfallacyis.com/
[10] https://owl.english.purdue.edu/owl/resource/659/03/
[11] https://en.wikipedia.org/wiki/List_of_fallacies
[12] https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-306
[13] https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-306
[14] https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-307
[15] https://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=1&cad=rja&uact=8&ved=0ahUKEwiOkOTRitHXAhVC1WMKHcEDDOQQFggnMAA&url=https://arxiv.org/pdf/1301.4597&usg=AOvVaw0H1Bo-xLQ8xl9Rp0skk6v0


I'll set aside being a jerk for a moment, in your post, you quote the following:

"BTW, that study is filled with holes, I can’t understand why you treat it as the gold standard besides the selfish reason that its the only document in existence backing you up. For one thing, they made their very own SPL meter! Did they document how they designed it or calibrated it? Nope! This study lacks transparency, and its data is an outlier. It’s simply not valid, and has made very little attempt to establish its own validity."

You quote something that makes 2 DIRECT assertions to contradict what you've cited.

[1] The data contained is an outlier, many other sources contradict it
[2] The cause of this could be that they used their own SPL meter without documenting how it was created

Then you kind of address the first one, but only by pointing out that other papers have the same weakness, which of course doesn't do anything to help yours out, and then _completely _fail to even attempt to address the more important assertion, number 2. That one is very important, because you have _repeatedly _asserted that your points are more valid because they're backed by serious studies, scholarly evidence, people way smarter than all of us plebs in here. Yet you fail to acknowledge that if the above assertion is true, then the work isn't repeatable, which is one of the hallmarks of research. You _*must *_be able to replicate experiments. Else you have no way for us to know why your data is an outlier - for all we know it could be completely made up. Here is a real citation that you should read: https://undsci.berkeley.edu/article/howscienceworks_17

Also, you actually used the author's bio as your other piece of evidence that it's a good paper. Are you even listening to yourself here?

But I know that I'm more right than you, because this post has more citations than yours. Plus, look at this bio:

Reginald Brown received a PhD in knowing way more than you in regards to audio. Then he went on to win 1 million Grammy awards, went back in time and founded every single headphone company ever, then went forward in time and got a 2-billion bit DAC from the future which has a way lower noise floor, and it can be ABX'ed all day long.


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## TheSonicTruth (Nov 21, 2017)

amirm said:


> Come again?  Much of what I listen to is modern music.  Let's take a case in point an album of mine which I bought back in 1999 called "UP UP UP UP UP"  by Ani DiFranco.  This is its amplitude response:
> 
> 
> 
> ...



Remember, it is possible that the band wanted the sound that way, as represented in your second waveform pic.  You're right, it looks(and probably sounds) like A$$, but try to understand what, largely, is driving the demand for such 'mastering'.  (!)

If there's one positive aspect of it, from purely a marketing standpoint, it's that the consumer will hear it and think, WOW, this new high-res audio is farrr ouuuut!


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## bigshot

I love it when people make judgements about music by looking at graphs. Personally, I use my ears.


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## Slaphead

bigshot said:


> There is some nice acoustic guitar playing on that album. Does she play that herself or does she just sing? Some of the mix is weird how the acoustic guitar is miked super close and the rest of the band is at a normal distance. They don't sound connected, that's why I asked if she played it.



Yeah she plays the guitar herself - quite an aggressive style as well

Her music is pretty much about her vocals and guitar, with any band really only providing backing, hence the closely miked and forward guitar.

I haven't listened to any of her stuff for a while - must dig some out again.


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## reginalb

Cutestudio said:


> ...This is why Mastered For iTunes is so great, we have a referee of sound quality to pass: *Apple*, who catch and reject the mangled rubbish that gets out from the dying labels, because they don't want to sully their _Mastered For iTunes_ brand with your junk. It works too. After reading about how bad CD and MP3 mastering - with it's non existent QA - is, take a look at how pleased people are with the Mastered For iTunes tracks....



Dude, how many times does Apple's own documentation have to be quoted back to you for you to stop espousing this claim? I've actually tested "Mastered for iTunes," they're just no different. 

Since you're probably in to the dynamic range database, let's use actual data points:

Master of Puppets (Mastered for iTunes - DR score of 10) http://dr.loudness-war.info/album/view/138940
Master of Puppets (Original CD - DR score of 12) http://dr.loudness-war.info/album/view/135194

Dance of Death (Mastered for iTunes - DR score of 7) http://dr.loudness-war.info/album/view/98034
Dance of Death (2004 DVD-Audio release - DR score of 10) http://dr.loudness-war.info/album/view/132211

Collapse into now (Mastered for iTunes - DR score of 7) http://dr.loudness-war.info/album/view/118925
Collapse into now (FLAC download - DR score of 7) http://dr.loudness-war.info/album/view/112308

So...no. They don't do a thing about loudness.


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## gregorio

amirm said:


> *Now that you have the real data, it is time to show that you care about learning about audio science.  Isn't that what we ask subjectivists to do?  Yet when it is our turn we cling to anecdotal onliners online instead over real data???*



You are doing it AGAIN amirm!! Yes the noise floor of certain concert halls, when they are empty with the HVAC switched off, can be 0dBSPL in the critical band. Yes, peak levels of say an orchestra can exceed 120dBSPL. I'm NOT arguing with those figures, I'm arguing what should, to anyone without a specific agenda, be painfully obvious: How can you possibly equate the two? In an empty concert hall there is no symphony orchestra, so can that non-existent orchestra produce 120dBSPL or in fact any SPL/Dynamic range at all? In "real life", which you like to quote, the noise floor is the hall plus musicians, plus audience, plus HVAC and all that most certainly is not 0dBSPL or anywhere near it!! So what actually is the noise floor and what is it's spectral distribution? I don't know, I can only guess based on experience. As far as I'm aware there are no AES papers with peer reviewed studies on concert hall noise floors including audience, musicians and HVAC. You seem to be saying that as it's not been published by the AES therefore the noise generated by all the musicians, audience and HVAC doesn't exist, that we can't mention it here in the sound science forum and you can completely ignore it and effectively state we need more than 16bit to reproduce with high fidelity the dynamic range of an orchestral symphony concert which was performed with no orchestra and no audience! 



amirm said:


> It is time you recognized the significant role your industry is playing in destroying the enjoyment we could get out of our music.



It's time, in fact well beyond time, you recognised that the industry responds to the demands of the market and if the market is demanding cheap, that is what it's going to get. You can't have both cheap and very high quality at the same time, I cannot take my money for a new Ford Focus and demand a Bugatti, and I can't blame the Ford engineers for not making a car with the same quality as a Bugatti for the price of a Focus!! Bugatti only survives because there are enough super-rich willing to spend silly money to own one, if there were enough people willing to spend serious money on high quality recording, mixing and mastering, the industry would absolutely be fighting for that market. Come on, this isn't doctoral level economics we're talking about here!

G


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## castleofargh

TheSonicTruth said:


> "_If you can't hear the damage and you
> refuse to look to see the extent of the vandalism
> you create and espouse; you are clearly in the
> wrong job. If I was your boss you'd have been
> ...


I'd like to take this opportunity to apologize to you. I thought you were one of those blind justice warriors who can only see good or bad in everything no matter how complicated the issue, and I've treated you poorly after seeing you doing all the grave digging of dynamic/mastering topics. but you've made a few reasonable posts like this one that are forcing me to see you in a different light. I really hope I was wrong about you and we'll keep seeing more of that type of posts in the future.


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## bigshot

Slaphead said:


> Yeah she plays the guitar herself - quite an aggressive style as well.



Color me impressed then. That's an idiosyncratic way of playing acoustic guitar. I like it. I bet she is really good live.


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## bigshot

Cutestudio said:


> Do you really? Because the amplitude graphs and histograms expose the incompetence of the mangling and show the exact damage the manglers force with their crude limiters and clips.



No. It's simply a graphic representation of sound. Depending on how a graph is drawn, it can show things that are completely inaudible, or it might make things that are forward and prominent appear to be small. Readouts and graphs are handy, but the ultimate arbiter is the ears. That is the first lesson of engineering.

In my sig file down there are two videos of seminars from the AES. The videos have links to the sample audio files they are using in their demonstration. You can download them, play them on your own headphones and hear the truth for yourself. A lot of people around here are really good at slinging graphs and quotes from books and abstract specs around. But unless you know what a dB or a Hz actually sounds like in real life, it's just printed words about sound or pictures of sound or numbers that represent sound... it isn't sound.

Speaking of which, now you'll have to check out all three links in my sig! I guarantee you that you will learn a lot. I think you're paying attention to the wrong sorts of websites. It's better to listen to professionals than it is to follow armchair self appointed internet forum experts. I don't claim to be an expert myself, but I do have some professional experience, and that has given me a practical outlook that seems to be in short supply around here sometimes.

Well, here are the links for you, Cutestudio. Go to town!


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## amirm (Nov 22, 2017)

bigshot said:


> I love it when people make judgements about music by looking at graphs. Personally, I use my ears.


Oh, I used my ears first.  But they didn't want to have anything to do with such content.  So I pulled it into Adobe Audition to analyze and the VU meter there protested and said it wanted to quit!!! 

Here is the track from 1990s UP UP UP UP album:



Delightful.  I can't find the track I analyzed  for their 2017 album (Binary) on Youtube but there is another one there, _Deferred Gratification.  _Here is its time domain waveform:






As you see, just prior to 1:00 minute it all goes to hell.  Here is the youtube video.  Start around 45 seconds and go:



Listen to those drum kicks and tell me they are not distorted to pieces.

Here is Ani with  live recording of the track I analyzed before and not subjected to so called "mastering":



Isn't that delightful?

Here is its spectrum:





Compare this with the abomination in my last post.

So go ahead, use your ears, tell us if you disagree.


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## bigshot

gregorio said:


> You are doing it AGAIN amirm!! Yes the noise floor of certain concert halls, when they are empty with the HVAC switched off, can be 0dBSPL in the critical band. Yes, peak levels of say an orchestra can exceed 120dBSPL



I'll actually quibble with those figures...

re 0dB: A concert hall has a noise floor too. Most of them are in cities with traffic noise outside. The walls aren't soundproof. I've been to Carlsbad Caverns National Park. When you get to the bottom of the cave, the guide asks you to close your eyes and stand perfectly still. THAT is 0dB, and it isn't silent... you can hear the blood pumping through your ears. Your body has a noise floor.

re 120dB: If you stand right in the middle of an orchestra with all the instruments aimed at you playing at full forte, you might hear 120dB of sound. But it would sound awful. It would be painfully loud and you wouldn't be able to sort out the various instruments because the brass would overpower everything with loud directional sound. No one listens to orchestral music like that and that isn't the perspective that classical music is recorded at. The conductor is controlling the dynamics with an ear for creating the perfect balance *for the audience.* The sound engineers place the microphones at a little distance from the band, often overhead. The goal is to capture a comfortable listening level which reflects the dynamic range limitations of the human ear- meaning 45-55dB of dynamics max. It's also important to note that the dB scale isn't evenly spaced. Normal listening volume in the home is around 70dB. 120dB is SIXTEEN times that.

I think I read a story about Bernstein recently (maybe it was here at HeadFi) where the score read full forte. So as an experiment, he instructed all of the instruments to play at full forte... no one holding back anything. It was deafening. They got as far as four or five notes before Bernstein cried uncle and stopped them. He never asked that of his players again. Just because an orchestra can hit 120dB, it doesn't mean that they play 120dB.

The only  thing that goes from 0dB to 120dB would be a jackhammer at close range in Carlsbad Caverns. That has absolutely nothing to do with recorded music.


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## amirm

71 dB said:


> I don't know about your priorities, but I think paychecks are far more important than dynamic CDs. Why should audio engineers sacrifice their livelihood for you to have your dinosaur rock dynamic?


Sure, they should chase the money to put food on the table.  But then don't come here and tell us that folks shouldn't charge more than 10 cents for a DAC or else we are being "ripped off."  I got ripped off to the tune of $15 for that Ani DiFranco album.  But no, let's complain about audiophiles.  They are fools for buying expensive audio gears but somehow are not for buying over compressed music???

That is why we have this discussion.  And the same one I was having with that mastering engineering bragging about how he tries to do good for his clients:






He is there on that gearsluz thread telling us what signal processing technology is good for us.  Yet his own house is in such disorder.  Now, I could fight his words with words but I like to get data.  I went to his web site, searched and found that album and bought it.  And the results you have seen.

That is the problem here.  We can't fix loudness wars in this thread.  We can however show a mirror to people, proving that their is no honest cause here.  They have no interest in real fidelity as the most important thing.

_"One of the best in the world, and my ears...same." _ Yeh right.


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## Niouke

quite frankly it's your taste speaking there, It's obvious each album wants to be different, some un plugged à la Chesky, and the other with a much much "fuller" bass sound. The Bass takes all the space in a recording and will inflate the VU meter.

I could create a track with a small VU meter than sounds shallow, distant, and not good to me, and another flirting with the peaks that would fill up my ears with pleasure. Mastering is part of the creation process and pointing finger at compression and limiters never helped any engineer.


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## bigshot (Nov 22, 2017)

amirm. you'll find that you can't really compare spectrum graphs like that unless both tracks are normalized to the same level. It can look completely different to the eye at -20dB or -30dB. Also, the track used on a rock video is almost always more compressed than the same track on the album. It may even be a slightly different mix. For videos, they mix/master the sound specifically to be played through TV speakers. I produced a video for Bjork and One Little Indian provided us with a "TV mix" of the song. It wasn't quite the same as the CD version we were using as a reference track before the layback. The tried to give us an entirely different mix of the song, but it didn't hold sync with what we had already edited.

I think the misunderstanding that many audiophiles have is that they think there is one "perfect" master. There isn't. There are multiple masters that are perfect for various applications. Even the original studio master isn't perfect. Each song is engineered individually. They usually won't play through as an album because the levels jump around from song to song. The problem is that the people doing print and tape who are in charge of distribution don't always know which master to send out. If an engineer was in a hurry and didn't label the master clearly, or if he wasn't aware of how the other elements were labelled, mistakes can happen.

I once caught a dub being made of a show I worked on that where the dubs were being labelled "Original Master". They were doing that because our studio was retaining the original master and we were sending out a sub master for distribution. The distributor had requested that their dub be labelled "original master" so they could tell it from their various distribution masters. The problem was that their license was for three years, and at the end of that their dubs would be returned and vaulted with the studio's original original master. Perfect opportunity for confusion there. I fixed it by having them label the dubs original distribution master. But stuff like that happens all the time.


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## amirm

Niouke said:


> quite frankly it's your taste speaking there, It's obvious each album wants to be different, some un plugged à la Chesky, and the other with a much much "fuller" bass sound. The Bass takes all the space in a recording and will inflate the VU meter.
> 
> I could create a track with a small VU meter than sounds shallow, distant, and not good to me, and another flirting with the peaks that would fill up my ears with pleasure. Mastering is part of the creation process and pointing finger at compression and limiters never helped any engineer.


No, it is not a taste issue.  I would enjoy fuller sound just the same.  Here, I am talking about *pure distortion.*  Did you listen to the tracks?  Did you think the highs and lows are clean?  They sound dreadfully distorted to me.  As I said, I noticed the problem first by listening and could not believe how distorted it was getting.

Let's remember that not only was this offered to general population but also given to us as a "high-resolution" album.  





You really don't think they could do better for us as the audience and for $16?  Didn't we get taken to the cleaners here?


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## gregorio (Nov 22, 2017)

bigshot said:


> I'll actually quibble with those figures...  re 0dB: A concert hall has a noise floor too. Most of them are in cities with traffic noise outside. The walls aren't soundproof.



I'm certainly not saying all concert halls. I was quoting amirm and his figures for the Dallas Symphony hall. I've actually worked at that concert hall and spent quite a few enlightening days with the hall's designer (Russel Johnson) back in the early 1990's when Birmingham Symphony Hall (UK) opened, which he also designed and to the same principles as the Dallas hall. If you turn the HVAC off and you're the only person in there (as I have done) they are amazingly quiet, as they're full suspended double shell constructions with no expense spared on acoustics and isolation. However, they are outliers! Two of the best symphony halls on the planet and certainly two of the quietest, which is presumably why amirm chose it! The great old concert halls such as the Vienna Musicverein and Concertgebouw, which I've also been lucky enough to work in, have lovely acoustics but their noise isolation is nowhere near that of the Dallas and Birmingham and of course we've got other famous concert halls such as the Royal Albert Hall, which I've worked in extensively, which has neither good isolation nor good acoustics but it does have a great atmosphere with a full house!

So yes, in general I would agree with you but amirm certainly does like his outliers!!

G


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## bigshot (Nov 22, 2017)

Amirm, do you know where the distortion was introduced? Was it caused by the encoder when the youtube video was uploaded? Was it a dub that had incorrect levels? Or was it on the original master? Red Hot Chili Peppers' Californication has some amazingly distorted stuff on it, and it doesn't sound deliberate. I'm told that they recorded it that way.


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## amirm

bigshot said:


> amirm. you'll find that you can't really compare spectrum graphs like that unless both tracks are normalized to the same level. It can look completely different to the eye at -20dB or -30dB.


Did you listen to the tracks, i.e. use your ears?

And yes, the problem remains whether normalized or not.  Here is the histogram which is normalized to 100% for the 1990s album:



 

I put a line at -30 db.  We see tons of audio samples at that amplitude and lower.  Again, normalized to 100%. 

Here is our 2017 "high-res" album:



 

As you see, there is hardly anyone home at 0-30 db and lower as indicated by the yellow line.


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## Niouke

listening to binary on spotify right now, while its obviously mastered for a "competitive" volume and the voice suffers from it, the use of stereo effect for certain tracks is interresting I find. My opinion is that the singer is so bad the engineer had to tone it down lol

I listen to much worst stuff than that, like Weezer the blue album, but a grunge album recorded at a moderate level would be weird wouldn't it?


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## bigshot (Nov 22, 2017)

That is a classic example of an album being mastered to suit headphones. On my speaker system, the top one would sound good. One a lot of headphones, the bottom one would. By the way, what does the bottom scale in those charts represent? I can't figure out the numbers.


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## amirm

bigshot said:


> Amirm, do you know where the distortion was introduced? Was it caused by the encoder when the youtube video was uploaded?


No, as I said I heard the distortion in the original album I bought.  Indeed the levels overall seem lower in Youtube and you can't hear the full (distorted) effect that I am hearing with the uncompressed original.

As to where it is, I see that where the meter pegs zero, is where the problem is so my money is that the analog EQ and compressor that Brian Lucey used is a lot at fault.


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## bigshot

amirm said:


> No, as I said I heard the distortion in the original album I bought



Then it's probably recorded into the music. Nothing to be done about it.


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## Niouke

A track I like, and was shocked by the low DR score (no so much as she features drake on the album). I'm seriously doubtful of DR as a measure for quality music, or quality mastering in any case. These guys are wasting their time.


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## Strangelove424

Low DR does not necessarily mean loud and obnoxious. I went to sleep last night listening to Spotify relaxation playlists, and when they'd play a dynamic track I got really aggravated because I wanted calming, steady music. Something like "strings for sleep" really won't have a very high dynamic range at all. It won't have a high level either, but the point is you pick the mastering for the job, not the mastering that suits a theoretical number. People assume low DR=Nirvana, but low DR=lullaby music too. 

I think I should stick this in my sig for a while: http://productionadvice.co.uk/its-all-about-great-sound/


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## 71 dB

amirm said:


> Sure, they should chase the money to put food on the table.  But then don't come here and tell us that folks shouldn't charge more than 10 cents for a DAC or else we are being "ripped off."  I got ripped off to the tune of $15 for that Ani DiFranco album.  But no, let's complain about audiophiles.  They are fools for buying expensive audio gears but somehow are not for buying over compressed music???
> 
> That is why we have this discussion.  And the same one I was having with that mastering engineering bragging about how he tries to do good for his clients:



I wonder what I am doing here. What a waste of time


----------



## bigshot

It certainly is easy to see who's actually recorded music and who just thinks about doing it!


----------



## Don Hills

bigshot said:


> ...  Red Hot Chili Peppers' Californication has some amazingly distorted stuff on it, and it doesn't sound deliberate. I'm told that they recorded it that way.



They did indeed. There's a lesson in that album that seems to have been forgotten by many artists, though. It was composed and performed with that extreme compression in mind. Everything exists in its own space so when it gets crushed together the various parts don't stomp on each other. Putting the parts in their own spaces can be done to a certain extent during mixing, but it's most effective if it was performed that way.


----------



## RRod

gregorio said:


> I'm certainly not saying all concert halls. I was quoting amirm and his figures for the Dallas Symphony hall. I've actually worked at that concert hall and spent quite a few enlightening days with the hall's designer (Russel Johnson) back in the early 1990's when Birmingham Symphony Hall (UK) opened, which he also designed and to the same principles as the Dallas hall. If you turn the HVAC off and you're the only person in there (as I have done) they are amazingly quiet, as they're full suspended double shell constructions with no expense spared on acoustics and isolation. However, they are outliers! Two of the best symphony halls on the planet and certainly two of the quietest, which is presumably why amirm chose it! The great old concert halls such as the Vienna Musicverein and Concertgebouw, which I've also been lucky enough to work in, have lovely acoustics but their noise isolation is nowhere near that of the Dallas and Birmingham and of course we've got other famous concert halls such as the Royal Albert Hall, which I've worked in extensively, which has neither good isolation nor good acoustics but it does have a great atmosphere with a full house!
> 
> So yes, in general I would agree with you but amirm certainly does like his outliers!!
> 
> G



This is Meyerson? If so, there are a whole boatload of recordings done in it. Perhaps a certain party could find one of those recordings and give us a peak measurement with the pot set to where he hears 16-bit errors.


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## Darren G

gregorio said:


> You are doing it AGAIN amirm!! Yes the noise floor of certain concert halls, when they are empty with the HVAC switched off, can be 0dBSPL in the critical band. Yes, peak levels of say an orchestra can exceed 120dBSPL. I'm NOT arguing with those figures, I'm arguing what should, to anyone without a specific agenda, be painfully obvious: How can you possibly equate the two? In an empty concert hall there is no symphony orchestra, so can that non-existent orchestra produce 120dBSPL or in fact any SPL/Dynamic range at all? In "real life", which you like to quote, the noise floor is the hall plus musicians, plus audience, plus HVAC and all that most certainly is not 0dBSPL or anywhere near it!! So what actually is the noise floor and what is it's spectral distribution? I don't know, I can only guess based on experience. As far as I'm aware there are no AES papers with peer reviewed studies on concert hall noise floors including audience, musicians and HVAC. You seem to be saying that as it's not been published by the AES therefore the noise generated by all the musicians, audience and HVAC doesn't exist, that we can't mention it here in the sound science forum and you can completely ignore it and effectively state we need more than 16bit to reproduce with high fidelity the dynamic range of an orchestral symphony concert which was performed with no orchestra and no audience!



I really loved this post.  Actually one live event I was at a noisy neighbor was so loud that we couldn't enjoy the music, but even at a relatively quiet event, there is a lot of noise.  We tune much of the noise out, and focus on the music, but the noise is part of the musical experience.  Listening to a live concert no audience, no wind or hvac, would probably sound unnatural


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## gregorio

amirm said:


> Let's remember that not only was this offered to general population but also given to us as a "high-resolution" album.
> You really don't think they could do better for us as the audience and for $16?  Didn't we get taken to the cleaners here?



I've already explained it to you and you just don't seem to be getting it; yes, you are getting taken to the cleaners, that's why I started this thread! There is no "high-resolution", the only "resolution" you are getting beyond 16/44 is a "ton of garbage". Yes, at one time and for a short time, there was some advantage to 96kHz but that time passed with the advent of upsampling processors and better filters and, 24 bits was useful only for the very large amount of headroom it provides, which is great for recording but pointless for distribution.

The loudness war is a different topic, although obviously, music being crushed into a smaller dynamic range obviates the need for more bits. Now, I don't object to going off topic and discussing mixing/mastering because if people concentrated on what is actually put in the container format rather than the container format itself, then a lot of the silliness would go away! ...



amirm said:


> Oh, I used my ears first.
> Here is the track from 1990s UP UP UP UP album ['Tis of Thee]: ... Delightful.  ...
> [Deferred Gratification] Listen to those drum kicks and tell me they are not distorted to pieces. ...
> [Pacifist's Lament] Here is Ani with  live recording of the track I analyzed before and not subjected to so called "mastering": ... Isn't that delightful?
> So go ahead, use your ears, tell us if you disagree.



Yes, I do disagree! Not with the generalities but at your conclusions and where you lay the blame:

'Tis of Thee: There are a few slightly strange things going on here; the snare for example, which while giving plenty of room for the breathy vocals, the big reverb but overall low level I find a little odd and I'm not convinced with how they've dealt with the ride cymbal either. The guitar is also unusual,  mid rangy, almost like a banjo but with bass. The kick is also unusual, with relatively little attack but a lot of resonance, almost like an orchestral bass drum but probably achieved with standard kick and a lot of compression. Together with the bass guitar we've got a very low end which could do with a bit of taming. None of this is really "wrong" per se, just that some of it could be better IMHO. Artistic choices obviously play a significant role and it's natural (and desirable) that my artistic choices are not the exact same as everyone else's. 

Deferred Gratification: Agreed, this is in a different league, there're a lot of serious problems but the kick distortion is one of the least of them! It sounds quieter and more insipid than the previous track, indicating it's more compressed and has then been loudness normalised, a very good demonstration for why loudness normalisation could help end the loudness war. The snare is much more prominent in this mix, although it's got far less reverb, as does the vocal, presumably to achieve a feel of intimacy but it doesn't work, it just sounds flat and dead. There's a lot of fighting in the track between the vocals and other parts, so it sounds like they've EQ'ed the vocal to help cut through but all they've done is made it sound rather boxy, made even worse by the over applied compression, which presumably was done to cut through the guitar which is EQ differently but also compressed. In the process they've manage to kill the life out of both the vocal and the guitar! The bass guitar is over-done, although the very low freqs (roughly below about 40Hz and down) sound better tamed than on the previous track. The kick has also been poorly handled and not so intelligently compressed as the previous track. ... All mixes are somewhat of a compromise, instruments fight with each other and the vocals, due to overlapping freq ranges and solving this, getting separation, creating width and depth and trying to make the song sound like a coherent whole is always a challenge. But it seems as if they've lurched from one issue to the next, making it worse at each step. It's impossible to say how much damage was actually done by the mastering itself but some/many of the worse problems could only have been done in the mix! It's a real dogs dinner of a mix and if it were me, I'd bin it and start again. A fair bit of what's wrong could even be due to poor recording, in which case the best thing might be to bin the whole lot and start again from scratch! But it certainly appears that some of the problems in the mix have been caused by looking ahead to a loud finished master. And this is the problem which I've tried to explain previously, most of the recent music genres are designed from the ground up to be very heavily compressed, in fact, they don't work nearly as well without it but this type of folk rock is NOT one of those genres and trying to mix and master similarly loudly is a train wreck waiting to happen. For me, this is where we find the real casualties of the loudness war! Maybe I'm doing it a disservice and without the loudness normalisation it would be a lot less dead, still not a great mix though.

Pacifist's Lament: No, it's not delightful! It's generally sort of OK but there's relatively little musical dynamic range, the severe fret noise gets in the way of some of the vocals, there are a lot of spurious instrument and clothes rustling sounds in there, some dodgy reflections in places and an obvious problem at 1:50. We obviously don't have the "fighting" issues to deal with as far as bass, kick, brass, snare, cymbals, etc., because there aren't any. But hey, it's a live acoustic performance so we can somewhat overlook some of it's issues but that makes it a musically apples to oranges comparison!

So yes, I do disagree, I disagree with you throwing all the blame on the mastering, when there's so much wrong with the mix and maybe even the original recording. And, yes, I've used my ears but I've apparently used them quite differently to the way you have used yours. You highlighted the distorted kick, which to be honest is one of the least of that mix's many problems! And this is the point which bigshot is trying to make and to which you seem completely impervious because presumably there is not an AES paper on it!

G


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## gregorio

RRod said:


> This is Meyerson? If so, there are a whole boatload of recordings done in it.



Yes, it was the Meyerson and I'm not surprised a lot of recordings are done there, it's probably much quieter than most orchestral sized commercial studios, plus it's got great acoustics! I remember having to deal with traffic rumble on a recording we'd done at Air Lyndhurst. 



Darren G said:


> [1] We tune much of the noise out, and focus on the music, but the noise is part of the musical experience.  [2] Listening to a live concert no audience, no wind or hvac, would probably sound unnatural



1. Yep, the "we tune much of the noise out and focus on the music" is an important point which I won't dwell on now, except to say that just recording the sound as it would hit the audience's ears is not how we make modern classical recordings and is why we use mic arrays and spot mics rather than a stereo pair direct to disk.
2. It's difficult to know for sure because that only occurs during rehearsals and during rehearsals the musicians perform differently, there's not the adrenaline and pressure of a live audience, plus the musicians will be physically saving themselves for the actual performance. However, there is typically one difference which is not difficult to isolate, with no audience present the acoustics are very different. In most concert halls the presence of hundreds/thousands of human bodies (the audience) does a great job of sound absorption. While they obviously make a fair bit of noise themselves, they do have an effect on the frequency spectrum and overall level of the music being performed but commonly, the most noticeable effect is in terms of reverb, the hall will sound much "wetter" without the audience. Sometimes this effect is so pronounced that the musicians actually have to change the way they play (more legato and sostenuto) to compensate for the drier acoustics when the audience is there.

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## gregorio

Don Hills said:


> There's a lesson in that album that seems to have been forgotten by many artists, though. It was composed and performed with that extreme compression in mind.



That's what I've been trying to explain, all this blame on the mastering engineers is unwarranted. Unless the music is composed, structured/orchestrated and performed that way to start with, then it's always going to sound like crap if you then demand it's mastered to the same loudness as last year's EDM favourite! There really is no easy solution to all this which is acceptable to everyone. I'm assuming it was entirely unwhittingly but cutestudio posted a link to actually a very good article which I'd not seen before. If you haven't already read it, it's worth the time: https://www.soundonsound.com/sound-advice/dynamic-range-loudness-war

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## amirm

gregorio said:


> I've already explained it to you and you just don't seem to be getting it; yes, you are getting taken to the cleaners, that's why I started this thread!


No, you are barking at the wrong tree in this thread.  First, as I have shown over and over again, all of your assumptions in the OP are wrong.  You did no research on your own to determine true dynamic range of live content and threshold of audibility, and never read any research on that topic either. Both of those let you to create a platform to shout about while at the same time damning the work of others who toiled to bring us the real information.

So on that front, the case is closed unless you have some research to put forward which has not yet occurred in 309 pages (i.e. I am not holding my breath).

The second part is that you are part of an industry that given any bit depth available, work hard to use as little of it as possible. That aims to please the stakeholders in your business, but not ours.  As audiophiles, we have cried over and over again about this but you don't want to use the hours and hours spending here complaining about too much fidelity, toward something that could do us some good.  You being critical to your industry, as much as I am do our industry would be the right thing to do.  But then again, you rather not make enemies in your space.  I get that.  But the whole thing smells of being disingenuous.  See what I quoted from Brian Lucey.

Back to your OP, *we are really done. * I have presented data that shows it be the case.  We need to put a fork in it unless some new authoritative data is available which you have yet to present.  

As it is, you have joined the many well meaning "objectivists" who go around spreading their own myths.  If there is one thing I absolutely hate about being an objectivist, is this casual and superficial approach to audio science by the few vocal posts online.  Gives the whole movement a really bad name and deters any moderate subjectivists to want to give us time of day let alone listen and adopt what we have to say.


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## amirm

71 dB said:


> I wonder what I am doing here. What a waste of time


What are you doing here if I may ask?  Dumbing down fidelity?  To what end?  Are we advocating that DAC chip vendors go back to 16 bits?  That they don't support sampling rates > 44.1?

I have asked this question a number of times and no answer is coming back.  

As I have said, you all are fighting a war that occurred nearly 20 years ago.  The battle of SACD and DVD-A ended and with it, should have been this anti-high res attitude.  Today we are not facing a format war.  We can download all of these formats and play them with total ease.  There is no royalty whatsoever.  Even if your DAC doesn't support high sampling rates, the software does the conversion on the fly.

So I will answer why you are here: to say something and to beat down a group of people who you hope are less technical than you.  When that dynamic changes, we get a post like you made.  Life is not so good when we can't throw numbers around and claim they are scientific when we have not shown one shred of reference to indicate them to be so.  Let's substrat this SPL from that SPL and you need 10 bits!  Go and tell your friends!!!  Right...


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## bigshot (Nov 23, 2017)

Hey Amirm. If you want respect from us, you need to offer us some respect. We aren't dumb here. And it isn't a good idea to speculate about our motives, because that will only lead to us to speculate why you are so bloody desperate to be seen as an authority that you are willing to go to disingenuous means to avoid admitting you can be wrong.

We're all willing to forgive and forget and move on here. But if you continue down the road you've been taking here, you'll end up exactly like you have at the other audio forums where you've used this technique. And that won't be productive for advertising your website.


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## amirm

gregorio said:


> You are doing it AGAIN amirm!! Yes the noise floor of certain concert halls, when they are empty with the HVAC switched off, can be 0dBSPL in the critical band. Yes, peak levels of say an orchestra can exceed 120dBSPL. I'm NOT arguing with those figures, I'm arguing what should, to anyone without a specific agenda, be painfully obvious: How can you possibly equate the two? In an empty concert hall there is no symphony orchestra, so can that non-existent orchestra produce 120dBSPL or in fact any SPL/Dynamic range at all? In "real life", which you like to quote, the noise floor is the hall plus musicians, plus audience, plus HVAC and all that most certainly is not 0dBSPL or anywhere near it!!


You are misunderstanding what the goal for high fidelity reproduction at home is.  Our goal *is to provide the absolute best experience we can. * And that experience comes from having noise floor in the *recording channel* that is below what we can hear, and peaks that rival what we could experience in a live event.  We do not degrade that range by saying, "well the guy next to you could be coughing so let's add 50 db to the noise floor."

As you well know, vast majority of classical music is recorded using close mic'ing techniques.  See my post here: https://www.audiosciencereview.com/...-ears-by-tom-nousaine.1669/page-28#post-54591

And picture from that post:







Those microphones will not pick up any audience noise so your case that we must include such is non-sequitur.   It is not the real life scenario.

When I go into my home theater, I am able to hear far more low level detail than I can in a commercial theater.  I can hear the slightest noises and sounds, teleporting me into the experience.  I don't want anyone to create that soundtrack thinking I am sitting in a theater with high ambient noise.

There is absolutely no reason to limit dynamic range of music today.  As I keep saying, the capability is here, has been for a number of years, and it costs peanuts at the source level.

We know the fully dynamic range for best fidelity and there is no reason to back off from it as an edict.  As a personal choice, sure, you can buy any system and any content at any bit depth you want.  But the goal of ultimate fidelity cannot be dumbed down.


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## amirm

bigshot said:


> Hey Amirm. If you want respect from us, you need to offer us some respect. We aren't dumb here. And it isn't a good idea to speculate that our motive is to beat down people who are less technical than us, because that will only lead to us to speculate why you are so bloody desperate to be seen as an authority that you are willing to go to disingenuous means to avoid admitting you can be wrong.


I have not asked you to believe anything *I say*.  I am asking you to believe peer-reviewed authoritative research that absolutely busts the OP points in the first post.  To date, you have not presented anything to counter that.

That is really the difference between us.  I don't drink water without having back up.   You like to repeat what you may have read elsewhere as real information.  That, I don't respect.  You need to do your homework.  You also need to accept authoritative data if you have none of your own to counter it.  I just have no patience when folks who say they are for audio science and the first thing they do is deny audio science presented to them.


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## 71 dB

amirm said:


> What are you doing here if I may ask?  Dumbing down fidelity?  To what end?  Are we advocating that DAC chip vendors go back to 16 bits?  That they don't support sampling rates > 44.1?



Of course not. I don't care how many bits my DAC has as long as it makes the digital-to-analog convertion well. We can have support for "everything", but in the end 16/44.1 is all we _need_ in consumer audio.



amirm said:


> I have asked this question a number of times and no answer is coming back.



I can't give a better answer than that I am affraid. I'm not banning high-res. I am saying it doesn't have benefits over 16/44.1. It's like if you go to a grocery store to buy something that costs $10, that's all cash you need with you. You CAN take $100 with you, but it doesn't help buying the food you want. If anything, there's just more hassle when you get $90 back for the $100 bill and if you are robbed while walking back home, you lose more money. Similarly high-res takes more space on your hard drive and may cause IM distortion due to ultrasonic content, but basically sounds the same as 16/44.1. That is if they don't charge extra for high-res downloads.



amirm said:


> As I have said, you all are fighting a war that occurred nearly 20 years ago.  The battle of SACD and DVD-A ended and with it, should have been this anti-high res attitude.  Today we are not facing a format war.  We can download all of these formats and play them with total ease.  There is no royalty whatsoever.  Even if your DAC doesn't support high sampling rates, the software does the conversion on the fly.



I don't think this is a war. Spreading awareness maybe? High-res plays with the ignorance and intuition of people. More must be better, right? Well, not always. Not in_ this_ case. I like SACDs and I own maybe 50 of them. They provide excellent multichannel recordings of classical music. I suppose I'd like DVD-As too, but interestingly I don't own any. So, I am not "anti-high res". I am just not anti 16/44.1 either!



amirm said:


> So I will answer why you are here: to say something and to beat down a group of people who you hope are less technical than you.  When that dynamic changes, we get a post like you made.  Life is not so good when we can't throw numbers around and claim they are scientific when we have not shown one shred of reference to indicate them to be so.  Let's substrat this SPL from that SPL and you need 10 bits!  Go and tell your friends!!!  Right...



10 bits isn't enough. 12 bits might just barely be enough with level optimation and strong shaped dither. 14 bits is enough and 16 bits is what CD has got, about 2 bits worth of safety margin so we can sleep well at night. All we need is to understand human hearing, digital audio and the (most demanding) listening conditions of consumer audio and coming to this conclusion isn't that hard.


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## bigshot (Nov 23, 2017)

amirm said:


> I have not asked you to believe anything *I say*.



You didn't read what I said. I'm not talking about science. I'm talking about your social skills and the way you present yourself. If you stomp around engaging in ad hominem attacks, refusing to admit when you are wrong, and not listening to what other people say to you except to twist it for your own self justification, you won't get anywhere with anyone. I'm giving you some constructive criticism here on the things you might want to consider and work on. If you continue acting like this, you're going to get squeezed out of this forum too, and you'll have to retreat to your own forum again. You obviously want to lead people from this forum to your own. Acting like this isn't a good way to do that.

I can point out the trouble spots in your posts if you are really not aware of what you're doing. But I think if you can think objectively about how you appear to other people, you can figure it out for yourself.

Also, I'm curious about the photo you posted of the orchestra recording. Why are they wearing headphones? They're not going to get any sort of feeling for their place in the ensemble with cans. Are they listening to playback for overdubbing something? And why is there a speaker on the conductor's podium? That photo makes no sense. By the way, classical music is definitely NOT recorded close miked. I've never recorded an orchestra before, but I can read Gregorio's post and know for sure he has. He knows what he's talking about there.


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## sonitus mirus

bigshot said:


> Also, I'm curious about the photo you posted of the orchestra recording. Why are they wearing headphones? They're not going to get any sort of feeling for their place in the ensemble with cans. Are they listening to playback for overdubbing something? And why is there a speaker on the conductor's podium? That photo makes no sense.



The headset only covers one ear, too.  That is very odd.  Maybe it is a mono recording.


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## Strangelove424 (Nov 23, 2017)

bigshot said:


> Also, I'm curious about the photo you posted of the orchestra recording. Why are they wearing headphones? They're not going to get any sort of feeling for their place in the ensemble with cans. Are they listening to playback for overdubbing something? And why is there a speaker on the conductor's podium? That photo makes no sense.



https://www.soundradix.com/radical-feed/2016/01/24/tom-player/

They're recording a film score. Headphones might be for syncing purposes. Maybe for dialogue, but perhaps for different orchestra sections too. I can't find percussion, so maybe they are syncing to that. Film scores have completely different demands than typical classical performances. All those mics, and their proximity to the players, are possibly for track mixing, which as far as I know is either not done at all, or not aggressively for typical live classical performances. There's also the matter of ambient noise on soundtracks tending to be rather low. The music, or the the recreation of the music in the context of its original performance is not important for a film score. It is there to serve the story. When the film gets silent, the soundtrack must too. You won't find much ambient noise in film scores for that reason, and the micing techniques reflect that. A soundtrack is like classical music walking into a giant recording booth. It's very contrived.


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## bigshot (Nov 23, 2017)

That makes sense. I guess those microphones won't pick up any audience noise because there is no audience. No hall ambience either. That's a recording studio. You wouldn't record classical music like that. I would love to visit the Vienna Musikverein or even the new opera house in Valencia. There's nothing like concert hall music in a concert hall the way it was intended to be heard. The great thing about hearing music that way isn't the extremes... no super high frequencies or stone dead room tones. The thing that makes it incredible is the liveness of the room. Sound is all around you bouncing and delaying and decaying in a beautiful way. Multichannel is the first I've ever experienced that in recorded music.


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## Strangelove424

Amen to that. Nothing beats a well designed concert hall. I haven't been to many, but the best I've seen is the Disney Hall in LA. The acoustics are surreal. The architecture too. It'd be nice to tour the world and visit halls around the globe.


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## bigshot

My favorite in LA is the Ambassador Auditorium, but it isn't used much since the quack religion that built it went bust.


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## castleofargh

we can go on and on about this so long as we fail to even settle down on what we're trying to achieve, we will obviously never reach an agreement. it's the same issue with Keith, we look for something different and then argue that we have different requirements and desires... well duh. 
a good deal of @amirm's limits are real within specific testing conditions, so I don't think he should be treated as if he was wrong or saying false stuff. again his point of view is to define limits so that *all possible conditions* will be met with audible transparency. there is nothing wrong with that, just like there is nothing wrong with using even higher resolution for those who wish to get beyond audible limits for the love of numbers. 

now we also can't possibly jump to the other side and assume that those extreme thresholds are a necessity for transparency in music. because that of course is false and easy to debunk with a 16/44 vs 24/192 of your favorite song on any DAC reasonably clean at 44.1. music isn't in general made of extremes, and even less made of all extremes at the same time. people don't usually listen to music in ideally extreme conditions in the first place. or apply extreme concentration throughout the album. so no we don't need Amirm's specs to achieve transparency on our library under our typical listening conditions. different conditions, different needs.  if I was passing 16/44 vs highres tests on a regular basis, I would be all about setting new criteria for my listening format of course. but it just doesn't happen which in itself is evidence that I at least don't need more than 16/44 for audible transparency. I feel that the dichotomy is quite clear and we don't have to kitty fight over what's necessary in practice. for each condition, we can all test it ourselves and find out if indeed 16/44 is already providing for our audio needs of transparency.


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## bigshot

I have no problem with Amirm's insistence on focusing on the extremes. My objection is his dismissal of people who put his facts in context. If he says that in the most extreme case, a dynamic range of 120dB might be necessary to reproduce the full range of what a human can hear, I'll agree with him. 120dB is the threshold of pain, so it's the loudest we can hear. But I'm going to point out that for the purposes of listening to recorded music in the home, 45 to 50dB is all you really need. We'd both be right and we would agree with each other. But that isn't happening because he is trotting out the most annoying of audiophile tactics- trying to pull rank and belittling people who don't follow his line of reasoning unquestioningly. Amirm is speaking here with people that have a wide range of experience on this topic. He could learn from us, and we could learn from him. We should all treat each other as peers. But he isn't allowing that. I think I pretty clearly pointed him to what his problem is. It isn't his facts. It's his attitude.


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## JaeYoon

bigshot said:


> I have no problem with Amirm's insistence on focusing on the extremes. My objection is his dismissal of people who put his facts in context. If he says that in the most extreme case, a dynamic range of 120dB might be necessary to reproduce the full range of what a human can hear, I'll agree with him. 120dB is the threshold of pain, so it's the loudest we can hear. But I'm going to point out that for the purposes of listening to recorded music in the home, 45 to 50dB is all you really need. We'd both be right and we would agree with each other. But that isn't happening because he is trotting out the most annoying of audiophile tactics- trying to pull rank and belittling people who don't follow his line of reasoning unquestioningly. Amirm is speaking here with people that have a wide range of experience on this topic. He could learn from us, and we could learn from him. We should all treat each other as peers. But he isn't allowing that. I think I pretty clearly pointed him to what his problem is. It isn't his facts. It's his attitude.


120 db is also a great way to develop tinnitus or lose your precious hearing ability real quick.

But I do like how Amirm and keith are putting their ideas and sharing information. I don't mind it.


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## jgazal (Nov 25, 2017)

A thread regarding a very specific topic (bit depth) has the highest number of replies in the sound science forum. Thus I would like to kindly ask here if anybody could check for errors in the following (unfortunately long) post: *A layman multimedia guide to Immersive Sound for the technically minded*

I would be very grateful.


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## reginalb

amirm said:


> I have not asked you to believe anything *I say*.  I am asking you to believe peer-reviewed authoritative research that absolutely busts the OP points in the first post.  To date, you have not presented anything to counter that.
> 
> That is really the difference between us.  I don't drink water without having back up.   You like to repeat what you may have read elsewhere as real information.  That, I don't respect.  You need to do your homework.  You also need to accept authoritative data if you have none of your own to counter it.  I just have no patience when folks who say they are for audio science and the first thing they do is deny audio science presented to them.



Except when there are legitimate and pointed criticisms to your "authoritative data," you respond with "Look at these 63 references" and then highlight the biography of the author. Both of these are completely meaningless and demonstrate your inability to actually defend the content of these studies you're referencing. It also suggests to me (who isn't about to pay to access the AES papers you've cited) that you might not have a firm enough understanding of the studies to be a good source of information in interpreting the data contained within, and thus I question the articles over on your web site. 

It's very common for people to not really know how to read and interpret research - Just read about any report in any major newspaper - including the big boys like the New York Times and Wall Street Journal about a medical research study. I actually took a class in college that was on understanding research with an assignment every semester wherein the professor would pull a new and current story from a major newspaper (ours was the NYT) about some research where the conclusions described by the story's writer was incorrect, and we had to contrast the popular news report with the actual writeup on the study. He taught that class for something like 10 years, and every semester over that period was able to pull an article from within the previous week of when he assigned it. Because these papers were _always _misunderstanding research. 

You clearly know more than I do about audio, but your posts make me less sure about your interpretations of research. Someone with a good grasp of the data you're claiming to have wouldn't need to repeatedly resort to ad hominem attacks, appeals to authority, the importance of certain aspects of the scientific method, and even going so far as using the number of citations in a paper to prove that it must be right (in response to a legitimate and direct critique). I know I'm beating a dead horse with that but I just can't get over it. You would be able to address the actual content of the critiques.


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## sonitus mirus

The way I understand the AES report to demonstrate is that someone took a super-duper mic and toured around the San Francisco music scene and captured a few events that had peak SPLs above 120 dB.  This was, of course, the level recorded with a crowd and any gimmicks that might be included in the concert.  Even if someone wanted to recreate as close as possible to being at one of those events, with modern dithering, 16 bits should still leave plenty of dynamic range, even if your listening room is as noisy as Bimbo's 365 or as crowded as the Filmore on a Saturday night. 

I'm still not certain how the reference disproves much of what was stated in the OP.  There was a bit of a side track with contesting the claim that 120 dB was necessary for DR, and we now see that there is at least one paper that supports the notion, but it doesn't appear to have any practical importance to the discussion.


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## WoodyLuvr

amirm said:


> I have not asked you to believe anything *I say*.  I am asking you to believe peer-reviewed authoritative research that absolutely busts the OP points in the first post.  To date, you have not presented anything to counter that.
> 
> That is really the difference between us.  I don't drink water without having back up.   You like to repeat what you may have read elsewhere as real information.  That, I don't respect.  You need to do your homework.  You also need to accept authoritative data if you have none of your own to counter it.  I just have no patience when folks who say they are for audio science and the first thing they do is deny audio science presented to them.


Extremism at it's finest.  Be careful @amirm as of late many of your posts are beginning to paint an ugly picture of yourself which I believe may be totally inaccurate of who you really are as a person.  Please don't let frustration cloud better judgement.  Remain civil as many of us really do want to see/read your input and thought provoking posts... you have provided very strong arguments (sometimes with enlightening information, especially for some of us audio dummies) however your style of presentation has regrettably rubbed many the wrong way and does come across a bit aggressive and lofty. Respects.


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## 71 dB

sonitus mirus said:


> Even if someone wanted to recreate as close as possible to being at one of those events, with modern dithering, 16 bits should still leave plenty of dynamic range, even if your listening room is as noisy as Bimbo's 365 or as crowded as the Filmore on a Saturday night.



- You are not supposed to recreate 120 dB levels in home listening, even if you live in the middle of a forest miles from other people.
- Amplifiers don't have large enough S/N ratio to allow real 120 dB dynamics. Your amp makes more noise than your CD!
- It's ridiculous to think you could hear the dither in a concert recording. The crowd alone completely masks it even when completely silent.

120 dB is huge. It's twice what the best vinyls give. Image a vinyl with very little noise playing the loudest sounds it can. Now image attenuating that loudest sound to the level of vinyl noise and how quiet the original noise would become. Anyone who has attenuated signal in a sound editor undertstands how much 120 dB is. Even 60 dB attenuation makes a loud sound almost inaudible. 120 dB is making such attenuation twice!

The dynamic range of human eye is about 140 dB from the darkest visible thing (when eyes have completely adjusted to darkness) to the brightest. How much do you think your TV uses of this? How much do you get in a movie theater? I think it's something like 30 dB, perhaps even 40 dB if you have OLED.  Image a TV picture with 100 dB of dynamic range. The darkest scenes would require eyes beeing adjusted to darkness and the brightest would be piercingly bright. In movies you cut between bright and dark picture too fast for this to make any sense. Same in music. You may experience 120 dB dynamic range of SPL during a day, but within a song or concert? Don't think so!


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## gregorio (Nov 24, 2017)

amirm said:


> [1] Our goal is to provide the absolute best experience we can. And that experience comes from having noise floor in the *recording channel* that is below what we can hear, and peaks that rival what we could experience in a live event.
> [2] As you well know, vast majority of classical music is recorded using close mic'ing techniques.
> [2a] Those microphones will not pick up any audience noise so your case that we must include such is non-sequitur.   It is not the real life scenario.  And picture from that post:
> 
> ...



1. Our goal is to create the best experience we can BUT the best experience is *ABSOLUTELY NOT* a 0dB noise floor and peaks of 120dB!!! HOW ON EARTH can a potentially dangerous/damaging dynamic range be "the best experience"? In fact it would be pretty much the EXACT OPPOSITE of "the best experience"! It should be more than obvious to anyone that the "best experience" is an experience which falls within the range of "comfortable", which is nowhere near the potentially dangerous range. I know some people get their entertainment from engaging in extreme sports but listening to a symphony concert is NOT supposed to be "engaging in an extreme sport"! You wanted published scientific papers and I provided you with one, which shows that orchestral musicians suffer hearing damage from their work and some have to wear hearing protection when playing in an orchestra but you've ignored/blanked that scientific evidence completely, a sad and hypocritical position to take as you are accusing everyone else of ignoring the scientific evidence.

2. I do know well, not only from working with many great orchestras in many great venues and recording studios as an engineer and with some of the world's best engineers but also from actually being an orchestral musician from my early teens, for a about 15 years before I became an engineer! It's for this reason that I absolutely KNOW for CERTAIN that you are talking absolute NONSENSE, orchestras are NEVER recorded using close mic techniques! ... as evidenced by ...
2a. Oh, the irony!! That's a photo of Air Lyndhurst studios, definitely a world class studio and I know it's Air Lyndhust because I recognise it from actually having worked there, I even mentioned a few pages ago that I had to remove some traffic rumble from a recording made there!!
So, let's have a proper look at that photo you posted:

Firstly and most obviously: Sitting beneath the mics are what appear to be living, breathing, musicians with musical instruments, they do NOT produce 0dB of noise. To produce 0dB of noise they would all have to be dead but then of course, you'd need a heater and some incredibly strong smelling salts to get anywhere near 120dB SPL out of them!!
Secondly: While Air Lyndhurst has lovely recording acoustics, great equipment and world class engineers, it's actually a converted church in London and even empty, it doesn't have a 0dB noise floor!
Thirdly: You apparently do not know what "close mic'ing" means. With close mic'ing, we're talking about distances between the sound source/instrument and the mic of inches. Maybe less than an inch in the case of mic'ing a snare drum, 2 or 3 inches when mic'ing the cab of an electric guitar, maybe 6-18 inches in the case of a pop/rock singer. Close mic'ing almost invariably means less than about 20 inches (0.5m) and of all the mics in your photo, not a single one of them is "close"!
Fourthly: Note the metal crucifix type structure above the conductor with the three mics attached. That's called a Decca Tree and has been a preferred orchestral recording technique since it was invented in the 1950s. It is specifically designed NOT to be a close mic technique, it's typically placed at least 10 feet (3-4m) above the conductor at a distance where the direct and reflected sound balance. The Decca Tree was designed for use with 3 Neumann M50 mics, due to their frequency and polar response and indeed, this is exactly what Air Lyndhurst uses. M50s are incredibly expensive and sort after, good original condition ones can go for tens of thousands, they have a max SPL of about 114dBSPL, a published "usable" freq response from 40Hz to 14kHz and although I can't remember exactly, a signal to noise ratio somewhere around the low to mid 70'sdB. They are also Omni-directional, so they pick up sound/noise equally from ALL directions!
Lastly: Notice there are outrigger mics and room mics placed so much further from the orchestra than the Decca Tree that they've been all but cropped out of your photo!! When you say "close mic'ing", do you mean "close" in comparison to a mic in a different city?

So from the photo you yourself have quoted: You do NOT have a 0dB noise floor recording environment, you do NOT have musicians making 0dB of noise, you do NOT have close mic'ing techniques (to reduce reflections and environmental noise) but you DO have mics with a 114dB max SPL, which is half of the 120dB you seem to desire and a signal-to-noise ratio of seventy something dB, also nowhere near the 120dB dynamic range you seem to be ridiculously after! And, even if all that were possible in the real world, absolutely no one would make such a product as no one, apart from apparently a few audiophile nutters, want to be deafened in the loud parts, hearing nothing but silence in the quiet parts and constantly be manually compressing the recording by turning the volume knob up and down all the time to keep it in the comfortable range!

3. You're joking right? When we're in the dubbing theatre we're mixing films for commercial cinemas/cinema audiences. You think maybe I should call Christopher Nolan and tell him he's doing it all wrong and should instead be making his next film mix for amirm and his home theater, at the expense of everyone else?

Honestly Amirm, for a supposedly intelligent man you really are saying/doing some dumb things. You obviously know next to nothing about recording and yet fight from this position of near complete ignorance with those who actually do it for a living and dumber still, use as "proof" a studio I'm well acquainted with personally!  No one is disputing your quoted scientific data or that's it's useful *IN CONTEXT*, what we're disputing is your blinkered determination to erroneously apply it completely out of context, as "real life". Just repeat posting the same nonsense about "real life" and ignoring all the responses explaining why it is not "real life" is not going to win you anything, it's just going to make you look ignorant and foolish, which would be a shame for someone who should have enough intelligence to avoid being either!! I think we've reached the point of no further progression with you, because you apparently can't even appreciate/understand that the evidence you're posting not only doesn't support your assertions, it actively contradicts them!

G


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## bigshot (Nov 24, 2017)

71, the dB scale is logarithmic, so the difference between 60dB and 120dB is a LOT more than double. And most LPs have dynamic ranges under 50dB... a lot of them are quite a bit lower than that.


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## Strangelove424

It doubles every 10db, so if the coffee has kicked in and my math is right it should be 64x louder. I think it's also interesting to point out that according to the Sound Radix promo piece, the composer pictured (Tom Player) uses loads of DSP including compression.


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## gregorio

Strangelove424 said:


> It doubles every 10db, so if the coffee has kicked in and my math is right it should be 64x louder.



There really is no direct correlation between dB and loudness. If we're talking about dB SPL (Sound Pressure Level) and we are, then +6dB is double, +20dB is 10 times, 40dB is 100 times and the difference between 60dB and 120dB is 60dB, which is 1,000 times more SPL!

G


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## amirm (Nov 24, 2017)

reginalb said:


> Except when there are legitimate and pointed criticisms to your "authoritative data," you respond with "Look at these 63 references" and then highlight the biography of the author. Both of these are completely meaningless and demonstrate your inability to actually defend the content of these studies you're referencing. It also suggests to me (who isn't about to pay to access the AES papers you've cited) that you might not have a firm enough understanding of the studies to be a good source of information in interpreting the data contained within, and thus I question the articles over on your web site.


What inability?  That the few people on the losing side of the argument declaring it so, doesn't make it that.  I mean the proof is in what you say: _none of you have bothered to go and buy the papers and read them! _ How little can you care about this topic when you have no motivation to go and spend the cost of one or two CDs to really understand the research?

So that it is clear what the research states, *it goes thought *every bit of the chain* in both recording and playback to see what the effective dynamic range could be.  I*t examines everything from venue, to hearing thresholds, microphones, amps, DACs, ADCs, etc., etc. And it builds on other published work.  Furthermore, this research is hugely references in later papers by many others.  It simply is end of the story.

As to whether I know how to interpret the data, this is my professional background: https://www.audiosciencereview.com/forum/index.php?threads/a-bit-about-your-host.1906/.  Bigshot walks around with an AES workshop link in his signature.  Did you know I hired one of those luminaries, J-J, to be my audio architect while at Microsoft?  This is what I did professionally.  I have the ability to understand the paper in question.  And have written articles on it that has gotten wide distribution without a single person writing in and saying, "oh, your author doesn't have the ability to understand the resaerch."  Now this may sound like bragging but it is not.   Your doctor is not bragging when he says that he understands research that you have not even read.  He can dismiss your protest out of hand and be right.

As to what you all are saying, I* have heard it countless times. * I have debated them at length on other forums with people taking your positions but frankly, doing a lot more research than you all that are doing by protesting.  *What is in the OP is one of the most frequent myths spread on forums about dynamic range of our hearing and what is recorded in content. * It is time to stop it and not use debating tactics of "oh, don't you want our respect" or "you don't get it."  I get it and am not looking for the respect from people who don't bother spending a few bucks learning about audio science as published in real world and not talking points in forums.  Yes the "logic" of it makes lay sense.  It even makes semi-technical sense.  That is why people run with it without doing any research of their own.  But ultimately it is just gut feeling stuff that is not correct.

Yes, we can push fidelity of audio way, way down.  An MP3 at 128 kbps will be transparent to vast majority of public and many audiophiles for that matter.  Better  yet, at highest level, a lossy codec can fool even the best of the audiophiles.  So the point you are trying to make is not in dispute in that sense.

What you are not considering is what I have said repeatedly: *a channel needs to be transparent for all people and all content.*  That is my standard of reference.  It is something we can achieve today.  It takes content that is not butchered and hardware implementations that are right.

While you may not be able to hear lossy codec degradations a few of us can.  I have post countless blind listening test showing this.  I have also done it with high-res vs CD rate.   So when you walk around and say things like the OP, you better indicate that your paper napkin math excludes critical listeners, some content and some equipment.  In that case, you better say lossy compression is good enough, because that would be true of that too.

In summary, I am very aware of your position and knowledge.  What you are not aware of is my position and knowledge.  If you want to get close to what I know, you need to start reading and understanding incredible body of research that is published that never makes it to forums.


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## amirm

gregorio said:


> There really is no direct correlation between dB and loudness. If we're talking about dB SPL (Sound Pressure Level) and we are, then +6dB is double, +20dB is 10 times, 40dB is 100 times and the difference between 60dB and 120dB is 60dB, which is 1,000 times more SPL!
> 
> G


That is true for the same reason that your OP is wrong!  We need to look at psychoacoustics to understand what these numbers mean.  For that, we look at Fletcher-Munson graphs which correlates loudness with SPL.







We see that our hearing is quite nonlinear.  As sound gets louder, the shape of the low frequencies (for the same level of loudness) changes (flattens).  For that reason, a 10 db increase in bass frequencies doesn't have the same effect as 10 db increase in mid-frequencies.

Now if you exclude low frequencies, the rule of thumb can apply as you see from the phon labels written on the graphs.


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## castleofargh

bigshot said:


> 71, the dB scale is logarithmic, so the difference between 60dB and 120dB is a LOT more than double. And most LPs have dynamic ranges under 50dB... a lot of them are quite a bit lower than that.


I was tempted to make the same remark as a joke. because I'll go on a limb here and bet that 71dB knows how dB work


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## amirm

WoodyLuvr said:


> Extremism at it's finest.  Be careful @amirm as of late many of your posts are beginning to paint an ugly picture of yourself which I believe may be totally inaccurate of who you really are as a person.  Please don't let frustration cloud better judgement.  Remain civil as many of us really do want to see/read your input and thought provoking posts... you have provided very strong arguments (sometimes with enlightening information, especially for some of us audio dummies) however your style of presentation has regrettably rubbed many the wrong way and does come across a bit aggressive and lofty. Respects.


Point well taken.  Thank you for taking the time to write it.


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## sonitus mirus

I found conflicting data with the link to noise exposure as provided earlier from OSHA.

This is what I have found from the World Health Organization.

http://www.who.int/occupational_health/publications/occupnoise/en/

From this document on page 95:

http://www.who.int/entity/occupational_health/publications/noise.pdf?ua=1


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## castleofargh

guys. how do you want this to end? because the way it is, the ending will be bad. I've already taken someone out yesterday and I end up having to delete more and more posts those days. not a great trend. headfi has clear rules about how to address each other, we all agreed to follow them when we registered. we're all a little more open to arguments in this section because trying to disprove things is a core part of science, but at no point do we have to turn every difference in opinion into a bar fight where someone is a liar or an idiot.
personal attacks bring nothing to the conversation, they have never helped convince anybody that he was wrong(in fact there is a proven and opposite effect), and once again, HeadFi forbids it!!!!!
I don't know it works for you guys, chew on something. proof read before posting.  don't post when you're mad at someone. reduce caffeine, etc. anything to limit and ideally stop judging people and attacking them. I'm always really trying to do things as if we were all reasonable people with some matter of self control, but if talking doesn't work, I'll have to become a dick and start treating you all like kids with no carrot and a lot of sticks.


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## JaeYoon (Nov 24, 2017)

castleofargh said:


> guys. how do you want this to end? because the way it is, the ending will be bad. I've already taken someone out yesterday and I end up having to delete more and more posts those days. not a great trend. headfi has clear rules about how to address each other, we all agreed to follow them when we registered. we're all a little more open to arguments in this section because trying to disprove things is a core part of science, but at no point do we have to turn every difference in opinion into a bar fight where someone is a liar or an idiot.
> personal attacks bring nothing to the conversation, they have never helped convince anybody that he was wrong(in fact there is a proven and opposite effect), and once again, HeadFi forbids it!!!!!
> I don't know it works for you guys, chew on something. proof read before posting.  don't post when you're mad at someone. reduce caffeine, etc. anything to limit and ideally stop judging people and attacking them. I'm always really trying to do things as if we were all reasonable people with some matter of self control, but if talking doesn't work, I'll have to become a dick and start treating you all like kids with no carrot and a lot of sticks.


:c but I like carrots! don't take them away!


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## amirm

sonitus mirus said:


> I found conflicting data with the link to noise exposure as provided earlier from OSHA.
> 
> This is what I have found from the World Health Organization.
> 
> ...


Again, none of this data can be used in this discussion.  Please see this footnote for reasons why in your above graph:





As you see, it is a slow reading meter which means it is not peak SPL numbers.  And further, A-weighted which is not what we store in your music files.  We store actual/peak values of audio samples, not any kind of weighted or averaged values.  As such, the channel must deal with the peak numbers as the highest value which as a rule are 5 to 10 db higher than average numbers indicate.


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## bigshot

It doesn’t take a lot of people to make trouble. It only takes one or two. But to build a good community to share information, it takes a lot of people who see themselves as part of a group.


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## sonitus mirus

amirm said:


> Again, none of this data can be used in this discussion.  Please see this footnote for reasons why in your above graph:
> 
> 
> 
> As you see, it is a slow reading meter which means it is not peak SPL numbers.  And further, A-weighted which is not what we store in your music files.  We store actual/peak values of audio samples, not any kind of weighted or averaged values.  As such, the channel must deal with the peak numbers as the highest value which as a rule are 5 to 10 db higher than average numbers indicate.



How would anything you just posted change the fact that your list showed a maximum exposure of 12.5% of an hour at 120 dB?  A peak SPL makes no sense for an exposure at over 7 minutes.  How can any peak last 7+ minutes that is not continuous?


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## amirm

gregorio said:


> 1. Exactly, hallelujah brother! Notice I've highlighted the* "could be*". If the orchestra are dead and make zero noise, if we're in a venue with zero noise, if we take the damage thresholds of human hearing, if we only use the mic with the largest dynamic range and "close mic" with them, use zero noise amps/mic pre-amps, etc., then YES, WE *COULD* achieve the numbers you are talking about and we would all be wrong about everything. HOWEVER, ...


As I noted, in my last post, and have said over and over again, you cannot use peak SPL numbers with regards to what causes hearing damage.  Hearing damage metrics use averages and importantly include a duration which is orders of magnitude longer than what is in dynamic music.

Now, if you sit there in mastering and boost the hell out of average loudness level and makes them just a hair less than peaks, then sure, your ears will bleed and complain.  The solution to that is not to advocate for less bits but to fix mastering.

So there is no "HOWEVER" here.  After so many posts we still see the same misunderstandings about SPL numbers repeated.  

Even if we go along with what you are saying, it is dead against your original post that started this thread:



gregorio said:


> *I know that some people are going to say this is all rubbish, and that “I can easily hear the difference between a 16bit commercial recording and a 24bit Hi-Rez version”. Unfortunately, you can't, it's not that you don't have the equipment or the ears, it is not humanly possible in theory or in practice under any conditions!!*



That is wrong, isn't it?  You are not only allowing the theory to show that, but also in practice in some cases.  Nothing pointing to impossibility.

Are we going to see an edit to that first post that evidence to the contrary has been provided?


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## gregorio (Nov 24, 2017)

sonitus mirus said:


> I found conflicting data with the link to noise exposure as provided earlier from OSHA.



Noise exposure levels are designed for industrial purposes, where the noise is fairly constant, say in a factory, and even for that they're not very good and some countries have laws lower than the OSHA recommendations. None of this is really applicable to music listening except obviously that any duration of 140dBSPL is dangerous. What we do have that is relevant to this line of discussion is some studies which show hearing damage of orchestral musicians, one of which I posted a few pages back but has been singularly ignored by certain parties with an agenda. Obviously, listening to orchestral music at the actual real life levels that an orchestra produces is potentially dangerous, much less so for an audience who get lower peaks SPLs than those close to/in the orchestra, far more compressed music genres would likely be more dangerous still at those same peak levels, as there are likely to be far more of them! For anyone out there, even if you have the equipment and inclination to do so, *do not* listen to music at 120dB SPL peak levels!!!!!

G


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## amirm

sonitus mirus said:


> How would anything you just posted change the fact that your list showed a maximum exposure of 12.5% of an hour at 120 dB? A peak SPL makes no sense for an exposure at over 7 minutes. How can any peak last 7+ minutes that is not continuous?


It shouldn't make sense because that is not what I said!    I have repeatedly said that the exposure metrics are average SPLs.  But that aside, I also wanted to indicate that there is a duration and hence that chart.  There are different charts by the way with different numbers so if that is what you are observing, that is fine.  It is just not applicable to discussion at hand.


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## amirm

gregorio said:


> For anyone out there, even if you have the equipment to do so, *do not* listen to music at 120dB SPL peak levels!!!!!


You say that based on what peak SPL measurement or study?


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## sonitus mirus

amirm said:


> It shouldn't make sense because that is not what I said!    I have repeatedly said that the exposure metrics are average SPLs.  But that aside, I also wanted to indicate that there is a duration and hence that chart.  There are different charts by the way with different numbers so if that is what you are observing, that is fine.  It is just not applicable to discussion at hand.


Ah, ok.  Thanks.  I admit, I was confused.


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## amirm

gregorio said:


> 1. Our goal is to create the best experience we can BUT the best experience is *ABSOLUTELY NOT* a 0dB noise floor and peaks of 120dB!!! HOW ON EARTH can a potentially dangerous/damaging dynamic range be "the best experience"?


Once again, all of our impressions of what is "dangerous" is reading SPL charts online and not realizing they are a) averages b) weighted c) for wideband noise and not tones in music and d) they are rough figures and not based on any specific research.  Importantly unless someone here has actually used a peak measuring technique -- which I assume no one has --  you are most likely exposed to such "dangerous" SPLs and actually liked it!  I know I have.   

In my entire library of research papers, I think I only have 3-4 papers that use peak SPL measurements.  Every other place such data is averaged and hence not appropriate for this conversation.  

On the noise floor, again, we cannot speak of single value SPL numbers.  We must, must look at the spectrum.  SPL numbers of 40+ db can be dead silent to us as a result!  As I show in my article on quietness of our listening rooms based on Fielder's reference room, people have rooms that are this quiet:







The solid line is threshold of hearing.  As we see, at 20 Hz it can be as high as 65 db and we still consider that dead silent!!!

The best measured room easily qualifies for such.  This research is from 1990s.  Today, there are many state of the art rooms built at far lower noise levels.  So if survey was done today we would uncover many such rooms.  And regardless, as long as these rooms can be built, then we owe it to those people to use a distribution format that delivers noise free experience to them.

As an aside, notice how the threshold of hearing at mid-frequencies where we are most sensitive is actually a negative SPL!  Not zero.  Our hearing is quite sensitive when it comes to that region (likely for the need to understand each other's voices).

Furthermore as I have explained, noise coming from speakers is point source and is more audible than ambiance which is diffused.  As such, it is not sufficient for it to be the same as ambient noise to be inaudible.  It needs to be lower.


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## 71 dB

bigshot said:


> 71, the dB scale is logarithmic, so the difference between 60dB and 120dB is a LOT more than double. And most LPs have dynamic ranges under 50dB... a lot of them are quite a bit lower than that.



Good call bigshot, but let me assure you logarithmic dB scale is very familiar to me. I mean double on_ logaritmic_ scale, 2 x 60 = 120. On linear scale 120 dB is of course a million* times more acoustic energy than 60 dB. Sorry, if I assumed too much other readers to know dB scale. I was trying to say that if you attenuate the loudest sounds of a vinyl 60 dB so that they became as quiet as the noise floor, it's easy to understand that the noise floor becames insanely quiet when attenuated 60 dB too and that's roughly the noise floor of CD with shaped dither. The noise floor of vinyl is in the "midway" of CD noise floor and 0 dBFS sounds on logarithmic scale.

Vinyl dynamic range being 60 dB is the best case scenario. Sometimes vinyl is given even 70 dB, but I don't believe that.


* 10^(60/10) = 10^6 = 1.000.000


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## 71 dB

castleofargh said:


> I was tempted to make the same remark as a joke. because I'll go on a limb here and bet that 71dB knows how dB work



That's a good bet man! The day I don't know how dB scale works is the day to die, because I have become too senile to live anymore.


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## 71 dB

amirm said:


> Again, none of this data can be used in this discussion.  Please see this footnote for reasons why in your above graph:
> 
> 
> 
> As you see, it is a slow reading meter which means it is not peak SPL numbers.  And further, A-weighted which is not what we store in your music files.  We store actual/peak values of audio samples, not any kind of weighted or averaged values.  As such, the channel must deal with the peak numbers as the highest value which as a rule are 5 to 10 db higher than average numbers indicate.



This is a good remark. Also, people want to listen to music at _pleasing_ levels, not at the highest possible levels our ears can possible take. A pleasing level is of course subjective. Teenagers want as loud as possible, 160 dB of bass that makes their first car pulse of the EDM beats! Old people want their evergreens softer because their ears hurt easily.


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## bigshot (Nov 24, 2017)

Listening to music with 120dB peaks probably won't damage your hearing, but it won't be comfortable. If there is a snare drum that hits 120dB with each hit, you'll be wincing with each one, and the music behind it will be so loud, you won't be able to hear anyone in the room speak to you, even if they are sitting right next to you. It's a great idea in theory to be able to reproduce 120dB, but it isn't a good idea to actually listen to music like that. If you doubt that, get a nice SPL level meter, set it properly and try it yourself. Listen to a whole album that way. I bet you don't even get through 30 seconds.


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## 71 dB

amirm said:


> As an aside, notice how the threshold of hearing at mid-frequencies where we are most sensitive is actually a negative SPL!  Not zero.  Our hearing is quite sensitive when it comes to that region (likely for the need to understand each other's voices).



Yes, between 3 and 4 kHz we have the ear canal resonance at 1/4 of wavelength. Due to this resonance, sound pressure levels below 0 dB are audible, but it work in the upper end too: The threshold of pain is lowered too for the same amount and goes under 120 dB at this frequency range. The dynamic range of hearing is pretty constant 120 dB between 800 and 4000 Hz and elsewhere it is less, for example at 100 Hz only about 80 dB!


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## RRod

Here's an example reverb tail from a non-live recording made at the symphony hall previously discussed. I've put random names on these test files (24/44100), but rest assured one of them is the original > 16-bit source. Once y'all have found a minimal level where you can hear some differences, I'll put up the loudest part of the album. For my own part, in my 35dBA bedroom, listening on my PM-3s (which isolate well >1kHz), with the volume cranked as loud as I care to for this album, I can maybe hear just a bit of difference in one file; whether that holds up in ABX I haven't tried.


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## old tech

71 dB said:


> Gnds on logarithmic scale.
> Vinyl dynamic range being 60 dB is the best case scenario. Sometimes vinyl is given even 70 dB, but I don't believe that.
> * 10^(60/10) = 10^6 = 1.000.000


70db would be a best case scenario and within the realms of possibility.  Depending on the quality of the pressing, the recording, the position of the groove on the record and the frequency, 70db is possible.  That of course also assumes that the T/T & cart is able to track that specific "sweet spot" with some degree of accuracy.


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## bigshot

i have a lot of audiophile vinyl, and I doubt any of it goes much beyond 50. thats in practice though, not in theory.


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## gregorio

amirm said:


> [1] As I noted, in my last post, and have said over and over again, you cannot use peak SPL numbers with regards to what causes hearing damage.
> [2] You say that based on what peak SPL measurement or study?
> [3] Once again, all of our impressions of what is "dangerous" is reading SPL charts online.



1. Yes you can! All the literature and evidence suggests that 140dB SPL is dangerous at ANY duration and if we're talking about absolute extremes, as you are, then 16bit is capable of up to 150dB dynamic range in Fletcher-Munson's critical hearing band, how is that not enough? Forget not being enough, how is that NOT very dangerous?

However, you are deflecting, again! None of that is directly relevant because there are no music recordings in "real life" which require the peak levels and dynamic range 16bit can achieve. Your own posted quote of a world class recording studio showed a typical orchestral recording session, using mics about 10ft above the conductor position with a 114dB SPL max peak and a SNR of roughly 75dB, how is noise-shaped 16bit not enough for that dynamic range? *Stop deflecting and answer this question!!*

2. The noise floor of an empty room with the HVAC switched off is a useless and inapplicable "real life" number! If you answer my question of the SPL/frequency distribution of the noise floor in a concert venue (or recording studio) *occupied with actual living musicians and an audience*, which of course is "*real life*", then I'll answer your question of "what peak SPL measurement"!

3. And once again, no it is NOT! I've already given you the study and then mentioned it numerous times but you just ignore it. You are the one asking for actual real life music levels and more than the dynamic range offered by 16bit. The study I linked to clearly shows hearing damage of orchestral musicians at real life levels. Additionally, just a few minutes on google will show quite a body of evidence of hearing damage for musicians. Here's another example: Incidence and relative risk of hearing disorders in professional musicians - "_A nearly fourfold higher adjusted HR for noise-induced hearing loss and a 57% higher adjusted HR for tinnitus was found for professional musicians in comparison with the general population._" This demonstrates that listening to music from a close perspective, to achieve/experience the real life peak levels in excess of 120dB, as musicians do, IS potentially dangerous/damaging!!

G


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## WoodyLuvr (Nov 25, 2017)

@gregorio
Very good point.  I personally know two very well known percussionists who have been playing since their early childhood and they both are now fighting severe hearing loss in their early 40s.  If I am not mistaken they both are suffering with a loss of nearly 25 dB!  They both have told me that the culprit was the continual long term exposure to 90+ dB noise levels typical for their particular section that frequently experiences drum rolls reaching 105 dB; sustained notes peaking over 120 dB; and single notes nearing 140 dB!  A percussionist section is by far the one most likely to cause deafness more so than even brass or strings.  I was completely shocked when I first heard/learned this as it never occurred to me that classical musicians are just as prone to hearing damage than any other genre.


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## castleofargh (Nov 25, 2017)

@amirm let's think a moment. you want to play the game of looking at all extreme circumstances as if they're the norm. I've said it many times and will repeat that I see nothing wrong with that in the perspective of establishing a 100% sure transparency format. I don't believe we need such a format for transparency in music while listening reasonably, but I'm fine with you having higher expectations.  thinking your way 16/44 should be abandoned right away for so many reasons from potentially hearing 20khz and the filters at 120dB, to simply having crap DACs that suck at 44.1 but do fine above. but people can also go with listening to music reasonably loud and avoiding crap DACs. it pretty much solves the 16/44 issue for the vast majority of situations as demonstrated by the consistent failure to pass blind tests using musical content.

now what I cannot possibly agree with, is you saying on an audio forum that it is fine to listen to music while setting the peaks at 120dB SPL. you must have a sense of how dangerous you are when saying that on a forum for consumers and amateur audiophiles. to them all you're saying is that 120dB(that some may identify as so loud it hurts) is fine for a few minutes as peak levels, meaning they were probably worrying about listening too loud for no reason.
despite always warning and always being dramatic about it, we seem to have a global increase in young people with hearing loss that is likely due in part to headphones and IEMs. so insisting that 120dB peaks are fine for realistic listening, I think it's super bad and you trying to be right is just not worth it at all.

here are my random points that you or others may decide to consider:
- you have to keep in mind that those stuff will always exist in 2 forms: the precautionary principle, and the limit that will let a factory open and avoid the country's industry to spend billions to adapt to the required noise standards.
and unless we're really gullible, we know that some compromises are usually reached in practice.

- even if those numbers are the reliable ones instead of the older standard that was way wayyyy more cautious. we know for a fact that we humans are not equals, some people will get damaged sooner, because they have a weak stapedius muscle, or because their contraction reflex starts at 75dB instead of around 65 or 70dB or simply because some part of the ear are just weaker or create a louder resonance somewhere... if you want to play the "every situation counts" game, then do it for humans too.

- and last but not least. let's say your paper is right and we can spend minutes at 120dB everyday. I really am no authority in this, so perhaps it's the reliable number. what about the rest of the song? what about the rest of the day? it's all about energy quantity over time, if you start playing with fire and spending a full album set for peaks at 120dB, it will likely be only a few secs total at 120dB, and more at 115 and more at 110.... it's not fine to spend 7minuts at 120dB+15mn at 115dB+30mn at 110db...  if I have to spend the rest of my day between a loud office and the streets of a city, and commuting, my remaining "capital" of sound for the day is not going to be what's in those graphs anymore.


in short, I can only agree with Gregorio and say: "people, do not listen to music at 120dB peak" and if you have no idea how to measure that, well just ask people around you if they think you're listening too loud. at least don't abuse listening levels if you like hearing stuff in a wide range of frequencies for as many years as possible. and for the guy living in the mountain in isolation after a full day of silence, lucky you! you can push your music to 11 and listen like you were with the musicians with the real levels and dynamic. and feel all the emotion of thinking "wow, that's really loud, is that really the most enjoyable way to listen to music?".


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## RRod (Nov 25, 2017)

If your album RMS is -20dBFS, then 120dBSPL peak means listening at 100dBSPL. I cannot fathom listening to an hour of material at that level. 80-90dBSPL sounds more reasonable, which means only albums down around -30dB would possibly get near the 120dB peak, and that's pretty much the lowest album RMS I've seen except for a couple of -35dB weirdos.  And I probably don't listen to those albums at the high end of that SPL range. I also seriously doubt that you're going to appreciate truncation artifacts in the reverb tails when you've just finished a few minutes of 100dBSPL RMS…


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## TheSonicTruth

manaox2 said:


> Quote:
> 
> 
> Originally Posted by *mark_h* /img/forum/go_quote.gif
> ...



Actually, you're never safe from the client's vision, which the engineers at each stage of the production process(record, mix, master) are being paid to fulfill. Even if that vision is to have the loudest finished product in the history of recorded sound.  LOLOL!


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## gregorio (Nov 25, 2017)

RRod said:


> If your album RMS is -20dBFS, then 120dBSPL peak means listening at 100dBSPL. I cannot fathom listening to an hour of material at that level. 80-90dBSPL sounds more reasonable



@amirm stated in an earlier post that peak levels were 5 or 10dB higher than average levels, so that would be 110-115dB SPL. You can't listen to that safely for more than a few minutes and personally I would not recommend listening at those average levels at all. There's two points to bear in mind:

1. 16bit is capable of 120dB, so for 16bit not to be enough, we've got to be talking about levels beyond 120dB.
2. The elephant in the room here is not just 120dB+ peaks which is bad enough, what we're talking about is worse, it's 120dB dynamic range. IE. Simultaneously 120dB peaks AND amirm's 0dB listening environment! The analogy has already been drawn (and ignored!). 0dB SPL, is like the darkest room imaginable where your eyes can still see something after they've become accustomed to to darkness. 120dB SPL, is like the brightest sunlight you can see for a short time without damaging your eyes after they've become accustomed to bright light. If you're in a bright room and you go outside into very bright direct sunlight, it's a little uncomfortable but bearable. Go into a completely dark room, give your eyes 15 minutes to adjust to that darkness and then go outside into that same very bright direct sunlight. It's no longer just a bit uncomfortable, it's painful! What amirm is doing here is going from one extreme human limit to the other, at the same time! With classical (and even with pop music in some places), we have fairly frequent loud notes/chords decaying into the noise floor, you are getting both extreme limits of human hearing in a short period of time. Now, I've done a quick search and can't find much research on this but my personal experience suggests that when the noise floor is very low, the comfortable and pain thresholds are lowered commensurately and pain is generally the body's feedback mechanism telling you that you're doing damage! In other words, in a very low noise floor environment, damage is caused at lower SPLs than would usually be the case. Maybe there is research which proves (or disproves) this suggestion, I haven't looked very well but I don't think there is. Amirm's approach appears to be; if there's not a published AES paper on it, then it doesn't exist. My approach, particularly when it comes to potentially damaging my hearing for the rest of my life, is to err on the side of caution, not push my ears to both extremes of human limits at the same time and turn it down when it gets uncomfortable!

G


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## jgazal (Nov 25, 2017)

gregorio said:


> The analogy has already been drawn (and ignored!). 0dB SPL, is like the darkest room imaginable where your eyes can still see something after they've become accustomed to to darkness. 120dB SPL, is like the brightest sunlight you can see for a short time without damaging your eyes after they've become accustomed to bright light. If you're in a bright room and you go outside into very bright direct sunlight, it's a little uncomfortable but bearable. Go into a completely dark room, give your eyes 15 minutes to adjust to that darkness and then go outside into that same very bright direct sunlight. It's no longer just a bit uncomfortable, it's painful! (...) Now, I've done a quick search and can't find much research on this but my personal experience suggests that when the noise floor is very low, the comfortable and pain thresholds are lowered commensurately and pain is generally the body's feedback mechanism telling you that you're doing damage! In other words, in a very low noise floor environment, damage is caused at lower SPLs than would usually be the case. Maybe there is research which proves (or disproves) this suggestion, I haven't looked very well but I don't think there is.
> G



I am quite sure our skin nerve cells detect temperature gradients instead of absolute temperature: https://www.scientificamerican.com/article/cold-or-warm-can-we-really-tell/.

But retinal cells works entirely different. I guess they detect the absolute rate of photons striking its atoms and generate a proportional nerve impulse. Your brain then reacts and close or open your pupil. I would not use such analogy.

Perhaps the nerve impulse of epithelial cells in the spiral organ is then similar to the one created by retinal cells and different from skin nerve cells.

I am not sure. I really would not use that analogy.


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## castleofargh (Nov 25, 2017)

where there is a small parallel is in how the eye will adjust the amount of light coming in and so will the ear to sound level. so our actual dynamic range can seem to be much wider than it really is. the ear taking a gunshot when in a quiet environment otherwise will take the full blow of it. repeat when the ambient noise is already high enough to trigger the dampening of the ear, and now what will get to the cells will be much attenuated. in that respect the "element of surprise" is a real issue for us both for the eye and the ears.
not that this is actually relevant because luckily there are very few music composers that would repeatedly be such jerks and make crazy calm passages a good 60dB or more below full scale, just to make something explode at max level suddenly. scary movies might be a better fit than music for such examples. ^_^  24bit audio for scary movies!


-edit: to make horrible angrish a little better.


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## jgazal (Nov 25, 2017)

I stand corrected. There is a mechanism in hearing which is similar to the pupil (apparently it does not affect all the detectable frequency range, although the posterior amplification in the ossicles may not be equal in the entire range also):



> The eardrum can also serve to protect the inner ear from prolonged exposure to loud, low-pitch noises. When the brain receives a signal that indicates this sort of noise, a reflex occurs at the eardrum. The tensor tympani muscle and the stapedius muscle suddenly contract. This pulls the eardrum and the connected bones in two different directions, so the drum becomes more rigid. When this happens, the ear does not pick up as much noise at the low end of the audible spectrum, so the loud noise is dampened.
> https://health.howstuffworks.com/mental-health/human-nature/perception/hearing2.htm


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## sonitus mirus

jgazal said:


> I stand corrected. There is a mechanism in hearing which is similar to the pupil (apparently it does not affect all the detectable frequency range, although the posterior amplification in the ossicles may not be equal in the entire range also):



If you have ever attended a music concert, especially for pop/rock, you may have experienced this effect.  Everyone that enjoyed the ZZ Top, Scorpions, Rush, or Pink Floyd show had to nearly scream at one another to be heard for several minutes after the concert ended.


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## jgazal

sonitus mirus said:


> If you have ever attended a music concert, especially for pop/rock, you may have experienced this effect.  Everyone that enjoyed the ZZ Top, Scorpions, Rush, or Pink Floyd show had to nearly scream at one another to be heard for several minutes after the concert ended.



I now recall I have been tested in audiometry with a short transient to verify such acoustic reflex: https://en.wikipedia.org/wiki/Acoustic_reflex

Too much to learn. Too much to remember...


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## TheSonicTruth (Nov 26, 2017)

JaZZ said:


> Quote:
> 
> 
> Originally Posted by *scompton* /img/forum/go_quote.gif
> ...



And that's the point!!  smh.  That is what Monte Montgomery has been saying on  https://people.xiph.org/~xiphmont/demo/neil-young.html  about high res downloads - as deliverable -  being "silly".

If you hear a *significant* difference between a high-res and your garden variety 16/44.1 file or CD, it's probably a different MASTER.


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## amirm

bigshot said:


> Listening to music with 120dB peaks probably won't damage your hearing, but it won't be comfortable.


I ask again, based on what?  Almost no one here has had a peak SPL meter to know what it sounds like.  So any subjective comments like this, is just speculation.

Let's remember that Fielder's research into peaks in various music genres came from listening in actual seats, at his prefered location:





Pretty sure he was not uncomfortable or he would normally sit somewhere else.


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## RRod

amirm said:


> I ask again, based on what?  Almost no one here has had a peak SPL meter to know what it sounds like.  So any subjective comments like this, is just speculation.
> 
> Let's remember that Fielder's research into peaks in various music genres came from listening in actual seats, at his prefered location:
> 
> ...



Minimum album RMS I have in my collection is -35dBFS. That means your 129dBSPL would equate to 94dB RMS, which is pretty darn loud for an hour or two of listening. So yeah, either there is some upward compression inherent to material made for home presentation, or something is wonky with Fiedler's listening position or some of the peak measurements.


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## amirm

gregorio said:


> 1. Yes you can! All the literature and evidence suggests that 140dB SPL is dangerous at ANY duration and if we're talking about absolute extremes, as you are, then 16bit is capable of up to 150dB dynamic range in Fletcher-Munson's critical hearing band, how is that not enough? Forget not being enough, how is that NOT very dangerous?
> G


What literature?  I have not seen one reference to any peer reviewed publication from you or anyone else advocating your point of view.  

As to the rest of your argument, let's review what you are saying so that everyone can follow us.

Much of music produced today starts at 24 bits, not 16.  If you deliver at 16, you need to dither the samples prior to conversion to 16 bits.  If you don't you actually create distortion.  The most recommended form of dither is TPDF.  Let's review the minimum amount of noise that creates relative to threshold of hearing from Stuart's J. AES paper, 
*Coding Methods for High Resolution Recording Systems*





I have circled the area where the noise level now exceeds threshold of hearing.  In other words, TPDF dithered 16 bit exceeds our threshold of hearing and is audible noise.

If we allow the bit depth to be 20 bits, that problem goes away:





We see that the 20 bit line is comfortably below the threshold of hearing.  Converting 20 bits to dbFS, we get max of 122 db or so, which nicely matches the peak SPL that we want to have per other research I have presented.

*Best yet, would be to leave 24 bits alone! * Again, that is how music is produced today.  If we can deliver that as is, why screw around?  Well, we are being told to screw around for reasons that have not yet been articulated.

Indeed we need to go to 24 bits if we want to leave headroom for signal processing at playback such as Room EQ, resampling, EQ, etc.  If those conversions are done at higher resolution and then dithered down, even 20 bits can start to get tricky:





We see that every time we mess with the 20 bit signal, we get closer and closer to threshold of hearing due to mandatory noise that dither adds to our signal.  So 24 bit becomes the right answer if we are to allow for some headroom.

One way to screw around is to make the noise that is created by too few bits "shaped" so that there is less of it in the mid-band area where our hearing is most sensitive.

We go back to Stuart's paper yet again for that:





We see that instead of a flat line, our dithered noise now has a shape that curves below the threshold of hearing.  So in theory we can achieve transparency using noise shaping.  In practice though, I am not sure we want to have huge amount of high frequency noise that we are pushing into our amplifiers and tweeters.  Ideally we would have more bandwidth to spread that noise into ultrasonic range, say at sampling rate of 48 Khz or higher.

So what is the problem here?  Lack of use.  Our friend here is advocating a solution but I bet that is not how he is mastering his content.  It is easy to tell if noise shaping is used by rising level of high frequencies.  Here is a random track in my library showing how it is NOT done:





As you see,  there is no rising high frequency levels where I have circled.

Another issue with noise shaping is that if you turn up the gain enough, the noise you now hear does not have flat response and can be much more objectionable.  Again, from Stuart:


 

Now, i*n the days we were stuck with CD as a forum, jumping through hoops with noise shaping may have had merit.  But today, we don't need to do that.  All of us from our cell phones to computers and DACs can play 24 bit files with ease.  There is no reason on earth left that music must be brought down to 16 bits for audiophiles.  None. * Give us the darn 24 bits for heaven sake.  We don't need you to convert anything.

And oh, while you are at it, *give it to us at high sample rate that was used prior to mastering that converted it to any other.  *If we want to, we can convert it to 44.1 or download that version if you offer it.  But no good is coming from someone in the middle trying to create a job for themselves by converting bits for us.  These kind of digital conversions are readily available to us.  In Roon for example I can set any of these limits and it works automatically for me.

Fortunately this is happening now and have access to such content.  The gates have opened and the arguments to the contrary are just that: arguments that are just polluting our forum airways.


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## bigshot (Nov 26, 2017)

I don't need a meter to know that peaks that reach the threshold of pain isn't a comfortable listening level.

There is no audible benefit to high bitrate / high sampling rate sound. 16/44.1 is all you ever need for the purposes of listening to music in the home.


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## RRod

amirm said:


> Another issue with noise shaping is that if you turn up the gain enough, the noise you now hear does not have flat response and can be much more objectionable.  Again, from Stuart:
> 
> 
> Now, i*n the days we were stuck with CD as a forum, jumping through hoops with noise shaping may have had merit.  But today, we don't need to do that.  All of us from our cell phones to computers and DACs can play 24 bit files with ease.  There is no reason on earth left that music must be brought down to 16 bits for audiophiles.  None. * Give us the darn 24 bits for heaven sake.  We don't need you to convert anything.
> ...



Right there in the quote it says "if the gain is increased sufficiently", so how is any of this an issue for any material that doesn't make you crank the pot? And of course there isn't any reason NOT to use 24 bits at this point. One could also argue there isn't any reason to use anything but lossy codecs for delivery at this point too. The question is "are people actually hearing real improvements or is it all in their head", and the "improvements" people tout are rarely just a bit less noise modulation in the final reverb tails of an album…


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## amirm

RRod said:


> Here's an example reverb tail from a non-live recording made at the symphony hall previously discussed. I've put random names on these test files (24/44100), but rest assured one of them is the original > 16-bit source. Once y'all have found a minimal level where you can hear some differences, I'll put up the loudest part of the album. For my own part, in my 35dBA bedroom, listening on my PM-3s (which isolate well >1kHz), with the volume cranked as loud as I care to for this album, I can maybe hear just a bit of difference in one file; whether that holds up in ABX I haven't tried.


I just did an ABX test on a couple of them at random and got this:

foo_abx 1.3.4 report
foobar2000 v1.3.2
2017/11/26 10:50:22

File A: C:\Users\Amir\Documents\Test Music\Headfi RR Samples\07094.wav
File B: C:\Users\Amir\Documents\Test Music\Headfi RR Samples\27776.wav

10:50:22 : Test started.
10:51:03 : 01/01  50.0%
10:51:17 : 02/02  25.0%
10:51:25 : 03/03  12.5%
10:51:36 : 04/04  6.3%
10:51:42 : 05/05  3.1%
10:51:52 : 06/06  1.6%
10:52:00 : 07/07  0.8%
10:52:10 : 08/08  0.4%
10:52:17 : 09/09  0.2%
10:52:25 : 10/10  0.1%
10:52:36 : 11/11  0.0%
10:52:41 : Test finished.

 ---------- 
*Total: 11/11 (0.0%)*

So complete ability to tell them apart.  Test system was my everyday laptop driving the Fiio E10 K DAC+headphone running through Windows audio stack set to 24-bit, 44.1 Khz (see my review here: https://audiosciencereview.com/foru...dslabs-odac-rev-b-compared-to-fiio-e10k.2068/).  Headphones were my Etymotic E4RS IEMs.  I am sitting in our living room with rain pouring on our roof.

Can I see a few others run the same test and report back?  I like to get calibrated on who can or cannot hear the difference.  That would give us good subjective reference in the conversation.


----------



## amirm

RRod said:


> Minimum album RMS I have in my collection is -35dBFS. That means your 129dBSPL would equate to 94dB RMS, which is pretty darn loud for an hour or two of listening. So yeah, either there is some upward compression inherent to material made for home presentation, or something is wonky with Fiedler's listening position or some of the peak measurements.


-35 dbFS is your minimum????  How did you determine that?  Is it peak, RMS or average?  What is the window size?

Here is the stats on my last clip I randomly posted:






You get numbers like that because there is fade to digital silence.  If I select just the loud portions I get:





That is still well below your number.  If I select a little less loud section I get this:








So we are down to -60 dbFS which is well below your -35 dbFS.

A full histogram shows the same:






And let's remember that these programs can NOT separate music from noise.  We can hear sound/music well below noise floor so effective music dynamic range is higher than these programs show.

The above are also RMS values and not peak (really dips).  And there is an averaging window.

This is why we don't want to split the peas this way.  The method used by Fielder is to measure the lowest level noise of microphones and ADCs and use that as the lowest limit.  After all, the music can get as soft as it wants.


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## amirm

gregorio said:


> However, you are deflecting, again! None of that is directly relevant because there are no music recordings in "real life" which require the peak levels and dynamic range 16bit can achieve. Your own posted quote of a world class recording studio showed a typical orchestral recording session, using mics about 10ft above the conductor position with a 114dB SPL max peak and a SNR of roughly 75dB, how is noise-shaped 16bit not enough for that dynamic range? *Stop deflecting and answer this question!!*



I did not post any pictures of "world class recording studio."  That picture was a random grab online showing that microphones are NOT put in the seating rows with the rest of the audience being around.  This was in response to you saying classical concert have high noise levels due to those factors.  So your argument there was wrong and misleading.  That is why I post that picture.

We have no data on how quiet that studio really is.  The data that we do have is in the research that I posted.



> 2. The noise floor of an empty room with the HVAC switched off is a useless and inapplicable "real life" number!


Of course it is real life for best recordings and techniques.  Here is the data for the Davies Hall by the way with HVAC on:





So still darn good.  But people who care about fidelity will absolutely turn off the HVAC to get the best noise floor.




> If you answer my question of the SPL/frequency distribution of the noise floor in a concert venue (or recording studio) *occupied with actual living musicians and an audience*, which of course is "*real life*", then I'll answer your question of "what peak SPL measurement"!


There you go again: vast majority of music we consume is recorded WITHOUT audience.  Here is one at Skywalker Ranch for example that I showed you before:



Honestly it is preposterous to sit there and define for the rest of the world what "real life is."  Real life is whatever it is.  We do not want to put limits on its performance to win some old arguments on forums.  Why on earth would we want to pick a digital format that would say making the recording room less noisy is useless because our format is too noisy???

The studies that I have posted are old.  Today state-of-the-art design for recording and playback spaces have advanced hugely.  There is for example full computation fluid dynamics analysis of HVAC systems to make sure they are far, far quieter than they have been in the past.

The world I live in strives to make some of the best recordings possible and played in rooms that are state-of-the-art.  Just look at some of the designers Keith Yates creates with full analysis per above.

Yes, you can dumb down fidelity as much as you want.  That is a personal choice.  It simply is not something you can force onto others who have higher standards.


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## amirm

Here is yet another data point showing that recordings are done with HVAC, etc. shut off:





Performance Condition is defined as:





So it is not true at all that all of these noise sources are left on when making recordings.


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## bigshot (Nov 26, 2017)

When I listen to music, I set the volume level at the beginning to a comfortable listening level and I leave it there. I don't crank the volume on the ring outs at the ends of songs. Imagine going to an art museum and looking at the surface of the paintings with a big magnifying glass. You would think Rembrandt was nothing more than just a blob of color there and a crack in the paint here.






Attention to scale is important. Without perspective you can chase down rabbit holes after details that pale in significance under the totality of the whole.






Human perception has limits. Abstract science doesn't. It's important to know when abstract perfection is called for and when it isn't necessary or even desirable. The point of having a stereo in your home is to appreciate listening to great music. It's not to split atoms or meet some abstract standard. I guess if your interest is only in the equipment and its performance, it's fine to focus on details and ignore the overall, but if you are intending to listen to music, you need to focus on the overall and be sure that the details serve it rather than detract from it. I think a lot of people in audiophile circles lose sight of these things.

In fact, I'm coming to realize that music may be tangential to the whole hobby for a lot of people. I look at the sort of music people talk about in audiophile forums and I'm shocked at how pedestrian a lot of people's tastes are. They spend tens of thousands of dollars on equipment to listen to the exact same music they listened to in high school! I know they don't want me to feel this way about them, but I can't help feeling sorry for them. The world of music is so vast and amazing. It can enrich your life much more than higher sampling rates and noise floors further down below any thresholds you can actually hear. Yet some people spend all their time talking about numbers on a page with great detail and thought, and never even consider the whole purpose behind those numbers- the appreciation of music. Imagine if they applied their analytical efforts to thinking about important things, like music.

It seems backwards to me, but if they enjoy it, it's good for them I guess.


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## 71 dB

amirm said:


>



That requires you play at level 114 dB SPL = Full scale. Let's assume your CD is so quiet it hits -20 dBFS at peaks (that's stupid waste of dynamic range, but possible of couse). So, the peaks are "only" 94 dB SPL. You think you can hear dither at 0-10 dB SPL level why listening to music with 94 dB SPL? You think your amp doesn't generate even more noise than dither is?

If you play at 100 dB SPL = Full scale (realistic real life "LOUD" listening level), dither goes down so that it's just at the hearing threshold, except to hear sounds at hearing threshold you need silence before it…

So, in practise you don't need more than 16 bits, something we have told you 1000 times already. You interpret these scientific facts a way that has no relevance.


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## bigshot (Nov 26, 2017)

I'm afraid I don't know what you're arguing against Amirm. Of course recordings are made with as quiet a noise floor as possible. But many great concert halls are not soundproofed. They are in the middle of busy cities with traffic noise outside that bleeds through into the hall itself. But it doesn't matter because by the time the orchestra is playing, there's enough energy in the room to drown it out, even if it would be clearly audible if the orchestra was totally silent. The same goes for your living room. Even if you listen with the A/C on and the fan on your computer running, once the music starts you can't hear it, because your ears can only hear about 45-50dB of dynamic range at a time. If your volume is set for 75dB, then a 30dB noise floor probably doesn't matter a whole heck of a lot.

The main difference is that in a recording, the engineers might have to boost something to achieve proper balance. That means that they need a quieter noise floor than we do in our living room. When we listen to music we set the volume level for the peak loudness to find a comfortable range, and we leave it there until the album is over. In that respect, listening has different requirements than recording.

The reasoning we were questioning the photo of the recording session you posted was that it didn't appear to be a music recording session. It was a film dubbing session, which has totally different requirements than a music recording session. Here is a photo of a more typical large scale classical recording. Notice that the microphones are quite a distance above the band. There's probably also mikes towards the back of the hall to pick up ambience.






This photo was taken at a recording of Wagner's Ring. It was produced by John Culshaw and it was one of the first opera recordings to take advantage of stereo soundstage. On the stage in the middle, there were squares with numbers on them which corresponded to the blocking of the singers. They would move around the space and it would be captured in perfect stereo spread by the recording. When played back on a proper speaker system, you could hear the actors moving around the stage in front of you.


----------



## 71 dB

bigshot said:


> In fact, I'm coming to realize that music may be tangential to the whole hobby for a lot of people. I look at the sort of music people talk about in audiophile forums and I'm shocked at how pedestrian a lot of people's tastes are. They spend tens of thousands of dollars on equipment to listen to the exact same music they listened to in high school! I know they don't want me to feel this way about them, but I can't help feeling sorry for them. The world of music is so vast and amazing. It can enrich your life much more than higher sampling rates and noise floors further down below any thresholds you can actually hear. Yet some people spend all their time talking about numbers on a page with great detail and thought, and never even consider the whole purpose behind those numbers- the appreciation of music. Imagine if they applied their analytical efforts to thinking about important things, like music.
> 
> It seems backwards to me, but if they enjoy it, it's good for them I guess.



I feel the same way. It is so easy to get lost in this hobby. I have been "mesmerized" by specs too at times. It's possible the need for "better" sound is a sign that you are bored with your music and you think more bits, larger sampling rates or maybe even new snake oil cables will fix the problem for you. It doesn't fix the original problem, it just turns our attention away from the "boring" music. Whenever I discover new exciting music I don't even think about sound quality issues. I just "gobble" the music! Music reproduction technology has reached a few decades ago a level where "good enough" sound quality isn't even expensive if you know how to spend your money. That doesn't mean I don't care about sound quality because I do. I just try to put limits on what is relevant and what's overkill. True understanding of sound reproduction is to know how much fidelity is needed, at what point it's nothing more than numbers on paper.


----------



## gregorio

amirm said:


> [1] I did not post any pictures of "world class recording studio."  That picture was a random grab online showing that microphones are NOT put in the seating rows with the rest of the audience being around.  This was in response to you saying classical concert have high noise levels due to those factors.  So your argument there was wrong and misleading.  That is why I post that picture.
> [2] But people who care about fidelity will absolutely turn off the HVAC to get the best noise floor.
> [3] There you go again: vast majority of music we consume is recorded WITHOUT audience.
> [4] Honestly it is preposterous to sit there and define for the rest of the world what "real life is."  [4a] Real life is whatever it is.
> [5] We do not want to put limits on its performance to win some old arguments on forums.  [5a] Why on earth would we want to pick a digital format that would say making the recording room less noisy is useless because our format is too noisy???



1. Now that's a lie, you used that same photo in the post on your own forum, which I had nothing to do with.
2. But the people who care about breathing above fidelity will leave it on! In fact, in many/most cases when the audience is present it is a legal requirement to have the HVAC on!
3. You think maybe I've failed to notice that when I've been recording an orchestra?
4. Then why are YOU doing it?
4a. Really, doesn't real life at least start with life? Your figures do not include living musicians! I feel strangely comfortable defining the real life recording of an orchestra as actually needing a living orchestra!
5. We put limits on the recordings to improve the perceived quality for the consumer, such as using mics with a 114dB SPL max peak. A point you've ignored, AGAIN!!
5A. Why on earth would we want to pick a digital format with a noise floor lower than -150dB?



amirm said:


> What literature? I have not seen one reference to any peer reviewed publication from you or anyone else advocating your point of view.



AGAIN, if you haven't seen it then it doesn't exist, right? Try that as a legal defence and see how far you get! For everyone else, the OSHA chart already posted in this thread and the legal requirements of many countries allow NO exposure to 140dBSPL of ANY duration without hearing protection! Amirm, again you are being ridiculously irresponsiblee and for what reason? For no better reason than what you are accusing others of, winning an argument on a forum!



amirm said:


> The most recommended form of dither is TPDF.



No it's not, it's noise-shaped dither and has been for many years! And, just for a change, you keep ignoring the asked question, if noise-shaped dither is not used on a master, why do you think that is?



amirm said:


> But no good is coming from someone in the middle trying to create a job for themselves by converting bits for us.



Again, you are ignoring real life. How many times amirm are you just going to repeat the same thing over and over and ignore the responses which demonstrate it's nonsense? We have NO CHOICE, most professional recordings are made in 64bit, we cannot distribute 64bit, therefore we HAVE to convert the "bits for you"! This isn't difficult to understand or look up, why are you deliberately ignoring so much "real life", what are you trying to sell?

G


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## jgazal (Nov 26, 2017)

bigshot said:


> Human perception has limits. Abstract science doesn't. It's important to know when abstract perfection is called for and when it isn't necessary or even desirable. The point of having a stereo in your home is to appreciate listening to great music. It's not to split atoms or meet some abstract standard. I guess if your interest is only in the equipment and its performance, it's fine to focus on details and ignore the overall, but if you are intending to listen to music, you need to focus on the overall and be sure that the details serve it rather than detract from it. I think a lot of people in audiophile circles lose sight of these things.
> 
> In fact, I'm coming to realize that music may be tangential to the whole hobby for a lot of people. I look at the sort of music people talk about in audiophile forums and I'm shocked at how pedestrian a lot of people's tastes are. They spend tens of thousands of dollars on equipment to listen to the exact same music they listened to in high school! I know they don't want me to feel this way about them, but I can't help feeling sorry for them. The world of music is so vast and amazing. It can enrich your life much more than higher sampling rates and noise floors further down below any thresholds you can actually hear. Yet some people spend all their time talking about numbers on a page with great detail and thought, and never even consider the whole purpose behind those numbers- the appreciation of music. Imagine if they applied their analytical efforts to thinking about important things, like music.
> 
> It seems backwards to me, but if they enjoy it, it's good for them I guess.



I find the distribution bit depth debate somehow boring.
But I confess that I feel very curious about psychoacoustics.
I wrote a very long post about technicalities of immersive audio so I guess then I may match your criteria of music alienation. I certainly don’t match any criterium of music erudition. And the immersive audio post received low feedback.
Do you believe that immersive rendering of music is then somehow irrelevant and that such rendering will be restricted to the gaming and virtual reality niche?
I am going to feel very sad if the demand for immersive rendering of music fails as bad as stereoscopic television. 
Any cheer up music that would you recommend?


----------



## amirm

71 dB said:


> That requires you play at level 114 dB SPL = Full scale. Let's assume your CD is so quiet it hits -20 dBFS at peaks (that's stupid waste of dynamic range, but possible of couse). So, the peaks are "only" 94 dB SPL. You think you can hear dither at 0-10 dB SPL level why listening to music with 94 dB SPL? You think your amp doesn't generate even more noise than dither is?


What do you mean 'let's assume?"  Assume based on what research?  Where did you get -20 dbFS?  And who says that is "stupid waste of dynamic range???"  As I have repeatedly shown, peak of 120 db SPL is supported in research of actual music halls.  The threshold of hearing is below 0 dbFS, so the total dynamic range needed for transparency is 125 dB, not 94:





Reducing this to 94 dB means leaving 21 db on the floor.  For no good reason whatsoever. 

Is there any hope in our future that you guys argue based on proper references?  Or are we doomed with assumption after assumption not verified in the slightest?


----------



## amirm

71 dB said:


> So, in practise you don't need more than 16 bits, something we have told you 1000 times already. You interpret these scientific facts a way that has no relevance.


Sitting here, you still have not read the research and yet have that opinion of it??? Here is the author's own summary since you doubt mine with respect to the chain:






You see 96 db anywhere???

And here is the final conclusion of the paper:









It couldn't be more clear about the requirement for a noise-free channel to be 122 db and not that 96 db number you just threw out there.

As you say, repeating what you think over and over again is not going to work here.  I have proper research that aimed to investigate this very topic.  It is strongly referenced in countless other papers and is not something you can just dismiss out of hand.

Go and do some research, find something that disputes it with the same credentials and then we can talk.

Until then, both your goals and assumptions therein are fallacious.  Myths created in the guise of dispelling myths.


----------



## amirm (Nov 26, 2017)

gregorio said:


> AGAIN, if you haven't seen it then it doesn't exist, right? Try that as a legal defence and see how far you get! For everyone else, the OSHA chart already posted in this thread and the legal requirements of many countries allow NO exposure to 140dBSPL of ANY duration without hearing protection!


No, all you have to do is present the research.  When you don't then it means you have not read such and just want to imply such data exists.  Seeing how I back everything I say with references, you need to do the same if you want your position to have as much force. I am sure you agree that is how it works in court of law.

As to OSHA, here is what it says: https://www.osha.gov/pls/oshaweb/owadisp.show_document?p_table=standards&p_id=9735





We are not talking about "impulsive or impact noise."  And an example of how they view db measurements:





"Slow response" which means it is not peak.

So no, that doesn't suffice even though I am not advocating 140 db anything.

Honestly you need to get this whole db, hearing damage thing out of your vocabulary.  Noise is not music.  And musical peaks are not averaged as we have been discussing.


----------



## 71 dB

amirm said:


> What do you mean 'let's assume?"  Assume based on what research?  Where did you get -20 dbFS?  And who says that is "stupid waste of dynamic range???"  As I have repeatedly shown, peak of 120 db SPL is supported in research of actual music halls.  The threshold of hearing is below 0 dbFS, so the total dynamic range needed for transparency is 125 dB, not 94:
> 
> 
> 
> ...



If an recording really needs (it doesn't, because 125 dB dynamic range in consumer audio is ridigulous idea) 125 dB, then shaped dither can do it, you see it in the picture. Are we done? Is there any hope in out future that you understand why 16 bit is enough in consumer audio?


----------



## 71 dB

amirm said:


> Noise is not music.



A lot of parents think the music their children listen to is noise.  It doesn't matter how it's classified. What matters is the frequency content.


----------



## 71 dB

amirm said:


> Sitting here, you still have not read the research and yet have that opinion of it??? .



I recall reading those papers long ago. Familiar stuff. Unlike you, I didn't draw some crazy conclusions for 16 bit not beeing enough in _consumer_ audio.


----------



## RRod

amirm said:


> -35 dbFS is your minimum????  How did you determine that?  Is it peak, RMS or average?  What is the window size?
> 
> Here is the stats on my last clip I randomly posted:
> 
> ...



Your two posts perfectly encapsulate the issues we are having with your arguments. In the ABX: What is your listening level? Anyone can ABX truncated 16 bit vs. 24 bit if you can arbitrarily up the pot, especially when you separate the reverb tail from the loud music right before it. For all you know this album has a full-scale 1kHz square wave thrown in for kicks somewhere. How are we supposed to interpret you result without any other loudness context?

Re RMS, -35dB RMS is for *the whole album*, and is indicative again of where I actually set my pot to listen to the material in my listening environment. The softest sections with musical material (non-fade/reverb) I've encountered are around -70dB for a 1s RMS. In essence you seem to think of listening-to-music in a way most of us don't. I think of setting the volume once (maybe with a bit of adjustment) in a library-quiet environment, as opposed to fiddling with the pot on fading tails with IEMS. Even then, what happens when your album is mastered at -20dB RMS instead of -35? That's either 2.5bits more quiet in the room or 2.5bits louder on the pot you need to hear the same tail issues. It never seems to end with you, and that's what doesn't make any sense.


----------



## bigshot

jgazal said:


> I wrote a very long post about technicalities of immersive audio so I guess then I may match your criteria of music alienation. I certainly don’t match any criterium of music erudition. And the immersive audio post received low feedback. Do you believe that immersive rendering of music is then somehow irrelevant and that such rendering will be restricted to the gaming and virtual reality niche?



Actually, I'm very interested in multichannel mixing techniques and I've posted about it here quite a bit. I think your post didn't get a lot of response because you overloaded the first post. Threads are discussions. You throw out a few points and see the reaction you get and go from there. If you put too much information up front, it becomes an article and you don't get much interaction. There should be a document of best practices for participation in internet forums. But I bet even if there was something like that, we'd still get the same sort of pointless nonsense we are getting in this thread!


----------



## bigshot

RRod said:


> In the ABX: What is your listening level? Anyone can ABX truncated 16 bit vs. 24 bit if you can arbitrarily up the pot, especially when you separate the reverb tail from the loud music right before it.



You aren't going to get any kind of satisfactory answer to that question. I asked it a long time ago and got a song and dance.


----------



## jgazal (Nov 26, 2017)

bigshot said:


> Actually, I'm very interested in multichannel mixing techniques and I've posted about it here quite a bit. I think your post didn't get a lot of response because you overloaded the first post. Threads are discussions. You throw out a few points and see the reaction you get and go from there. If you put too much information up front, it becomes an article and you don't get much interaction. There should be a document of best practices for participation in internet forums.



I see. You are absolutely right. I did it with good intentions. I thought it would be easier or more clear to readers to identify any mistake from my part if people knew exactly where I got each concept I was describing. And I wanted to get together in a single post all the insights you all gave in disperse threads. I will be more concise and patient to build one concept at each post. I am sorry.


----------



## amirm

71 dB said:


> I recall reading those papers long ago. Familiar stuff.


Then please link to some place you have posted them so that people have the full information.


----------



## amirm

bigshot said:


> You aren't going to get any kind of satisfactory answer to that question. I asked it a long time ago and got a song and dance.


Precisely what I get when I ask you to run a single blind test and report the results.  You can start with the one in front of us.


----------



## sonitus mirus

bigshot said:


> You aren't going to get any kind of satisfactory answer to that question. I asked it a long time ago and got a song and dance.



It was already asked, and it was ignored.

https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-291#post-13854799

I'm a critical listener, hear me roar!


----------



## bigshot

Amirm, I've done lots of blind tests. I know my thresholds pretty well. 16/44.1 sounds exactly like 24/96. AAC 256 VBR is totally transparent, as is MP3 320 LAME. I can hear up to about 17kHz. Music with dynamics beyond 50dB is uncomfortable to listen to, and my normal listening levels are always under 80dB.

I think everyone who is serious about sound should know as much about the specs of human hearing as they do the specs of their equipment. If you don't know what you can and can't hear, you chase after numbers without a solid grasp of what they represent. It's also important to remove your ego from your listening tests. When you allow yourself to judge your self worth on your perceptual thresholds, you can be tempted to fudge the results by reaching for the volume knob or focusing in on tiny insignificant bits at the ends of tracks. That doesn't reflect any kind of practical application of knowledge. It isn't even knowledge. We're all human and we all hear music with human ears. The people who perceive things in music that other people can't are the ones who have studied music, not the ones that have studied audio specs.

We live in a great time for electronics. We can go to a Walmart and pick up a cheap player with perfect sound. Speakers and headphones are better than they've ever been. Quality standards continue to improve, even beyond our ability to hear the improvements. We're very lucky. We can go to Amazon and order just about any midrange piece of audio equipment they sell and have it delivered to our door... and it will perform to a very high standard. We don't need to worry about that sort of thing any more. We're free now to focus on our room acoustics, mixing and mastering and most of all musical creativity. Music is what matters.


----------



## bigshot

jgazal said:


> I see. You are absolutely right. I did it with good intentions. I thought it would be easier or more clear to readers to identify any mistake from my part if people knew exactly where I got each concept I was describing. And I wanted to get together in a single post all the insights you all gave in disperse threads. I will be more concise and patient to build one concept at each post. I am sorry.



No problem. It's an interesting subject and it really deserves its own active thread.


----------



## amirm

RRod said:


> Re RMS, -35dB RMS is for *the whole album*, and is indicative again of where I actually set my pot to listen to the material in my listening environment. The softest sections with musical material (non-fade/reverb) I've encountered are around -70dB for a 1s RMS. In essence you seem to think of listening-to-music in a way most of us don't.


That is exactly it: I am not worried about "*most of you*" (putting aside that you have no data that represents most of you -- just assumptions).  I am worried about *all of you.  *As is the research.  

When we talk about what the music industry should release to us, we need to be inclusive of the needs of everyone, at all listening levels, for all content, and all situations.

Now, if this was not achievable, OK, we fall back on what we can deliver.  OP claims this should be 16/44.1  We absolutely can certainly capture and playback at > 20 bits and far faster sampling rates with no cost to any of us.  So there can't be a motivation to back off on some arbitrary reason, or heaven forbid, job security for someone who thinks their job is to do that conversion for us.

Heck, if we want to go  by what most people find adequate, lossy compression should be the limit.  Why did OP say it has to be 16/44.1 lossless?  Isn't what 99% of the world enjoys and so is more "real life" than lossless compression?

The argument gets slippery real fast.  That's why we approach this topic methodically and determine what we *can determine to be an audibly noise-free channel. *We can prove and defend such a standard.

As for me, I have been in rooms including our own at work where my pant legs move with bass.  I have also stood near my son playing drums and it doing the same thing.  In neither case did I go deaf, or run right out of the room.  Yes, the system is loud.  This is not for everyone.  But dynamics can be fun and it is not something we want to deprive people from based on averaging entire albums, etc. type of numbers.  We are talking about instantaneous peaks and valleys.  That is what our channel stores.

Answering your question regarding level of my ABX test, I don't know how to answer that as I am using headphones and I have no way of measuring levels there.  I turned it up 'till I could hear what was there and did my ABX test.

What was the purpose of the test and can you run it and and post results of success or failure regardless of level?  I am trying to get us calibrated subjectively.

Oh, I forgot to say that the DAC headphone combo I used retails for $60.


----------



## RRod

sonitus mirus said:


> It was already asked, and it was ignored.
> 
> https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-291#post-13854799
> 
> I'm a critical listener, hear me roar!



Yeah, I figured I try again in the context of an actual example.


----------



## amirm

And to offset the people saying they don't listen loud, he is an online friend "basspig" with his system:


----------



## bigshot (Nov 26, 2017)

Oh good lord! You realize that isn't sound doing that, it's the air from the bass ports.


----------



## bigshot (Nov 26, 2017)

amirm said:


> Heck, if we want to go  by what most people find adequate, lossy compression should be the limit.  Why did OP say it has to be 16/44.1 lossless?



That's absolutely true. I have yet to find anyone who can discern high bitrate lossy from redbook or 24/96 in a controlled test using normal listening conditions. So I would say for the purposes of listening to music in the home, all of those formats are interchangeable. I'm actively seeking someone that can prove that they can discern a difference under real world listening conditions. If I find one, I'll change my opinion.

"Whosoever draweth this sword from this stone and anvil..."


----------



## amirm

castleofargh said:


> @amirm let's think a moment. you want to play the game of looking at all extreme circumstances as if they're the norm. I've said it many times and will repeat that I see nothing wrong with that in the perspective of establishing a 100% sure transparency format. I don't believe we need such a format for transparency in music while listening reasonably, but I'm fine with you having higher expectations.  thinking your way 16/44 should be abandoned right away for so many reasons from potentially hearing 20khz and the filters at 120dB, to simply having crap DACs that suck at 44.1 but do fine above. but people can also go with listening to music reasonably loud and avoiding crap DACs. it pretty much solves the 16/44 issue for the vast majority of situations as demonstrated by the consistent failure to pass blind tests using musical content.


You guys keep using this "vast majority of situations."  Where is the data to back it?  How hard have people/OP looked?  Who here knew about he research I have been presenting?

Do you know that peer review reports of recent tests show detection of downsampled high-resolution music?  Like this: https://audiosciencereview.com/forum/index.php?threads/high-resolution-audio-does-it-matter.11/

And what is with "playing games?"  I have proposed no new methodology on my own.  I have shown you research from multiple peer reviewed journal research papers that use the identical methodology: finding threshold of detection and comparing with peak music in real life to determine dynamic range.  This is it not a game.  It is how to do this. 

We have a responsibility to not cheat the membership from this data by constantly mispositioning it.  



> now what I cannot possibly agree with, is you saying on an audio forum that it is fine to listen to music while setting the peaks at 120dB SPL. you must have a sense of how dangerous you are when saying that on a forum for consumers and amateur audiophiles. to them all you're saying is that 120dB(that some may identify as so loud it hurts) is fine for a few minutes as peak levels, meaning they were probably worrying about listening too loud for no reason.
> despite always warning and always being dramatic about it, we seem to have a global increase in young people with hearing loss that is likely due in part to headphones and IEMs. so insisting that 120dB peaks are fine for realistic listening, I think it's super bad and you trying to be right is just not worth it at all.


I don't know how many times I need to repeat this: *you have no idea how loud peak 120 db is. * It is not what you think it is.  Your ideas are all based on slow averaging db meters used for noise pollution in workplaces.  Don't you think the research papers would be full of warnings if you were right?

What kills your hearing is not a peak here and there anyway.  What kills your hearing is the loudness compression and forces the peaks and valleys to be so close together.  That then causes the *average* loudness to be  close to peak SPL and then you do have to worry.  Such is not the case in well recorded music.

And any rate, I am pretty sure people coming reading this forum are not here to be warned about listening too loud.

What we here for is because OP told us this:






I have shown you noise floor of recording spaces that is at the threshold of hearing.  And we have measurements of peak music in the same rooms in excess of 120 dB.  That makes the statement in yellow total folklore and nonsense.

You agree that neither OP nor anyone else has presented one bit of research that validates that, yes?  I mean that is not a small difference.  We are talking about claimed 60 db to > 120 db!!!

Am I the only one here who cares about us not saying things that are so, so wrong?


----------



## amirm

bigshot said:


> That's absolutely true. I have yet to find anyone who can discern high bitrate lossy from redbook or 24/96 in a controlled test using normal listening conditions. So I would say for the purposes of listening to music in the home, all of those formats are interchangeable. I'm actively seeking someone that can prove that they can discern a difference under real world listening conditions. If I find one, I'll change my opinion.
> 
> "Whosoever draweth this sword from this stone and anvil..."


Don't keep repeating that.  You will have your hat handed to you on that.  Here are my results with content picked for a high-resolution audio challenge on AVS Forum so not critical content necessarily for lossy compression: http://www.avsforum.com/forum/91-au...-high-resolution-audio-test-ready-set-go.html

---

foo_abx 1.3.4 report
foobar2000 v1.3.2
2014/07/31 15:18:41

File A: C:\Users\Amir\Music\AIX AVS Test files\On_The_Street_Where_You_Live_A2.mp3
File B: C:\Users\Amir\Music\AIX AVS Test files\On_The_Street_Where_You_Live_A2.wav

15:18:41 : Test started.
15:19:18 : 01/01  50.0%
15:19:30 : 01/02  75.0%
15:19:44 : 01/03  87.5%
15:20:35 : 02/04  68.8%
15:20:46 : 02/05  81.3%
15:21:39 : 03/06  65.6%
15:21:47 : 04/07  50.0%
15:21:54 : 04/08  63.7%
15:22:06 : 05/09  50.0%
15:22:19 : 06/10  37.7%
15:22:31 : 07/11  27.4%
15:22:44 : 08/12  19.4%
15:22:51 : 09/13  13.3%
15:22:58 : 10/14  9.0%
15:23:06 : 11/15  5.9%
15:23:14 : 12/16  3.8%
15:23:23 : 13/17  2.5%
15:23:33 : 14/18  1.5%
15:23:42 : 15/19  1.0%
15:23:54 : 16/20  0.6%
15:24:06 : 17/21  0.4%
15:24:15 : 18/22  0.2%
15:24:23 : 19/23  0.1%
15:24:34 : 20/24  0.1%
15:24:43 : 21/25  0.0%
15:24:52 : 22/26  0.0%
15:24:57 : Test finished.

 ---------- 
Total: 22/26 (0.0%)

---
Here are the properties of the file:





As you see, it is 320 kbps MP3.

No, levels were not adjusted, or played at max volume.  Compression artifacts are audible without such means.

Heck, I can even teach you to hear the artifacts!  But, you may not want me to.  

I plan to do a training video on this anyway, and identifying exactly what to listen for including the clips themselves.  You will go from not hearing the difference to perceiving them in double blind tests like above.

The original files may be on that AVS thread still.  If so, download and give them a try on this clip.


----------



## sonitus mirus

amirm said:


> And any rate, I am pretty sure people coming reading this forum are not here to be warned about listening too loud.


Where is the link to your scientific, peer reviewed paper verifying your assumption?


----------



## bigshot

I just read the yellow highlighted part of your post. It's correct. Orchestra recordings don't have more than 60dB of dynamic range. In fact, many of them have less than that. Orchestras aren't recorded close miked. They may produce 120dB if you were sitting right next to the percussionist, but microphones are placed at a distance from the band to capture hall ambience and the dynamics aren't as broad that way. I do know of one exception though... I have one "audiophile" recording where they put the microphones in the middle and seated the orchestra in an equidistant circle around them. It's a violin concerto, but every time the percussionist hits the tympani, you jump out of your chair. Impossible to listen to! A total mess. It probably has a dynamic range of a bit over 50dB. Not pleasant.


----------



## bigshot

amirm said:


> Here are my results.



Yeah. Very nice.


----------



## sonitus mirus

amirm said:


> Don't keep repeating that.  You will have your hat handed to you on that.  Here are my results with content picked for a high-resolution audio challenge on AVS Forum so not critical content necessarily for lossy compression: http://www.avsforum.com/forum/91-au...-high-resolution-audio-test-ready-set-go.html
> 
> ---
> 
> ...




What was used to encode the MP3 file?  What was the source?  Fairly basic properties provided. 

I was expecting something more like this:


----------



## bigshot (Nov 26, 2017)

I'd like to see AAC 320 VBR as the test codec please. In fact, I'd like to see a range of codecs and bitrates like my lossy vs lossless test. And I'd like a neutral third party administering and monitoring the test please.


----------



## sonitus mirus

bigshot said:


> Yeah. Very nice.


 
Why 24 trials?  If there is something different that can clearly be identified, why not just start a new test and send us the 10/10?  It is clear from the results that something is identified after a few trials. Are the results of the test hidden until the test is completed?  I've seen other test results provided by Amir.  They aren't always 24 trials.  Why not?  That suggests to me that there is some learning involved that requires the listener to see the results of the testing after each decision.  If this is the case, and the listener is unable to display any statistical relevance that a difference is heard until after several trials, than how would these differences matter when listening to music normally?  

If the difference is so subtle, how subtle is too subtle?  I mean, what if a listener is able to identify a difference between 2 of the same brand and model of a DAC or even a cable of the same length and material?  Can anyone hear a difference in 2 properly working audio cables with the same properties made by the same manufacturer?  Where would that leave us if anyone could?


----------



## RRod

amirm said:


> That is exactly it: I am not worried about "*most of you*" (putting aside that you have no data that represents most of you -- just assumptions).  I am worried about *all of you.  *As is the research.
> 
> When we talk about what the music industry should release to us, we need to be inclusive of the needs of everyone, at all listening levels, for all content, and all situations.
> 
> ...



I'm actually behind all of that. I've actually said on this site before that there's no reason to deliver 16/44.1 any more. Just move CD prices to hi-res downloads and have cheaper alternatives for lossy downloads and streaming. But there are all kinds of imaginary effects of 24 bits, beyond noise-free reproduction, that are being used to justify a variety of products and prices on this site and others. Coming up with edge cases for the audibility of 16-bit truncation without-noise-shaping is the kind of thing that can easily be used for the wrong purposes.

I am down with loud music; after all, I can at a whim produce numerous albums where a 90dB SPL average level will results in 120dBSPL peaks. I'm also aware that in all my years of listening to such music, I've never been bothered by 16 bits.

I don't have foobar because I'm not on Windows; name an ABX platform you like for Mac and I'll happily pass the 24 vs 16-bit truncation example. I will also post the loudest section of that album for reference.


----------



## TheSonicTruth

sonitus mirus said:


> What was used to encode the MP3 file?  What was the source?  Fairly basic properties provided.
> 
> I was expecting something more like this:



Again, choices during mastering will have more audible impact than the fact it is a lossy format file.


----------



## RRod

If you want a file that you can ABX at 320k CBR, all you need is eig.wav. You don't even need to jack the volume!


----------



## sonitus mirus

jgazal said:


> I am going to feel very sad if the demand for immersive rendering of music fails as bad as stereoscopic television.
> Any cheer up music that would you recommend?



Have a listen to the 1963 jazz album "Idle Moments" from Grant Green.  The 1999 24-bit remaster from Rudy Van Gelder (RIP) is quite nice in just about any format.  Be careful around the 7:40 mark if you are listening at 129 dB peaks, as the saxophone may hurt your ears a little.


----------



## Arpiben

amirm said:


> Answering your question regarding level of my ABX test, I don't know how to answer that as I am using headphones and I have *no way of measuring levels* there.  I turned it up 'till I could hear what was there and did my ABX test.
> 
> *What was the purpose of the test* and can you run it and and post results of success or failure regardless of level?  I am trying to get us calibrated subjectively.
> .



The audio layman I am, is rarely getting above the following estimated levels with headphones:

80 dB SPL with average or high noise environment
65 dB SPL at night or low noise environment
I am well aware that those above levels settings are estimated taking into account the headphone SPL @1kHz specification and a 0dB FS True Peak Level.

At my level, @RRod's files helped me realized how much* futile* the actual dynamic range discussion is, since:

In order to discern sound in them I needed to crank up volume around +25dB,
In order to discern noise nuances I needed to add up to +15dB ( total +40dB) or more.
Checking those files with a digital Peak Meter (ITU BS 1770) indicates a True Peak Level around -49dB LKFS.
I will omit the RMS / Momentary / Short / Integraded values all below -60dB LKFS since values under -70dB are not taken into account by specs.

In conclusion, since I am not having an AutomaticTransmit (volume) Power Control reacting as fast a 100dB/s to appreciate noise subtilities while keeping safe levels for loudy portions, I will keep my volume settings steady or within a 10 dB range.
No matter if I am missing extreme low level subtilities in such tracks with 24bits or even 16bits.


----------



## gregorio

amirm said:


> [1] No, all you have to do is present the research. When you don't then it means you have not read such and just want to imply such data exists.
> [2] Noise is not music. And musical peaks are not averaged as we have been discussing.



1. It's already been presented in post #4740. Allowable exposure time for 139dB = 0.11 seconds per 24 hour period. Exposure to 140dB is not allowable.
2. Which is why I posted papers on hearing damage caused by MUSIC to musicians!!!



amirm said:


> I have shown you research from multiple peer reviewed journal research papers that use the identical methodology: finding threshold of detection and comparing with peak music in real life to determine dynamic range. This is it not a game. It is how to do this.



No, that is not how to do it!! Dynamic range is NOT the limit of hearing and then just measuring peak music in real life. If you want "real life", then dynamic range is from the noise floor of the concert/performance environment to the peak of that performance and obviously that environment MUST include musicians, audience and exterior and system noise, in the case of an outdoor and/or amplified gig. For the umpteenth time, your "real life" does not exist in real life and therefore cannot possibly be real life! Jeeeeez.



amirm said:


> [1] I have shown you noise floor of recording spaces that is at the threshold of hearing.
> [1a] And we have measurements of peak music in the same rooms in excess of 120 dB.
> 2. That makes the statement in yellow total folklore and nonsense.



1. You have shown us the noise floor of a few select recording spaces.
1a. Now that's a lie, you know it's a lie because you've been told and because simple logic should tell you and yet you just keep repeating the same nonsense. They are absolutely NOT peak measurements in the "same rooms", they are peak measurements in DIFFERENT ROOMS, the rooms have a whole bunch of musicians in them and for a real life concert, an even bigger bunch of audience! It wouldn't matter if the rooms have a -100dB SPL noise floor, what matters is the noise floor with the musicians!!! And, you are ignoring the peak and SNR levels of typical mics, which you yourself posted photos of. And, you are ignoring the target dynamic range of masters for consumers.

2. No, due to what I've stated in 1a, that makes what you are trying to state nonsense!



amirm said:


> Don't keep repeating that.



If only you could apply that sentiment to your own statements!

G


----------



## sonitus mirus

Arpiben said:


> The audio layman I am, is rarely getting above the following estimated levels with headphones:
> 
> 80 dB SPL with average or high noise environment
> 65 dB SPL at night or low noise environment
> ...



I could not hear anything with 2 of the files used in Amir's ABX at the volume level I typically listen to music.  My SPLnFFT app indicates <80dB(A) FAST at my normal listening level.


----------



## RRod

Arpiben said:


> The audio layman I am, is rarely getting above the following estimated levels with headphones:
> 
> 80 dB SPL with average or high noise environment
> 65 dB SPL at night or low noise environment
> ...



Thanks for giving a listen, @Arpiben. Yes, if you aren't accustomed to this kind of material, the volume settings will seem unreasonable. I'll post the loudest section of the track (happens to be the loudest section of the album too) and you'll see that you might be able to up the pot just a tad more.


----------



## RRod

Here's the loudest section of that album. By loudest I mean it contains the sections with the highest momentary and short-term loudness measured under EBU R128.


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## reginalb (Nov 27, 2017)

amirm said:


> What inability?  That the few people on the losing side of the argument declaring it so, doesn't make it that.  I mean the proof is in what you say: _none of you have bothered to go and buy the papers and read them! _ How little can you care about this topic when you have no motivation to go and spend the cost of one or two CDs to really understand the research?



What the what dude. You have perfectly encapsulated my entire point here. I did not buy it. I am not an audio engineer or researcher, so I trust others to do this for me. Others who HAVE READ THE ARTICLE offered you pointed, direct, and compelling contradictions, and your response is to throw logical fallacies back and never address the point. If you need help, this was a 2 part critique: 1. Data is an outlier, 2. they used a homebrew mic without specifications provided to get the outlier data. If that is true, then the paper is bunk. Period. Note the reason for this is that the data is an outlier, and when you come up with an outlier data point, then others must be able to replicate your work. This is a basic part of the scientific method. I gave you a link to a university's "How Science Works."

You could say that is not true - that they used a standard mic, that they provided the specs and others have replicated their work, etc., and that would be an actual refutation. By dodging the critique you clearly imply that it is true, and you're using obfuscation to cover up that fact. I have actually read a lot of AES papers, and they're almost universally really bad research, maybe it's all you're exposed to, and to be honest it's fine if you are. It's way more important that drugs are researched well, for example, than audio gear. But it's typically bad science. Knowing that, and hearing a legitimate critique of the paper you cite that you seem unable to debunk tells me what I need to know.

Feel free to actually address the claim now. But I am sure you'll instead point me to more biographies and tell me that I haven't read enough science and therefore am way less smart than you. Because you either are so arrogant that you don't care about properly defending your assertions, or they're indefensible.



amirm said:


> So that it is clear what the research states, *it goes thought *every bit of the chain* in both recording and playback to see what the effective dynamic range could be.  I*t examines everything from venue, to hearing thresholds, microphones, amps, DACs, ADCs, etc., etc. And it builds on other published work.  Furthermore, this research is hugely references in later papers by many others.  It simply is end of the story.



That it is referenced by others is meaningless. It could be referenced in papers that refute it, it's still referenced. It is not the "End of the story," and if you think it was then you don't actually understand how science works.



amirm said:


> As to whether I know how to interpret the data, this is my professional background: https://www.audiosciencereview.com/forum/index.php?threads/a-bit-about-your-host.1906/.  Bigshot walks around with an AES workshop link in his signature.  Did you know I hired one of those luminaries, J-J, to be my audio architect while at Microsoft?  This is what I did professionally.  I have the ability to understand the paper in question.  And have written articles on it that has gotten wide distribution without a single person writing in and saying, "oh, your author doesn't have the ability to understand the resaerch."  Now this may sound like bragging but it is not.   Your doctor is not bragging when he says that he understands research that you have not even read.  He can dismiss your protest out of hand and be right.



Cool, so this is what we call appeal to authority. I've pointed this out to you time and again, yet you _*still*_ go back to it. I do not care about your resume or who you've hired. It does not give you carte blanche, does not make you right, and your _repeated_ falling back on it demonstrates that you can't argue on the merits. You don't use appeal to authority if you can argue the merits of something. I get it, you were a VP at Microsoft, and I am therefore to grovel at your feet. I don't care because you've repeatedly demonstrated an inability to back your claims - and when your fallacies are pointed out to you by I and others, you use the _*same *_fallacies in your responses! I know lots of VPs, and I can tell you your titles don't impress me. Some people in authority never have to point out that they're in a position of authority.

When I was in the Army, you could tell a Ranger from a Green Beret, the Ranger always told you how bad and tough he was. The Green Beret just was. Never had to say it.



amirm said:


> As to what you all are saying, I* have heard it countless times. * I have debated them at length on other forums with people taking your positions but frankly, doing a lot more research than you all that are doing by protesting.  *What is in the OP is one of the most frequent myths spread on forums about dynamic range of our hearing and what is recorded in content. * It is time to stop it and not use debating tactics of "oh, don't you want our respect" or "you don't get it."  I get it and am not looking for the respect from people who don't bother spending a few bucks learning about audio science as published in real world and not talking points in forums.  Yes the "logic" of it makes lay sense.  It even makes semi-technical sense.  That is why people run with it without doing any research of their own.  But ultimately it is just gut feeling stuff that is not correct.



What _*I *_am saying is not what others are saying. What *I *am saying is that you keep using logical fallacies in all of your posts, which makes them unconvincing. I come to this and other forums to learn, mainly, because this is not the area where I am expert. I am trying to point out to you that your posts are unconvincing to someone like me because they're riddled with straw men, appeals to authority, and other logical fallacies.

 If you want me to make a point regarding audio, it's that there is plenty of dynamic range in CD's. I know this because I have a CD that, if I am listening to it at normal listening volumes, there is a section where suddenly a horn plays at such a volume that it causes me physical pain. Again, this is contained within the 16-bits of a CD. If that CD can fit within its dynamic range very quiet sounds, and also at the same volume on my amplifier it can _hurt my ears _then why would I need more? To hurt my ears more?

Instead of responding to this with your typical explanations of why I must take everything you say at face value because of who you are rather than the merits of your argument, or to promote my purchasing of papers for no other purposes than an internet argument, why don't you explain to me why I am mistaken. Why I need more bits than all the way up to physical pain from all the way down to extremely quite sections also contained on that CD. If you need to know what I am talking about, the one I always think of is on _Ella and Louis_, the tack is _Isn't This a Lovely Day, _around 4:15. I love this album and am so used to this part that I don't have to look up from my work, I instinctually reach over and turn down the volume as I approach this point in the song.



amirm said:


> Yes, we can push fidelity of audio way, way down.  An MP3 at 128 kbps will be transparent to vast majority of public and many audiophiles for that matter.  Better  yet, at highest level, a lossy codec can fool even the best of the audiophiles.  So the point you are trying to make is not in dispute in that sense.
> 
> What you are not considering is what I have said repeatedly: *a channel needs to be transparent for all people and all content.*  That is my standard of reference.  It is something we can achieve today.  It takes content that is not butchered and hardware implementations that are right.
> 
> ...



You have clearly demonstrated that both you don't know my position and knowledge, and that you have a wildly inflated sense of your own. For starters you seemed to have jumped from the topic, telling me that the OP is wrong (the OP being about 24-bit music, and you saying that the OP is a frequently stated myth) and now you're telling me about lossy compression, which I never mentioned once in this thread. Maybe you read me talking about the bit-rate of the Opus files on my phone - but I never brought that up to you, you're bringing that up now. As an aside, I'd love to see you in a double blind test of Opus vs your format of choice where there was an actual impartial referee to keep you honest.

For my own purposes, I don't care about ABX ability. I care about music and I find the topics here interesting - mainly because I was once duped by drivel when I knew even less than I do now about music formats - and I'm a bit of a gear head. I never brought this up, you've put those words in my mouth. All I did was to point out to you that you obfuscate every single argument made against your own. I suspect that is intentional.


----------



## bigshot

It's really kind of comical to  keep trotting back to the same paper over and over again when any of us can walk over to our shelf full of CDs and quickly find proof that redbook has ample dynamic range. It makes me wonder what kind of person would be taken in by an argument like this. Probably someone with very little technical knowledge so the appeal to authority might have weight, and someone with no real interest in listening to music so they wouldn't have personal experience to contradict the hairbrained theories. The kind of person who hires someone to put together a hideously expensive stereo system for them that looks impressive and never gets used much. I knew someone like that once. His speakers were over six feet tall and the only thing he had to play on them was Mannheim Steamroller albums.


----------



## WoodyLuvr

bigshot said:


> ...Mannheim Steamroller albums.


Just experienced a horrible 80s flashback of being tortured with Steamroller Christmas music...


----------



## reginalb

bigshot said:


> ...any of us can walk over to our shelf full of CDs and quickly find proof that redbook has ample dynamic range...



BUT I HAVE PRESENTED PEER REVIEWED RESEARCH WITH OVER 60 CITATIONS CONTAINED IN THE PAPER, YOU DIDN'T PROVIDE A CITATION FOR YOUR TRIP TO THE CD RACK!!!!!!!!!



bigshot said:


> ...The kind of person who hires someone to put together a hideously expensive stereo system for them...



Also, the kind that sell and install those systems. Just saying.


----------



## JaeYoon

WoodyLuvr said:


> Just experienced a horrible 80s flashback of being tortured with Steamroller Christmas music...


oh god no, I remember those days when working at those shops that played christmas music 24.7 after thanksgiving and for a whole month straight.

Would make any sane person go on a killing spree towards the north pole.


----------



## 71 dB

bigshot said:


> It's really kind of comical to  keep trotting back to the same paper over and over again when any of us can walk over to our shelf full of CDs and quickly find proof that redbook has ample dynamic range. It makes me wonder what kind of person would be taken in by an argument like this. Probably someone with very little technical knowledge so the appeal to authority might have weight, and someone with no real interest in listening to music so they wouldn't have personal experience to contradict the hairbrained theories. The kind of person who hires someone to put together a hideously expensive stereo system for them that looks impressive and never gets used much. I knew someone like that once. His speakers were over six feet tall and the only thing he had to play on them was Mannheim Steamroller albums.



Years ago around year 2000 when I was working in the acoustics lab of an university, there was a guy in our lab doing his "civil service" because he didn't go to army. He was into quality sound reproduction and knew a high end dealer who was selling a pair of Duntech Princess speakers (numbered 007A and 007B I recall), but the client wanted to hear those speakers in a good acoustic environment, so the civil service guy arranged him the opportunity to bring the gear to the listening room of our lab with excellent acoustics for the client to hear. The system:

Duntech Princess speakers
Bow Wizard CD player
Tact Millennium digital amp
Some "High end" cables and interconnectors. CD player connected to amp digitally.​
Needless to say the system sounded amazing in the listening room with excellent acoustics. I brought my best friend from another lab to the listening room to hear this system. We had fun when the seller of the system was listening to the system moving his head side to side while commenting:_ "Something is wrong! The sound isn't coming over!"_ Well, I found the system excellent and I spend the next weekend in the listening room listening to my favorite music (Alone! Nobody else bothered to use the opportunity). Elgar's The Apostles, Op. 49 (Boult) got a stunning big presentation! The only negative thing about this system was that the CD player was picky. I had trouble playing some of my discs, with were scratch free and played perfectly with my own cheapo Denon I had back then. Next monday the system was gone and I don't even remember what the client thought about the speakers.


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## castleofargh (Nov 28, 2017)

amirm said:


> You guys keep using this "vast majority of situations."  Where is the data to back it?  How hard have people/OP looked?  Who here knew about he research I have been presenting?
> 
> Do you know that peer review reports of recent tests show detection of downsampled high-resolution music?  Like this: https://audiosciencereview.com/forum/index.php?threads/high-resolution-audio-does-it-matter.11/
> 
> ...


my idea of peak amplitude is based on a voltage RMS and a device of known sensitivity. how much time the signal will spend at 0dB depends on the music, not on me. but you're half right about me not knowing how loud it is with a proper reference, I never test for anything past 105dB, only rarely listen to music that will peak at 90dB, and I've started to bring my earplugs even to the theater nowadays just in case it's one of those movies.
as you have the er4sr, it comes with a custom certificate giving you the loudness to expect at 1khz into 0.2v. so it's not too hard to work it backward by measuring the voltage output if you don't have a calibrated mic to use IEMs. as a full doubling of the voltage is only +6dB, you don't even have to bother measuring the amp into the proper load. the potential change should still get us real close to the actual value(at least just as close as pushing the IEM further down the ear canal).

I say game because you take factual information obtained under hyper specific conditions, and decide that it now applies to consumers listening to music on a playback system. we're not contesting the papers(at least not all of them), we're contesting your tendency to pick an extreme and decide that the conclusions are systematically relevant for music listening. I'm sure you're well aware of the risks when taking conclusions outside of the conditions used to obtain them.
I guess I can put my issues with you under 2 categories:
1/ extreme circumstances that could apply to music listening but are very unlikely to actually happen with normal listening of random albums.
2/ extreme hearing thresholds obtained outside of music listening conditions.

for 1/ our disagreement is about how inclusive of all the possible circumstance you want to be. we don't disagree on the test results or how more than 16/44 is necessary for total audible transparency under all possible circumstances(including crazy listening habits and non musical content).
for 2/ now I believe you're just wrong in your approach. wrong because you're not proving anything when taking conditions that aren't music listening. and wrong because for most variables, it is easy to achieve one or 2 orders of magnitude better so long as we use a nominal test signal under nominal conditions instead of actual musical content. and that can be demonstrated for most variables I believe. a quiet test signal is not going to be as audible alone as it will be mixed with music. so I feel that it is wrong to just assume a listening test is a listening test is a listenin...
  and no, people shouldn't have to listen at 120dB peak to make a point about audible transparency. many people here spend more than an hour a day listening to music, many have no idea how loud they're going, when you say they don't come here to be told how loud to listen, I again disagree energetically. I imagine that many people are hurting their ears and aren't aware of it. posting ideas about how live event levels are fine for everyday use at home is not alright. this is a pubic forum, not AES, people know what they know and need to be constantly reminded of the potential dangers of listening too loud for too long. in your mind you have that one crazy dynamic track with crazy loud passages and crazy quiet passages, but what happens when the next track has some Justin bieber level of compression and the average SPL level is at 110dB? and what happens when the same guy spent 40mn with his IEMs in the morning and same coming back at the end of the day, with music pushed to cover the noises, is it still fine to enjoy a full album set with peaks at 120dB? it's a crappy idea to suggest that it is alright. and as far as I know, ideal listening conditions (aside from specific dynamic stuff), is around 60-65dB before the stapedius muscle contracts. I have no paper on that, it's just something I've read a few times as a suggestion for critical music listening, and tend to agree with it based on my very subjective impressions.  just like the idea that listening to flat speakers let us perceive the most of a track. we can think of many counter examples of course, but in general I suspect it is a correct assumption.

as for highres, I'm with @RRod, if it's the same price I'm fine with all resolutions and all formats and all kinds of dither. when I want smaller stuff I make them myself and everybody is happy.

 Gregorio mentioned 60dB as something we would almost never pass on recorded albums. I have a few tracks going beyond so it's not a strict limit for sure. but it was never intended to be, as some random guy can always go and make a CD with silence and then 0DB signals. so I don't see the point of taking his sentence too literally. I believe the most I've found was near 70dB when I was looking for that specifically one or 2 years ago in my library. and of course most of what I have, contains far less variations. the quiet passages are often 30dB below the average loudness which is logical as it's enough to feel quiet yet still clearly audible. so it makes sense to me to have albums made that way for the average consumer to enjoy it without having to become a human compressor holding the volume knob throughout the song. anyway, with my anecdotal little life, I feel inclined to agree with his number despite having found more myself. not as an absolute, but as a realistic expression of what we're likely to encounter.


----------



## Niouke

mom there's an evil person on the internet that wants to put 120dB in my ears !! Please don't, I can barely stand 90dB as it is...and most classical music is already uncomfortable to me as the dynamic range is just too large already...I'd rather have more compression


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## reginalb

Niouke said:


> mom there's an evil person on the internet that wants to put 120dB in my ears !! Please don't, I can barely stand 90dB as it is...and most classical music is already uncomfortable to me as the dynamic range is just too large already...I'd rather have more compression



Have you ever listened to a very dynamic recording with IEMs? Getting hit with a trumpet suddenly can be...not fun. 

And yeah, you can damage your hearing, if you love music, why would you want to diminish your ability to listen to it?


----------



## Niouke

I was NOT being sarcastic I hate very dynamic records. I used to be a violin player and the idea of covering the full dynamic range of the instrument from a close position in a single track would be interresting for a demo, but no way I'd listen to that on a daily basis.


----------



## reginalb

Niouke said:


> I was NOT being sarcastic I hate very dynamic records. I used to be a violin player and the idea of covering the full dynamic range of the instrument from a close position in a single track would be interresting for a demo, but no way I'd listen to that on a daily basis.



Ah, sorry for the misunderstanding, the line about mom and the evil guy got me.


----------



## Darren G

The really short version is, are you someone who has stopped enjoying music because you are listening for flaws, limits, listening to gear, or...  are you going to enjoy so much amazing music that is clearer, and more dynamic, than anytime in the past (short of a live event)?

I don't think anyone is arguing we must limit technology.  What they are saying is consumer level reproduction is good! very good!   At any reasonable listening level it's more than good enough!


----------



## WoodyLuvr

Niouke said:


> I was NOT being sarcastic I hate very dynamic records. I used to be a violin player and the idea of covering the full dynamic range of the instrument from a close position in a single track would be interresting for a demo, but no way I'd listen to that on a daily basis.


One of the former professional classical percussionists I mentioned in an earlier post always tells me that the very thought of a long drum roll makes his stomach turn and gives him a headache (à la motion sickness).  He refuses to listen to any dynamic or loud music and has happily limited his listening to very quiet ambient new age and electronic works with a careful eye on the volume (aka Brian Eno).  Surprisingly (to me) he finds custom in the ear headphones to be a safer and more controllable option than closed or open backs and especially speakers.


----------



## amirm

gregorio said:


> 1. It's already been presented in post #4740. Allowable exposure time for 139dB = 0.11 seconds per 24 hour period. Exposure to 140dB is not allowable.


I don't see anything from you n post 4740.  Regardless, I addresse that in detail here: https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-314#post-13875323







"Should not" is not the same as "now allowable."  And it is for impulsive or impact noise, not music.






And it is for averaged SPL level, *not peak.  *I don't know what it takes to get the message across that average and peak numbers are not the same.  

Occupational noise measurements also have other characteristics unique to them like exponential drop-off, polar response, etc.  See ANSI S1.4-1979.

Really, this is at the heart of what is wrong with your first post.  It confused average levels and intuition regarding noise exposure levels and confuses that for what is actually recorded in music files.  It is also blind to psychoacoustics and what a just noticeable noise level is from speakers in real rooms and playback levels in live venues.  Continued confusion with these posts regarding noise levels shows that nothing is learned and science is dismissed out of hand.


----------



## TheSonicTruth

WoodyLuvr said:


> One of the former professional classical percussionists I mentioned in an earlier post always tells me that the very thought of a long drum roll makes his stomach turn and gives him a headache (à la motion sickness).  He refuses to listen to any dynamic or loud music and has happily limited his listening to very quiet ambient new age and electronic works with a careful eye on the volume (aka Brian Eno).  Surprisingly (to me) he finds custom in the ear headphones to be a safer and more controllable option than closed or open backs and especially speakers.



What kind of drum roll are we referring to?  The kind in the circus right before someone jumps off a high ladder into a net?
I personally do not find that typical drum roll to be either very loud or all that dynamic.


----------



## amirm

sonitus mirus said:


> Where is the link to your scientific, peer reviewed paper verifying your assumption?


Well played.   I take that back!  Why are you guys here?  Worried as to whether there is real resolution increase in 24 bit audio or worried about what damages your hearing?  If the latter, what is with the title of this thread and OP?


----------



## amirm (Nov 28, 2017)

71 dB said:


> If an recording really needs (it doesn't, because 125 dB dynamic range in consumer audio is ridigulous idea) 125 dB, then shaped dither can do it, you see it in the picture. Are we done? Is there any hope in out future that you understand why 16 bit is enough in consumer audio?


The picture?  *This is the picture.  *In words that is, too lazy to draw some.  

1. Almost all music recorded in 24 bits. 

2. Conversion to 16 bits (without dither and noise shaping) chops off 8 bits of that.  Let's be generous and say that half of that was noise anyway.  So that leaves us with 4 bits thrown away.  That represents 4*6 = 24 db increase in noise floor.

3. We are then told to use noise shaping to push that 24 db of extra noise to somewhere else.  In 44.1 Khz, there is no place left to properly push that unless you cherish driving your tweeter and amp harder in 20 to 22 Khz.  Best to increase the sample rate.  But oh wait, we can't do that since you advocate sticking with 44.1.

But let's say we did use noise shaping.  What that translates into is that we *took perfectly good 24 bit music file, added 24 bits of noise to it, and then applied a transformation to that noise.  * Why do we want to do that?  *Why not just give the original 24 bit file to people who want it?*

You see what the problem with the whole argument is?  We had to jump through those hoops to get better performance out of CD which has fixed specification of 16/44.1.  *With digital distribution, that restriction is gone, gone, gone. * There remains no reason to apply that signal processing anymore unless you are trying to distribute on CD. 

This is on top of the fact that *it is not up to you and I to apply noise shaping.  *It is up to the entire content production industry, vast majority of which have no background in signal processing and you will be lucky if you get dither for conversion of 24 to 16 let alone noise shaping.  The whole concept of adding "noise" sounds like a bad thing to people without signal processing so it is not surprising that proper conversion is not assured whatsoever from 24 bit files.

So in summary, conversion to 16 bits from 24 bits is a form of lossy compression.  Noise shaping makes it a perceptual technique so no different than lossy codecs.  We no longer have a need for such a conversion.  *Give the bloody 24 bit files and let's be done with it!  No one has stipulated a single reason why it is good for me or anyone else to not have the option of getting 24 bits delivered to consumers. 
*
That is how we get "done" with this conversation, not with hopes and dreams of the world all of a sudden deciding to noise shape dither all 24 biles.  And even if they did, it would be a solution looking for a problem.

Really folks, physical formats that forced these things on us are gone.  We have total freedom.  Let's offer the consumer the *option to download music at originally captured rates.  *Technology and big companies are no longer a barrier to us doing so.  Let's celebrate our newly found freedom! I know I am....

Edit: fixed typo of "bits" instead of "db."


----------



## amirm

Darren G said:


> The really short version is, are you someone who has stopped enjoying music because you are listening for flaws, limits, listening to gear, or...  are you going to enjoy so much amazing music that is clearer, and more dynamic, than anytime in the past (short of a live event)?


I do enjoy music at all fidelity levels.  When in the car I stream lossy audio and when my favorite tracks come, I enjoy them immensely.  

But that is only a part of this hobby.  This entire forum exists, including these long threads because people are interested in more than enjoying music.  They like to learn about the technology, argue about it, and feel that when they purchase something, it is better than the alternatives.  And yes, they all look for flaws when shopping for gear and routinely when they own them.  It is what the hobby is.  The day you stop that, is the day you become a music lover but not an audiophile.

That said, it is true that once you become a critical listener, you will be bothered more in system performance flaws.  As audiophiles we are already there compared to general public.  So that is not anything new either.



> I don't think anyone is arguing we must limit technology.  What they are saying is consumer level reproduction is good! very good!   At any reasonable listening level it's more than good enough!


Sure, consumer level reproduction is good, very good.  What does that have to do with the OP saying higher resolution is a myth and that classical music can have 12 db of dynamic range?  And that this is so as a matter of audio science?

Let's go back and change the OP to what you say and we will truly be done.


----------



## reginalb (Nov 28, 2017)

amirm said:


> ...And it is for impulsive or impact noise, not music...



You keep repeating this without source. And you hammer everyone else for not sourcing their information.

How is music any different from other types of noise? Do you deny that there is a high rate of hearing loss in musicians? Do you think somehow that music magically doesn't harm your hearing like other sounds, as if the waves are made up of something more pillowy to your eardrums?



amirm said:


> ...So in summary, conversion to 16 bits from 24 bits is a form of lossy compression.  Noise shaping makes it a perceptual technique so no different than lossy codecs.  We no longer have a need for such a conversion.  *Give the bloody 24 bit files and let's be done with it!  No one has stipulated a single reason why it is good for me or anyone else to not have the option of getting 24 bits delivered to consumers....*



What many of us would (and do) argue against, is the fleecing of consumers by trying to convince them that there is some magic in that extra 8-bits when there isn't, and then charging them more for it. 

Here is an example: 
Macy Gray - Stripped (I use this since I checked on prices recently and there are a bunch of options where to buy it)
Vinyl at Acoustic Sounds: $30
HD Tracks 24/192: $25
HD Tracks 24/96: $18
Amazon CD: $14
7Digital 16/44.1: $12.50
HD Tracks 16/44.1: $12
7Digital MP3: $10.50
iTunes: $9.99 
Amazon MP3: $9.50

Or included with monthly streaming services. 

Now you, as well as a lot of people with skin in the game, make fantastic claims about the vast differences between these files, while in the real world people aren't going to hear the difference between them. There is profit to be made off of deceiving the public in to believing those HD Tracks versions contain some special sauce in them. Someone, for example, who founded a company that designs home audio systems for poor saps who they are able to con out of their money.

With training, people could probably hear slight differences, or they could turn the volume up during quiet sections and hear the noise floor. Or they could be saved from fleecing and be happy listening to the music.


----------



## JaeYoon (Nov 28, 2017)

Well if that is the case, I never was an audiophile to begin with, I want to love the voices of singers I like. I want to love sounds of the instruments that bands play.

I don't want to love my equipment. I mean I do use DAPs and my IEMs, but I don't want to worship those on a shrine.


----------



## amirm

reginalb said:


> What the what dude. You have perfectly encapsulated my entire point here. I did not buy it. I am not an audio engineer or researcher, so I trust others to do this for me. Others who HAVE READ THE ARTICLE offered you pointed, direct, and compelling contradictions, and your response is to throw logical fallacies back and never address the point. If you need help, this was a 2 part critique: 1. Data is an outlier, 2. they used a homebrew mic without specifications provided to get the outlier data. If that is true, then the paper is bunk. Period. Note the reason for this is that the data is an outlier, and when you come up with an outlier data point, then others must be able to replicate your work. This is a basic part of the scientific method. I gave you a link to a university's "How Science Works."



You could say that is not true - that they used a standard mic, that they provided the specs and others have replicated their work, etc., and that would be an actual refutation. By dodging the critique you clearly imply that it is true, and you're using obfuscation to cover up that fact. I have actually read a lot of AES papers, and they're almost universally really bad research, maybe it's all you're exposed to, and to be honest it's fine if you are. It's way more important that drugs are researched well, for example, than audio gear. But it's typically bad science. Knowing that, and hearing a legitimate critique of the paper you cite that you seem unable to debunk tells me what I need to know.

Feel free to actually address the claim now. But I am sure you'll instead point me to more biographies and tell me that I haven't read enough science and therefore am way less smart than you. Because you either are so arrogant that you don't care about properly defending your assertions, or they're indefensible.[/QUOTE]
There is a lot of protests in there but no information to share on this topic.  The information you ask for was provided but seemingly not read.  Here it is again:





Microphone information is provided down to model number.  And there is no attempt at finding one or two data points.  It is an extensive survey, and as much as anyone could ask in determining the collective knowledge.  And this is the summary of it:





I post all of this before.  I have seen no one demonstrate that they have read any of this.  Nor have they, like you, read it now.

As you see, there are no logical arguments here.  Only data and research.  It has nothing to do with me being smart.  It have simple read the research and combined this with the work I used to do professionally. 

You are not interested in learning if the first thing you do is say some research is bad.  *What AES papers have you read anyway?  Can we see a list of a few of them?*


----------



## amirm

reginalb said:


> You keep repeating this without source. And you hammer everyone else for not sourcing their information.


I provided the source. Here is the image of the original post with the link now highlighted in yellow:


----------



## TheSonicTruth

amirm said:


> The picture?  *This is the picture.  *In words that is, too lazy to draw some.
> 
> 1. Almost all music recorded in 24 bits.
> 
> ...



Using my profile pic as an example: When down-converting from 32 or 24bit to 16, is that what happens to the audio?  In other words, the left hand waveform represents the 24bit session take, and the right hand represents the final 16bit master?


----------



## amirm

reginalb said:


> How is music any different from other types of noise?


It all reads to intent of research and how data is gathered.  And also the application.  A gunshot makes the noise it wants to.  We don't listen to 10000 gun shots in a row and call it music.  Yet that maybe exactly what an employee at a gun range be exposed to.  

Surely we agree that if live performances resembled occupations noise in constructions or factory floor, people would not go pay and sit there to listen to them for one hour or more.  



> Do you deny that there is a high rate of hearing loss in musicians? Do you think somehow that music magically doesn't harm your hearing like other sounds, as if the waves are made up of something more pillowy to your eardrums?


No, but none of that is at play here.  What is at play is confusing how noise levels are measured in occupational safety (averaging, weighted, etc.) vs what peaks may occur in real life concerts.  You need to disassociate the two numbers.  Just because they both use "db SPL" it doesn't mean they are the same.  No more than thinking average and peak value of a set of numbers are the same.


----------



## reginalb (Nov 28, 2017)

amirm said:


> You could say that is not true - that they used a standard mic, that they provided the specs and others have replicated their work, etc., and that would be an actual refutation. By dodging the critique you clearly imply that it is true, and you're using obfuscation to cover up that fact. I have actually read a lot of AES papers, and they're almost universally really bad research, maybe it's all you're exposed to, and to be honest it's fine if you are. It's way more important that drugs are researched well, for example, than audio gear. But it's typically bad science. Knowing that, and hearing a legitimate critique of the paper you cite that you seem unable to debunk tells me what I need to know.
> 
> Feel free to actually address the claim now. But I am sure you'll instead point me to more biographies and tell me that I haven't read enough science and therefore am way less smart than you. Because you either are so arrogant that you don't care about properly defending your assertions, or they're indefensible.
> There is a lot of protests in there but no information to share on this topic.  The information you ask for was provided but seemingly not read.  Here it is again:
> ...



Man you're a jerk. Seriously, what is your problem?

For the actual content, THANK YOU. You finally took something and actually responded accordingly. This is the first post I've read where you actually directly addressed something, and it was actually enlightening. I learned what I already know, peak levels at rock concerts are WAY too loud. I wear hearing protection at rock concerts. I apologize if I missed the information about the mic, but when this direct criticism was levied you absolutely DID NOT present this in response. You literally responded by saying that the paper has 63 citations at the end (meaningless), and then the author's bio (also meaningless).

I wish they would have chosen a different shape (or at least two that are more discernible in reproduction if they were different) for jazz and classical.

I started my responses to you asking civil questions, and you immediately sniped at me, and have repeatedly told me what I'm arguing is wrong in instances where I've literally not argued anything at all.

With regards to bad research, I'm not going to go down the list, but there are a ton of pseudo science papers published in this field, and quite a few in that particular paper. Typically low sample sizes, poor documentation of methodology, and questionable conclusions have been the hallmark of papers I've read in this community. 

Or a recent meta-analysis that used debunked papers to show that people can definitely hear the difference between hi-rezzzzzz and other formats. 



amirm said:


> I provided the source. Here is the image of the original post with the link now highlighted in yellow:



Which is, of course, not what I said you didn't cite. That citation has nothing in it about how "impact" noise as you call it, is different than music in terms of its affects on long term damage to your ears.



amirm said:


> It all reads to intent of research and how data is gathered.  And also the application.  A gunshot makes the noise it wants to.  We don't listen to 10000 gun shots in a row and call it music.  Yet that maybe exactly what an employee at a gun range be exposed to.
> 
> Surely we agree that if live performances resembled occupations noise in constructions or factory floor, people would not go pay and sit there to listen to them for one hour or more.
> 
> ...



Again, you're missing the point. You've come in and asserted that you won't damage your hearing at sound levels that are equivalent to those, because the sounds are different. But you haven't backed this claim up. Depending on the weapon, a gunshot can be nearly 200dB, do you think that 200dB music, even if it's just a brief peak, wouldn't hurt your ears?

My ears ring at the end of a concert if I don't wear hearing protection. Just the same as they ring when I shoot a weapon without hearing protection. I've done the latter only once, even with 7 years in the military, because I care about my hearing.


----------



## TheSonicTruth

reginalb said:


> Man you're a jerk. Seriously, what is your problem?
> 
> For the actual content, THANK YOU. You finally took something and actually responded accordingly. This is the first post I've read where you actually directly addressed something, and it was actually enlightening. I learned what I already know, peak levels at rock concerts are WAY too loud. I wear hearing protection at rock concerts. I apologize if I missed the information about the mic, but when this direct criticism was levied you absolutely DID NOT present this in response. You literally responded by saying that the paper has 63 citations at the end (meaningless), and then the author's bio (also meaningless).
> 
> ...



"_My ears ring at the end of a concert 
if I don't wear hearing protection._"


Then something is not being done correctly at that concert if it's that loud.  It's not a Nascar race.


----------



## bigshot (Nov 28, 2017)

TheSonicTruth said:


> What kind of drum roll are we referring to?  The kind in the circus right before someone jumps off a high ladder into a net?
> I personally do not find that typical drum roll to be either very loud or all that dynamic.



That's because it's halfway across a circus tent from you. Put it at the end of your arms and it's an entirely different story.




amirm said:


> 1. Almost all music recorded in 24 bits.



I'm just going to address this one because it's all I need to address...

Music is recorded with 24 bits of dynamic range so there's plenty of latitude to adjust levels in the mix. But the final mix doesn't include even a fraction of that dynamic range because all sorts of compression is applied to make the music comfortable to listen to. This is a good thing. Too much dynamic range is irritating and forces you to keep reaching for the volume control to turn up quiet parts to make them audible, and turn down loud ones so they don't blast your ears. This compression can be electronic in the form of compressors and limiters, or it can be acoustic in the form of miking from a distance and allowing hall ambiences to soften the dynamic peaks. This process is the same for pop music, jazz and classical... just to different degrees. All music is balanced to sit comfortably within the ear's natural dynamic range of about 45 to 50dB at a time. That's why LPs can have a dynamic range of less than 50dB and still sound good. That's why we never hear the noise floor of a CD without jacking the volume control on the fade outs. Music is mixed to sound good to ears. It isn't mixed to conform to abstract numbers relating to the extremes of human perception. If you want to recreate the sound of a jackhammer turning on and off in the depths of Carlsbad Caverns, then you are going to need 24 bits to do that. But if you want to recreate the sound of a symphony orchestra or rock band, redbook is perfectly capable of doing that. In fact, redbook has more dynamic range than you need, and that is proved by some overly dynamic recordings on the BIS label that are a chore to listen to.

CD sound is all you need. See the link in my sig.

That said, I have a lot more respect for people who make and record music than I do people who just enjoy posting in forums about recording quality. It's all about the music. That is all. Do not pass Go. Do not collect $200. Music is the purpose of all this. The equipment is just the means to the end. Great music with great sound is great. But great music doesn't get any greater by including frequencies only bats can hear and noise floors that reveal the heartbeat of the guitar player between songs.



amirm said:


> I have actually read a lot of AES papers, and they're almost universally really bad research



If you want to participate in peer review you have to work to become a peer. It isn't enough to be an armchair quarterback. You have to get out there and actually produce something. I think if you were more involved in the career of music recording, you might have more understanding of the way music should be reproduced and perceived. Those aren't totally separate things. They're inter-related on a million different levels. It would help if you listened to other people and made an effort to understand what's being said to you. It would also help if you stuck to honest discourse without letting your ego make you wander into logical fallacies and argumentative tactics that obscure the truth instead of revealing it. Just a suggestion. Feel free to ignore it if you want.


----------



## reginalb (Nov 28, 2017)

Looking at Amirm's graph here, I think I finally have a good grasp of what he's arguing and that it's impractical at best.







The key shows what on close inspection appear to be an open circle for classical, an open box for jazz, an open triangle for rock, and upside down triangle for others.

Then they're filled in if there is electronic augmentation.

In the section above 120dB there are I believe 6 rock concerts, 5 "Others", 3 or 4 jazz and 0 or 1 classical.

I include this summary because that is one hard chart to read and welcome corrections to what I'm seeing here.

I guess the argument is that we are pretending that noise shaping is not a thing, and thus 96dB is what we have to work with, and 129-96 is only 33dB, which isn't low enough and therefore the full dynamic range of 24-bits is necessary. So if I listen to my music so loud that my ears ring when I leave (which they do at the average rock concert) and somehow I'd hear the noise floor over the ringing in my ears, then I would need 24-bits (assuming that noise shaping doesn't exist, which it totally does) for proper reproduction of sound. All this because it has to be "perfect," and by "perfect" we mean, "Damaging to our ears." And again, I don't think I'd hear the noise floor at all, *because my ears would be ringing.*

And yes, Amir, it would damage our ears. Exposure to sound at the level that causes temporary hearing loss will, over time, cause long term damage to our ears. Since you're obsessive about citations, here you go:
http://www.jneurosci.org/content/29/45/14077.short

Here's some more reading on the affects of levels as low as 96dB average (in clubs) on DJ's, listening to music mind you, not "impact noise" which you seem to feel is much more harmful than musical noise:
https://www.cambridge.org/core/jour...n-nightclubs/573BAE71AFF87AB06570E5991E5AC6CA


----------



## reginalb

TheSonicTruth said:


> "_My ears ring at the end of a concert
> if I don't wear hearing protection._"
> 
> 
> Then something is not being done correctly at that concert if it's that loud.  It's not a Nascar race.



I agree completely, but it is a near universal experience for me unless I wear hearing protection (which I do). My wife likes to be near the speakers (she is actually hard of hearing) so that has something to do with it, I'm sure. But if you need more proof, just look at those peak levels from Amir's citation. They're insane.


----------



## RRod

reginalb said:


> Looking at Amirm's graph here, I think I finally have a good grasp of what he's arguing (and that it's insane).
> 
> 
> 
> ...



Are there rock concerts where the ambient noise levels are much around 33dBSPL? Also, what are the crest factors here. I can believe a 130dB peak if the crest is 40dB, but do I really expect that at a rock concert? The numbers just add up to way too high an average listening level to be at 'optimal' positions, and even then, there is crowd noise...


----------



## bigshot

TheSonicTruth said:


> "_My ears ring at the end of a concert
> if I don't wear hearing protection._" Then something is not being done correctly at that concert if it's that loud.  It's not a Nascar race.



The level of attention to response curves in amplified concert venues is often dismal. They sometimes depend too much on automated calibration that leaves big narrow spikes of level imbalances in the center of the most sensitive hearing range. If the spike is high enough and narrow enough, it isn't very audible. But it can wreak havoc on your eardrums. I've experienced this too many times at large rock shows, especially at outdoor arenas where they don't figure they need to tightly control the response.


----------



## amirm

reginalb said:


> For the actual content, THANK YOU. You finally took something and actually responded accordingly. This is the first post I've read where you actually directly addressed something, and it was actually enlightening. I learned what I already know, peak levels at rock concerts are WAY too loud. I wear hearing protection at rock concerts. I apologize if I missed the information about the mic, but when this direct criticism was levied you absolutely DID NOT present this in response. You literally responded by saying that the paper has 63 citations at the end (meaningless), and then the author's bio (also meaningless).


Rock concerts are but one category represented in that research.  And I provided the same answer a week ago: https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-306#post-13864815

This is from the same post where I talked about the 67 references which sounds like you had read and hence your complaint about it:











We have a Jazz unamplified sample nearing 130 db SPL.  



> I wish they would have chosen a different shape (or at least two that are more discernible in reproduction if they were different) for jazz and classical.


I have quoted text which makes much of this clear.  Again from the above post:






And from the paper I quoted earlier from *other research* than Fielder's






As you see even solo piano can have peak level of 103 db SPL.  So please don't twist the data into it being about rock concert.   It is improper to spin data this way.


----------



## amirm

reginalb said:


> With regards to bad research, I'm not going to go down the list, but there are a ton of pseudo science papers published in this field, and quite a few in that particular paper. Typically low sample sizes, poor documentation of methodology, and questionable conclusions have been the hallmark of papers I've read in this community.


You read in the "community?"  In other words you personally have not read of the AES Journal papers?  Just going by what is said on forums?  And based on that damned everything published by J.AES, J. ASA, etc.?


----------



## reginalb (Nov 28, 2017)

amirm said:


> You read in the "community?"  In other words you personally have not read of the AES Journal papers?  Just going by what is said on forums?  And based on that damned everything published by J.AES, J. ASA, etc.?



Nope, I've read papers that don't cost money to access. Which would be silly for me to do given that I don't work in audio. It would be a waste of money.



amirm said:


> Rock concerts are but one category represented in that research.  And I provided the same answer a week ago: https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-306#post-13864815
> 
> This is from the same post where I talked about the 67 references which sounds like you had read and hence your complaint about it:
> 
> ...



103 dB would be easily contained within the dynamic range of CD, even without noise shaping....

I didn't twist the data, I literally counted out each genre and the number that were over 120 dB. How about you stop misrepresenting everything I say?



bigshot said:


> The level of attention to response curves in amplified concert venues is often dismal. They sometimes depend too much on automated calibration that leaves big narrow spikes of level imbalances in the center of the most sensitive hearing range. If the spike is high enough and narrow enough, it isn't very audible. But it can wreak havoc on your eardrums. I've experienced this too many times at large rock shows, especially at outdoor arenas where they don't figure they need to tightly control the response.



I have personally found small venues that are essentially bars to be the worst offenders. But have had the problem at outdoor venues as well.


----------



## amirm

RRod said:


> Are there rock concerts where the ambient noise levels are much around 33dBSPL?


33 db SPL what?  I have explained so many times that when it comes to minimum noise level, you must examine the spectrum.  https://audiosciencereview.com/forum/index.php?threads/dynamic-range-how-quiet-is-quiet.14/






But no, I don't know what the SPL level is at some random concert.  The studies I have shown you represent the noise level in the same halls where peak measurements were performed such as Davies hall above.


----------



## bigshot (Nov 28, 2017)

amirm said:


> As you see even solo piano can have peak level of 103 db SPL.



If you shove the microphone right up next to the hammers with the lid down it would definitely register 103dB in full forte. After all the piano is a percussion instrument. It's capable of loud transients. But that isn't generally how you mike a piano. You normally want some of the woody body of the sounding board. Usually I've seen pianos miked with the lid closed a little above the performer's head at 45 degrees down, or with the lid up at a distance of a few feet from the harp to the side. This takes the curse off the loud percussive "thump" and gives a more musical sound. Miked this way, I doubt if the peaks would get much above 85dB. Ideally the performer would modulate his dynamics with pedaling and touch to keep within a comfortable level and wouldn't necessarily play balls out (Jerry Lee Lewis excepted of course!) This is another case where the instrument is capable of producing a very loud volume in theory, but when it comes to practice, music just doesn't call for that sort of thing because it's uncomfortable to listen to.

In a live performance in a concert hall, the room ambience would suck up a lot of the dynamics. The audience would be getting something in the range of normal music... a dynamic range of 50dB or so. The idea of recorded music is to simulate that sort of experience, so the various microphones would be mixed together at levels that give a pleasing dynamic, even though the natural dynamics close miked would be off the chart.



reginalb said:


> I have personally found small venues that are essentially bars to be the worst offenders. But have had the problem at outdoor venues as well.



oh yeah! tight enclosed space... sloppy work at the board... a band that is used to playing larger venues... that's a recipe for painful ears.


----------



## RRod

amirm said:


> 33 db SPL what?  I have explained so many times that when it comes to minimum noise level, you must examine the spectrum.  https://audiosciencereview.com/forum/index.php?threads/dynamic-range-how-quiet-is-quiet.14/
> 
> 
> 
> ...



The thrust of my post was questioning the levels of the rock data given an optimal listening position. I understand that the 130dB peak for orchestral material would probably be under more stringent noise conditions.


----------



## sonitus mirus

amirm said:


> We have a Jazz unamplified sample nearing 130 db SPL.



$10 is a lot to me.  I don't have money to burn to spend on an AES subscription.  I appreciate your willingness to present some of the data from papers that I would not otherwise have an opportunity to read.  How often did these peaks occur and for what duration?  Is there any evidence that these occurrences were audible?  I don't need an out of context example, I mean specifically in the situations where these peak levels were recorded?

Nice to see that a majority of the classical genre has peaks that appear more reasonable.There were no occurrences recorded over 120 dB with any of the classical concerts that were measured.  90 dB peak?  Was it some elementary school student's solo?


----------



## 71 dB

Almost 5000 posts in this thread. In my opinion the first three pages of discussion board threads are the most fruitful and after that they seem to always become endless fights were people are hitting each other from their foxholes. For me 16 bit dynamic range has always been enough and will be in the future. Those who for some reason suffer from the noise floor of 16 bit audio have 24 bit downloads. Maybe they are even offered 64 bit files in the future and have nearly 400 dB of dynamic range to listen to exploding stars at realistic levels in the silence of Skywalker Scoring Stage. The music I want to ever listen to (at my noisy _home_, not at Skywalker Ranch) fits in 16 bits just fine. Heck, I'd manage with 14 bits only!


----------



## gregorio (Nov 29, 2017)

castleofargh said:


> I've started to bring my earplugs even to the theater nowadays just in case it's one of those movies.



The maximum peak in the middle of a cinema is going to be about 115dB SPL. However, that assumes the standard calibration level (called "Dolby 7.0") and a vast number of cinema screens actually use a lower level setting due to audience complaints. Dolby 5.5 is very common and results in peak levels in many cinemas being lower than 110dB SPL.



castleofargh said:


> Gregorio mentioned 60dB as something we would almost never pass on recorded albums. I have a few tracks going beyond so it's not a strict limit for sure. but it was never intended to be ...



Absolutely! There are literally millions of commercial recordings, the vast majority have a dynamic range of around 50dB or less. A tiny fraction have a dynamic range of 60dB and of those, a very tiny fraction have a dynamic range greater than 60dB. I don't believe any one knows for sure but possibly only a handful of those millions.



amirm said:


> 1. Almost all music recorded in 24 bits.
> 2. Conversion to 16 bits (without dither and noise shaping) chops off 8 bits of that.  Let's be generous and say that half of that was noise anyway.  So that leaves us with 4 bits thrown away.  That represents 4*6 = 24 db increase in noise floor.
> [4] But let's say we did use noise shaping.  What that translates into is that we *took perfectly good 24 bit music file, added 24 bits of noise to it, and then applied a transformation to that noise.  * Why do we want to do that?
> *[4a] Why not just give the original 24 bit file to people who want it?*
> ...



1. NO, no music is recorded at 24bits, it's recorded in a 24bit container but no ADC achieves more than about 20bits performance.
2. No, we've only got about 20bit performance to start with, then we need headroom! And on top of that we've got the recording noise floor; environmental noise (room, musicians, etc.) and equipment noise, SNR of the mics + mic-preamps. So in practice, "almost all music is recorded"at no more than about 18bits and typically fewer!
4. No, *that is NOT true!* We do not apply noise shaping to the original 24bit container tracks, we ONLY apply noise shaping as the final step in mastering and the music at that stage is typically at 64bit float.
4a. I asked you before and you ignored the question! You want the original recorded channels or do you want the music actually mixed and mastered? Not that it really matters because we are NEVER going to give you the originally recorded channels!!
5. No it's not. Noise shaped dither is applied as the final mastering step, therefore the only person who applies it is the mastering engineer and NOT the "entire content production industry"!
6. I disagree that applying noise shaped dither is the same as a lossy codec but even taking your weird definition of "lossy", there is NO option as even 24bit is effectively "lossy" because the music is created in 64bit float! Whatever container we distribute the music in it will always be "lossy", according to your definition!
7. Again, you cannot have the "originally captured" rates/channels, commercial artists will only sell you a finished product. So, continuing to scream and shout for something you cannot have comes across as nothing more than the ravings of one of those extreme audiophile nutters!

You can have a 24bit file which has been truncated twice or you can have a noise-shaped 16bit file. The reason for this thread is; what are you gaining with the double truncated 24bit file which makes it audibly superior to the noise-shaped 16bit file and worth the higher price of high resolution (which is not in effect any higher resolution)? The problem, as has been explained to you, is that going down the road of higher price for something only marketed as hi-res is the FREEDOM to get ripped-off by companies offering ever higher resolution for content and equipment which is NOT higher resolution. That's why we're starting to see 768/32!



amirm said:


> Surely we agree that if live performances resembled occupations noise in constructions or factory floor, people would not go pay and sit there to listen to them for one hour or more.



Of course we cannot agree that! Even governments don't agree with that, as many have now passed laws restricting the SPL levels of music performances on the grounds of health & safety and, I doubt there are many here who have not at some time in their life paid to go to a club/gig and not suffered hearing symptoms (tinnitus or threshold shift for example). Why are you ignoring the supplied scientific data of hearing damage caused by MUSIC?



amirm said:


> 33 db SPL what? I have explained so many times that when it comes to minimum noise level, you must examine the spectrum.



*Then WHY DON'T YOU?* Why are you steadfastly refusing to do EXACTLY what you are accusing others of not doing? Why???? YOU quoted the highest peak of a live gig was 129dBSPL, so, *what was the noise floor and it's spectrum at that live gig?* Are you saying that the noise floor of that gig in the critical band was less than 9dB SPL, which it would HAVE TO BE in order for the 120dB of dynamic range with 16bit to be insufficient. If you are saying that, where's your evidence?

All the above (and more) has been explained to you before, you just ignore it and just keep repeating the same INCOMPLETE and/or INAPPLICABLE data/studies. For this reason, you have now firmly entered the realm of trolling!!

G


----------



## reginalb

gregorio said:


> ...YOU quoted the highest peak of a live gig was 129dBSPL, so, *what was the noise floor and it's spectrum at that live gig?* Are you saying that the noise floor of that gig in the critical band was less than 9dB SPL, which it would HAVE TO BE in order for the 120dB of dynamic range with 16bit to be insufficient. If you are saying that, where's your evidence?
> 
> ...G



You left out an extremely important part of Amir's conditions, which is that noise shaping isn't a thing, so you can only say 96dB of dynamic range, not 120. So we're looking at a 33dB noise floor.


----------



## RRod

reginalb said:


> You left out an extremely important part of Amir's conditions, which is that noise shaping isn't a thing, so you can only say 96dB of dynamic range, not 120. So we're looking at a 33dB noise floor.



I think the argument is that you can't guarantee that the engineers will use shaping in the end, though I would assume that these days all of this is pretty automatic. I'll have to search for some non-shaped material with a low RMS.


----------



## reginalb

RRod said:


> I think the argument is that you can't guarantee that the engineers will use shaping in the end, though I would assume that these days all of this is pretty automatic. I'll have to search for some non-shaped material with a low RMS.



You're correct, of course - I should probably use less sarcasm. But I would say that engineers that fail to use noise shaping are probably likely to come up short in more ways than just that, resulting in a master that isn't going to be great on 24-bit either, unless they're intentionally making the 16-bit worse to serve a narrative.


----------



## Darren G

What amp or DAC do you have at home, or carry with you, that exceeds 20 bits of dynamic range; or call it 21 if you really want to push it before you hit the noise floor of your gear?  Even really good gear can only questionably claim this level of performance.

4-5 bits is potentially significant, but who listens at 100+ db?  If you listen at an reasonable level, then those 4 bits get pushed down into the noise, and at that point we are talking 1/1000, 1,10,000, or lets say 1/100,000 thousands level of noise.  The sound of a gnat farting in the wind level.   Who is really hearing this?


----------



## castleofargh

Darren G said:


> What amp or DAC do you have at home, or carry with you, that exceeds 20 bits of dynamic range; or call it 21 if you really want to push it before you hit the noise floor of your gear?  Even really good gear can only questionably claim this level of performance.
> 
> 4-5 bits is potentially significant, but who listens at 100+ db?  If you listen at an reasonable level, then those 4 bits get pushed down into the noise, and at that point we are talking 1/1000, 1,10,000, or lets say 1/100,000 thousands level of noise.  The sound of a gnat farting in the wind level.   Who is really hearing this?


Amirm will reject that idea with his argument that we can still get audible cues below noise under the right conditions. we can't really make a point so long as he can argue with extreme circumstances. it's like a guy digging a bomb shelter in his garden. we all know it's too much money and efforts for something "just in case" that may or may not prolong his life a little if ever the day comes. but at the same time we can't claim that a bomb will never fall on our head. 
he's not making stuff up, he's only picking all the worst possible situations, many of which will never happen on the album or with our listening habits. but they could!

I believe we can still try to be realistic and practical. if the argument is that 16bit forces the master to cut or compress realistic dynamic ranges within an album, how many 24bit albums show more than 90dB of dynamic within the signal? if any exists, and someone happens not to find them horrible and annoying, let him get those in 24bit for when he wishes to listen very loud and have a slightly more hifi experience.
and for those who want to listen to music as if it was a live show all year long, congrats, you're likely to achieve the same hearing damages as the real professionals on tour. if that's not the experience like you're with the artist, I don't know what is. you guys go ahead and get everything in 24bit to enjoy all the greatness of the silent passages(if they exist on the album). and I hope it's worth it. 

and of course those who just want 24bit albums because it's more, go get them and have fun. no matter if it doesn't sound different or if you only listen at reasonable levels. not everybody has to prove that he needs something to have it if he wants. at some point it was supposed to be a fun hobby. ^_^


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## bigshot

The problem with some audiophiles is that they focus so much on the stuff from the bleeding edge of our ability to hear well into the range of the inaudible, they end up soft peddling the sound we clearly can hear. If they spent as much time refining and perfecting the core as they do fussing with the extremes, they would have perfect sound and wouldn't need to fret so much any more. But I guess sound fidelity really isn't their focus. They're more interested in the theories of the thing than the actual sound of it.


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## RRod

For some measured thoughts on the subject, here's what a former AES president and fellow, Grammy and Technical Grammy winner, and decades-long consultant with JBL thought of the state of digital audio back in… 1992:


> I'm very happy to say that most of the problems of digital have been dispatched—or essentially dispatched—in recent times. There are so many improvements in converters being made today that I really haven't got a problem with it. In a purely intellectual sense, I think we all wish that we had a higher sampling frequency, just for the sake of maybe that very, very tiny percentage of the population who can still hear a difference at 20 kHz. I don't really object to the 16-bit word length because we now have a very benign way of handling the bottom end of the scale.  That's by dithering the signal and getting almost analog-like performance at the low end [of the dynamic range], where the signal can be heard to fade into a noise floor without any abrupt things going on, and no more of this nonsense that we heard earlier of the signal disappearing. You know, the old stuff about the reverberation disappearing when it got down to the least significant bit. I doubt that ever really happened anyhow because there's always been dither in the form of amplifier/preamp noise or even room [noise]: That's not ideal dither, but it can be effective enough to keep some of these things from happening.  In any event, I feel that the quality of conversion today—the low-bit conversion techniques, noise-shaping methods—really do create a superior medium, one that I have absolutely no problems with and one which I'm confident—despite what some people say—can be cloned ad infinitum. As long as all the errors are detected and corrected, there's no reason cloning can't go on forever.


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## gregorio

castleofargh said:


> Amirm will reject that idea with his argument that we can still get audible cues below noise under the right conditions. we can't really make a point so long as he can argue with extreme circumstances.



I believe we can make such an argument/point. We can easily make this point with reference to real life, practical application but the difficulty is making the point with science alone, not because science disproves the point but because there isn't, as far as I'm aware, published science which covers all of the required factors.

To hear beyond the limits of 16bit requires ALL the following to occur simultaneously:
1. A recording with a noise floor lower than -120dBFS. AND, 2. A DAC, an amp, headphones/speakers and a listening environment, the noise of which COMBINED does not exceed 0dBSPL and which are ALL also capable of >120dSPL peaks. AND finally, 3. The desire to listen to your music at >120dBSPL peaks AND the ability to actually hear a 0dBSPL noise floor after such peak levels.

1. In theory it wouldn't be impossible to achieve this with a combination of a low noise floor recording environment, an ADC with greater than 120dB dynamic range, the use of mics with a very large dynamic range and very close mic'ing techniques. In practice this does NOT occur though, we require headroom when recording because we do not know in advance what the peaks are going to be, typically this would be anywhere from 6dB to about 18dB. When we normalise the recording, bring the peaks close to 0dBFS, the noise floor of the recording will also obviously increase by around 6-18dB, so even with the very quietest ADC's available, this would still bring our peak to noise floor ratio to within or well within 120dB. Additionally, if we do use mics with a very high dynamic range and close mic'ing then we also always apply significant compression. In real life, as @amirm data does not disprove, the highest peak performance levels occur where we also have the highest noise levels, rock/pop or club gigs for example, resulting in peak to noise level of probably no more than 60dB or so. Where we find the highest peak levels vs noise floor is almost certainly at a symphony concerts, where the audience is actively trying to be as quiet as possible but, we do not use close mic'ing techniques when recording an orchestra, as amirm's quoted photo also demonstrates. We choose mics based on artistic requirements, NOT on dynamic range performance and even that quiet audience and/or quiet musicians are not achieving a 0dBSPL noise floor. We cannot however prove any of this with published science because, AFAIK, science has not published anything in this regard; how do you measure the noise floor of a concert venue with musicians and an audience WHILE the musicians are playing?  The mistake being made is to argue purely on what science has published, completely ignore the real life factors science hasn't studied/published and to call the resultant conclusions "real life". It's akin to arguing that the published data on car fuel economy, before the time of more applicable/appropriate methods of measuring fuel usage, was accurate "real life" figures and those arguing against them were ignoring the science. Today, we have other published "scientific" fuel economy figures which are more appropriate because they account for real life factors which the previous tests did not, real roads, air resistance and real driving conditions for example. 
The only potential to encounter such a dynamic range on a recording is after the musicians have stopped playing, for example, during a fade out applied by the engineers at the end of some pieces. 

2. Very rare but possible.

3. Exceedingly rare to have both and even if there are such people, it's certainly inadvisable as the published science demonstrates that MUSIC peaks above 120dBSPL are potentially dangerous/damaging and that's with a significantly higher than 0dBSPL noise floor! 



castleofargh said:


> if the argument is that 16bit forces the master to cut or compress realistic dynamic ranges within an album, how many 24bit albums show more than 90dB of dynamic within the signal?



That argument holds no water, dynamic ranges are routinely compressed but due to other limiting factors, not the dynamic range limitations of 16bit. The argument has been made that on those almost non-existent occasions where close mic'ing has been employed with very wide dynamic range mics, not to normalise or apply compression. The problem with this argument is that it effectively means not actually mixing or mastering the originally recorded channels or specifically making a recording with huge dynamic range rather than artistry as the goal and for a very tiny number of potential consumers who fulfil all the above requirements and are willing to potentially damage their hearing. There is no precedent for such commercial content, certainly no precedent for such content to make any money and if anyone did make such content, they could potentially open themselves up to law suits!



RRod said:


> I think the argument is that you can't guarantee that the engineers will use shaping in the end, though I would assume that these days all of this is pretty automatic.



It is the recommended practise but it is not always applied. The reason why it's not always applied had been deliberately avoided by amirm, despite being asked a number of times. I know some mastering engineers who do not apply any sort of dither, on the basis that on a particular track/group of tracks they cannot hear any truncation error and that if they can't hear any in their mastering studio with their experienced hearing then it's not going to affect consumers, with their lesser environments and hearing. It is expected that those occasions when the recording is faded out, that it fades out into the consumer's noise floor, not into the dither, noise-shaped dither or truncation noise of the recording. Obviously, we are not accounting for the likes of amirm, people with years of specific training to detect errors/noise, specific equipment to isolate environmental noise and using that equipment in a non-recommended (and potentially damaging) way to make the noise on recordings far more noticeable. It's typically not possible or desirable to account for such an extremity, economics dictate a limited amount of time to create a recording and accounting for such an extremity would grind each of the recording, mixing and mastering phases virtually to a standstill! If we _were_ to account for such people, we could still do so with 16bit, if we chose, which allows us to have a dynamic range of up to 150dB in the critical band. The fact that we rarely choose such a potential dynamic range is a choice, not a limitation of 16bit.

Amirm has chosen to interpret my OP as a rant against higher than 16bit. That is an INCORRECT interpretation! I've been professionally recording EXCLUSIVELY in greater than 16bit since 1992. My OP was not designed to argue against 24bit, it was designed to expose the myth that 24bit as a consumer distribution format provides higher resolution than 16bit. It doesn't, it provides exactly same resolution but with a theoretically lower noise floor which in practise (real life) is either not employable or discernible ONLY if one were to playback recordings significantly differently to how the artists intended AND/or at potentially dangerous or very dangerous levels!!

G


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## vatch

Awesome stuff Gregorio.  Easy to understand and exceedingly well done.


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## 1800yolk

my main question from the OP: what does all that extra data end up being then? This is the mind boggling part, everything else makes sense to me.


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## bigshot

Recordings of sound you can't hear with human ears. Also sound that doesn't even exist in music.


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## 1800yolk

bigshot said:


> Recordings of sound you can't hear with human ears. Also sound that doesn't even exist in music.


Thanks! This whole topic got me thinking, though... Say you're using a tube amp. Any and all audio that passes through it will generate "even order harmonics" (I just learned about this stuff lol), which become more prominent the more you crank your amp towards its limits. Those inaudible sounds will still be inaudible, but when all those sounds hit the air, they're going to interact with each other and will very, very subtly change the harmonics of the audible range, and arguably for the better. On the other hand, the amp is going to be doing that with 16/44 stuff, just not quite as much. Obviously factors such as mastering and mixing matter way more, but I think that it isn't completely unreasonable to acquire high fidelity music for this reason. I'm curious about your thoughts though!


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## 71 dB

1800yolk said:


> Thanks! This whole topic got me thinking, though... Say you're using a tube amp. Any and all audio that passes through it will generate "even order harmonics" (I just learned about this stuff lol), which become more prominent the more you crank your amp towards its limits. Those inaudible sounds will still be inaudible, but when all those sounds hit the air, they're going to interact with each other and will very, very subtly change the harmonics of the audible range, and arguably for the better. On the other hand, the amp is going to be doing that with 16/44 stuff, just not quite as much. Obviously factors such as mastering and mixing matter way more, but I think that it isn't completely unreasonable to acquire high fidelity music for this reason. I'm curious about your thoughts though!



Odd harmonics are generated when you have symmetric non-linearities, in other words positive and negative sides of the signal are treated similarly.
Even harmonics are generated when you have asymmetric non-linearities, in other words positive and negative sides of the signal are NOT treated similarly.

Most non-linear devices are a combination of these two types of distortion. Depending on how asymmetric they are they generate more or less odd/even harmonics.

Tube amps generate also odd harmonics, but even harmonics are dominant. In order to inaudible sounds interact with each other (generate differential frequencies) in the audible frequency range you need non-linearities in the air. Well, acoustic waves are VERY linear at reasonable listening levels and non-linearities start to occur at level such as 160 dB!
Distortion is distortion whether you like it or not. Hi-fi is about avoiding distortion. Put the _pleasing_ distortions in the source material! There are tons of tube amp simulator plugins available. Use them if it makes the sound better! Sell us the final product, not something we need to finalize ourself with historical amp technology.


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## gregorio

1800yolk said:


> my main question from the OP: what does all that extra data end up being then? This is the mind boggling part, everything else makes sense to me.



Noise! 

If we record at 24bit using a high-end pro ADC (Analogue to Digital Converter), at least the last 3 or 4 bits are just going to be thermal noise generated by the components (resistors, etc.) in the ADC itself. At absolute best (best ADC with nothing plugged into it), we can't have more than about 20-21 bits of material. Of course, having nothing plugged into our ADC and therefore recording nothing other than thermal noise in bits 20-24 isn't going to be a great music product! We're going to have to plug something into our ADC, mics for example. Now though of course, we've got the thermal noise of our ADC + the thermal noise of our mic pre-amps + the thermal noise of our mics + the noise floor of the recording studio and both of these last two items are being amplified several (or many) times by our mic pre-amps! Our 20-21 bit starting point is now considerably lower, typically somewhere around 13bits or so, the rest of the bits just contain noise from all the various sources mentioned (plus usually some other sources as well, such as guitar amps/cabs for example).



1800yolk said:


> On the other hand, the amp is going to be doing that with 16/44 stuff, just not quite as much.



I'm not sure I understand, why would the amp not be doing that quite as much with 16/44 stuff? 0dBFS is the loudest a digital file can be and 0dBFS is exactly the same with 16bit as it is with 24 or any other bit depth. What changes with increased bit depth is the quietest signals we can theoretically record.

G


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## SilverEars (Apr 5, 2018)

Does anybody have an explaination for this?

On my DAP(portable digital audio player) which is capable of 32bit using XMOS chip inside.

I tried the XMOS mode and automatic(which chooses the recording's original bit depth and sampling rate I believe), and automatic just sounds more detailed and XMOS 32bit mode sounds smoothed out in details.  Why would that be?  Is this result of over-sampling?


I also notice this with Tidal as well.  When I set the streaming mode for which I believe to be WASAPI, the DAC chooses the original bit depth and sampling rate, and it sounds better and more detailed.

Is this the difference between oversampling vs the original sampling rate?  Does over-sampling reduce perception of resolution?  Does this have to do with interpolation?


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## bigshot

I'd suspect that something more than just upsampling is involved there.


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## gregorio

SilverEars said:


> Does over-sampling reduce perception of resolution? Does this have to do with interpolation?



No and no. Changing bit depth from a lower bit depth to a higher bit depth should result in absolutely identical data/sound and over-sampling should be completely audibly transparent. I'm with bigshot on this one, something else must be going on when you choose that mode.

G


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## 1800yolk

gregorio said:


> Noise!
> I'm not sure I understand, why would the amp not be doing that quite as much with 16/44 stuff? 0dBFS is the loudest a digital file can be and 0dBFS is exactly the same with 16bit as it is with 24 or any other bit depth. What changes with increased bit depth is the quietest signals we can theoretically record.
> G


Thank you for your reply! The 16/44 vs 24/44 would make no difference with the amp in regards to generating any harmonics that would influence the sound. I think there is a difference in 16/44 vs 24/88 though.


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## bigshot

If you're taking 16/44.1 and bouncing it up to 24/96 it shouldn't change the sound at all. Everything is identical within the range covered by 16/44.1. The rest is all zeros so to speak.


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## 1800yolk

bigshot said:


> If you're taking 16/44.1 and bouncing it up to 24/96 it shouldn't change the sound at all. Everything is identical within the range covered by 16/44.1. The rest is all zeros so to speak.


I'm discussing digital releases that are 24/88 to begin with, obviously putting 16/44 in a 24/88 container would do nothing


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## bigshot

Well then it's recorded sound that you can't hear because it's frequencies outside the range of human hearing, or it's sound so quiet, you would have to turn the loud stuff up to deafening levels to hear it. Check out the link "CD Sound Is All You Need" in my sig.


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## RRod

1800yolk said:


> Thank you for your reply! The 16/44 vs 24/44 would make no difference with the amp in regards to generating any harmonics that would influence the sound. I think there is a difference in 16/44 vs 24/88 though.



The harmonic distortion for any content > 11kHz will be inaudible, so you're asking in essence about audibility of IMD. This would require that the high-res frequencies contributing to the IMD be loud enough so that the IMD product is audible *over the other content potentially masking it in the audible range*, unless you have some kind of unbounded distortion occurring. If you look up the people trying their damndest to prove audibility of hi-res, you'll find examples such as playing back recordings of gamelan music really loudly, because only certain content has a snowball's chance in hell of any hi-res induced IMD being audible, and that's assuming the distortion heard is due only to the non-linearities of the ear and not the speakers.


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## vatch

This thread is quite old and any and all questions have been answered suitably.  Before asking questions I suggest reading it through.  *There's really no purpose for all this extra data; it simply is very high frequency noise, something we almost always try to reduce in audio.*

I highly doubt a tube amp capable of producing much greater than 20hz-20khz frequency range anyway which is the same for most speakers, ss amps, etc.  There is EMI and RFI noise on the power lines we always try to reduce.  Why add to the noise?

It's not nice to blow your dog/cat/guinea pig/parrot's ears out!


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## gregorio

1800yolk said:


> The 16/44 vs 24/44 would make no difference with the amp in regards to generating any harmonics that would influence the sound. I think there is a difference in 16/44 vs 24/88 though.



That is possible, in terms of the sample rate rather than the bit depth though. Expanding on what @RRod stated, we'd effectively be talking about IMD. If we feed a signal to an amp beyond what the amp is designed to accept, say a 40kHz signal input into an amp designed for signals in the range of 20Hz to 20kHz, an amp can respond with distortion in the audible range. I read an article or paper from a reliable source some years ago, which tested a number of speakers and amps and concluded that this type of audible distortion occurs much more commonly than is generally assumed, even with high sample rate material which only contains relatively small amounts of ultrasonic material. Unfortunately, I can't remember the title or enough other details of the article to locate it with a cursory search. Does anyone out there know the paper/article in question?

G


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## 1800yolk

gregorio said:


> That is possible, in terms of the sample rate rather than the bit depth though. Expanding on what @RRod stated, we'd effectively be talking about IMD. If we feed a signal to an amp beyond what the amp is designed to accept, say a 40kHz signal input into an amp designed for signals in the range of 20Hz to 20kHz, an amp can respond with distortion in the audible range. I read an article or paper from a reliable source some years ago, which tested a number of speakers and amps and concluded that this type of audible distortion occurs much more commonly than is generally assumed, even with high sample rate material which only contains relatively small amounts of ultrasonic material. Unfortunately, I can't remember the title or enough other details of the article to locate it with a cursory search. Does anyone out there know the paper/article in question?
> 
> G


This is great! I appreciate everyone's answers, b/c there are lots of points to consider. I found an interesting article about IMD that explains why the even order harmonics I was hoping to achieve aren't as natural and desirable as they're made out to be:
http://sound.whsites.net/valves/thd-imd.html
So in the end, pumping 24/88 music through a tube amp, in addition to other sources of distortion mentioned above, will create distortion that may or may not be better than settling for clean, predictable 16/44 music. It's hard to say what music benefits from this in any capacity, but my list would include albums that were poorly mastered. It could also make them worse! It's an interesting gamble.


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## gregorio

1800yolk said:


> So in the end, pumping 24/88 music through a tube amp, in addition to other sources of distortion mentioned above, will create distortion that may or may not be better than settling for clean, predictable 16/44 music.



Pumping high frequency material through an amp MAY (not necessarily "will") create IMD. If our goal is high-fidelity reproduction, then in no sense could the introduction of IMD during reproduction be considered "better". Furthermore, as introduced IMD is typically unrelated to the content, there would be few occasions where it would be "better" even purely in terms of personal subjective preference. Often with tubes, the goal is the addition of even harmonics, which although by definition is deliberately "lower fidelity", can nevertheless be euphonic (pleasing). Even the addition of odd harmonics can be desirable under various conditions but tones which are not harmonics, not related to the fundamental frequencies at all, are much more rarely desirable. IMO, the use of IMD should be restricted to specific places and specific amounts, and that means restricting it's application to the mix/master itself and not randomly applying random amounts during playback!

G


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## 71 dB

gregorio said:


> ...and over-sampling should be completely audibly transparent.
> 
> G


_Audibly_ transparent perhaps, but:

Oversampling 44100 Hz to 88200 Hz adding zeros between every sample point and low-pass filtering the result is very transparent.
Oversampling 44100 Hz to 96000 Hz with linear interpolation is theoretically not very transparent (interpolation distortion).
Oversampling 44100 Hz to 96000 Hz with zinc-interpolation is very transparent.



gregorio said:


> If we feed a signal to an amp beyond what the amp is designed to accept, say a 40kHz signal input into an amp designed for signals in the range of 20Hz to 20kHz, an amp can respond with distortion in the audible range. I read an article or paper from a reliable source some years ago, which tested a number of speakers and amps and concluded that this type of audible distortion occurs much more commonly than is generally assumed, even with high sample rate material which only contains relatively small amounts of ultrasonic material.
> 
> G



Was it a paper by Lavry?

The benefit of "low" sample rates 44.1 kHz and 48 kHz is that after proper reconstruction filtering hardly anything above 20 kHz is fed to amps.


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## gregorio

71 dB said:


> [1] Oversampling 44100 Hz to 96000 Hz with linear interpolation is theoretically not very transparent (interpolation distortion).
> [2] Was it a paper by Lavry?



1. True but I don't believe that still occurs much (or at all) these days does it? The ADCs/DACs I was using more than 25 years ago were oversampling 128 times.

2. I don't think so. While Lavry has certainly mentioned IMD and IMD avoidance with the standard sample rates (44 and 48) in at least one of his papers, the paper/article I'm referring to actually tested some amps and speakers for the occurrence of IMD and concluded it's occurrence was far more prevalent than generally assumed. IE. It's generally assumed to be fairly rare, except when using signals designed to elicit IMD but it may in fact be quite/very common, even under normal usage consumer usage conditions, when reproducing typical commercial music recordings.

G


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## skwoodwiva

While not science at all!
I believe this fundamental is why fone, the Italian, label is the finest producer of CDs Then they skipped so called better pcm formats and went exclusively DSD.


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## RRod

skwoodwiva said:


> While not science at all!
> I believe this fundamental is why fone, the Italian, label is the finest producer of CDs Then they skipped so called better pcm formats and went exclusively DSD.



And their microphones are from 1947!


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## skwoodwiva

RRod said:


> And their microphones are from 1947!


You know this to be true 
Or joking I cannot tell.


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## RRod

skwoodwiva said:


> You know this to be true
> Or joking I cannot tell.



It says right at the top of their site. Seems like a company specifically tailoring to the type of people who think all modern sound recording is evil. I feel quite the opposite.


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## skwoodwiva

RRod said:


> It says right at the top of their site. Seems like a company specifically tailoring to the type of people who think all modern sound recording is evil. I feel quite the opposite.


Golden info, man thank you.

 Now WHY The @ell would I lust after this label? I have 40 of them even some of the CDs are better in musicality & sonics the anything in my 2 TBs....


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## 71 dB

gregorio said:


> True but I don't believe that still occurs much (or at all) these days does it? The ADCs/DACs I was using more than 25 years ago were oversampling 128 times.
> 
> G


That's fair. One-bit sigma-delta modulators and all. I was talking about other kind of processing. In early 2000's I had to write a Matlab sinc-interpolation script to resample impulse responses measured with MLSSA (which had weird sample rate) to 44.1 kHz.


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## gregorio

71 dB said:


> That's fair. One-bit sigma-delta modulators and all.



Even by around the mid/late 1990's, the vast majority of pro ADC/DACs were initially sampling at rates of 11MHz or higher using 5 or so bits, and then decimating down to the required bit depth/sampling rates. I'm not so well versed in the history of consumer DACs.

G


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## 71 dB

gregorio said:


> Even by around the mid/late 1990's, the vast majority of pro ADC/DACs were initially sampling at rates of 11MHz or higher using 5 or so bits, and then decimating down to the required bit depth/sampling rates. I'm not so well versed in the history of consumer DACs.
> 
> G


Sure, but if you have a MLSSA impulse response .TIM -file and you need to convolute some test signals at 44.100 Hz with it to be burned on a CD-R for playback, you need to resample the .TIM file to 44.100 Hz. Why not write a Matlab script that does sinc -interpolation numerically at astronomical precision and do also the convolution stuff while we are at it?


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## TheSonicTruth

RRod said:


> It says right at the top of their site. Seems like a company specifically tailoring to the type of people who think all modern sound recording is evil. I feel quite the opposite.



By "modern sound" do you mean compress, squash, brick-wall limit and makeup gain, rinse and repeat?  If so, _that's _evil!  Production values matter - far more than vintage of mics used, or final delivery format.  And the stuff we're going around in circles about on this thread, this 16/44 vs 24/88 and so on, even DOGS have a hard time hearing!  What difference does it make? Grab your favorite drink, kick back and just enjoy the music!


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## bigshot

71 dB said:


> _Audibly_ transparent perhaps, but:



Audibly transparent is a threshold. There's no such thing as being more transparent than transparent. Our ears are a limiting factor. We can create sound that includes all kinds of extra information, but the ears are going to ignore all the stuff they can't hear.


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## amirm

bigshot said:


> Audibly transparent is a threshold. There's no such thing as being more transparent than transparent.


Of course there is if you understand the practice as opposed to lay assumptions.  You need headroom for processing.  And threshold for distortion varies between people.  In psychoacoustic studies, a range of data is gathered and shown as averages, not absolutes.    You can see this even in the famous fletcher-munson graphs:






See the difference between red and blue graphs above even when averages are considered.

So to assure transparency, you need to go beyond what a study says.  Perception is not a precise science.


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## bigshot

None of that matters if it isn’t audible. Inaudible is inaudible. Transparent is transparent. There is such a thing as “good enough for government work”.


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## HotIce

We are at it again, applying terms which are normally used in other perception fields, to hearing 
From a visual point of view, transparency of a medium is its capacity to leave the image unchanged, when interposed between a scenery and the observer.
Applying this to hearing/audio, seems the natural fit would be to replace "image" with "frequency response".
I am pretty sure though, that the location within the FR where the changes happen, have large impact to the perception of what people (I know I am one of them) usually imagine associated with "transparency".
More specifically, an FR drop in the lower range would be much less associated with lack of transparency, compared as a similarly audible drop in the higher range.


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## amirm

bigshot said:


> None of that matters if it isn’t audible. Inaudible is inaudible. Transparent is transparent. There is such a thing as “good enough for government work”.


Doesn't look like you understood my answer, nor the mistake in yours. 

Let me give you an analogy.  If you are designing a bridge and don't want it to collapse, do you just weigh 4 of them and assume all cars passing will weigh that much and build the bridge to that spec?  I sure hope not!  You have to have good amount of safety margin because your data is not representative of full set of car weights the bridge has to hold.

Same is here with respect to perception tests.  Here is a quote from bible of psychoacoustics,










Bottom line, you don't know with absolute precision what the threshold is.  Only by leaving a good margin beyond what psychoacoustics has determined you can be confident you are below threshold of audibility.


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## bigshot (Apr 28, 2018)

I believe that human perception has limits. Those limits might vary from person to person. Some may be able to hear to the absolute edge of human perception, other people may have degraded hearing that limits them to a range below that. Variability extends *downward*, not upward. There comes a point where no human on earth can hear it. That is what we call the point of transparency. If you want to claim that the established thresholds are incorrect, start gathering evidence to prove it and publish it and if you're correct and your proofs are repeatable by others, the thresholds will be modified. If you think you personally can hear things you shouldn't be able to hear, present yourself to the audiology department of your local university and suggest that they test you.

It isn't logical or practical to pile one "fudge factor" on top of another "a little bit to grow on". That is what audiophools do to convince themselves that they need one more zero on the right of the decimal point. That leads to wasting time, energy and money on sound you can't hear. There is nothing better than audible transparency. If you suspect you're missing something, do a blind comparison and see if you can hear it. I bet you can't hear beyond the established thresholds. If you absolutely must build in buffer, then just follow the thresholds using tones. Those are significantly broader than the thresholds when listening to music. But for the purposes of listening to music, that is overkill.


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## amirm

bigshot said:


> There comes a point where no human on earth can hear it. That is what we call the point of transparency.


Let me know when you have tested all humans to determine that....Typical psychoacoustic study uses a handful of listeners.


----------



## amirm

bigshot said:


> If you think you personally can hear things you shouldn't be able to hear, present yourself to the audiology department of your local university and suggest that they test you.


Spoken as someone who has not submitted yourself to audiology tests.  If you had, you would know that they are not there to test the limits of your hearing.  For example, the highest frequency they test is 8 kHz.

You seem to have no use for audio research or actual experience with the topic.


----------



## bigshot

No, I didn't mean for a checkup. I meant for research to see if we need to extend the thresholds.


----------



## bigshot (Apr 28, 2018)

amirm said:


> Let me know when you have tested all humans to determine that.



I'll refrain from saying that the Sun will rise in the East tomorrow until I have a chance to gather my evidence. I'll have it for you tomorrow by noon.


----------



## amirm

bigshot said:


> No, I didn't mean for a checkup. I meant for research to see if we need to extend the thresholds.


Go ahead and call them to ask if they can measure your threshold for FM modulation (jitter/Wow and flutter), they will refer you to the psychology department!  

That is not what they do.  They worry about medical consequences.  If there is none, it is not of interest to them.  To wit, for diagnosing some cancers frequency response up to 20 kHz is measured in audiology tests.  

So no, that is not what you do.  First thing you should do is study the research that is already out there.  Understand what is truly tested.  Learn about statistics and their meaning as applied to listening tests.  To the extend you are unwilling to do any of that, then please don't waste time lecturing the other camp to learn audio science.  Learning needs to start with our team, not theirs.


----------



## amirm

bigshot said:


> I'll refrain from saying that the Sun will rise in the East tomorrow until I have a chance to gather my evidence. I'll have it for you tomorrow by noon.


Good luck with that in winter in Alaska....


----------



## bigshot




----------



## colonelkernel8

amirm said:


> Good luck with that in winter in Alaska....


I know you’re the master of pedantry, but I’ll do you one step further, polar nights only occur at roughly 66.5 degrees north latitude and above in Alaska, leaving most of the population in Anchorage without that phenomenon.


----------



## bigshot

And if you're on the moon, the sun doesn't rise at all!


----------



## Glmoneydawg

bigshot said:


> And if you're on the moon, the sun doesn't rise at all!


It rises....its just not that impressive....was gonna post a pic,but they all suck...new appreciation for the view from here.


----------



## bigshot

How do you howl at the moon when you are on it, dawg?


----------



## colonelkernel8

bigshot said:


> How do you howl at the moon when you are on it, dawg?


He looks down at his feet I suppose.


----------



## chef8489 (Apr 29, 2018)

amirm said:


> Spoken as someone who has not submitted yourself to audiology tests.  If you had, you would know that they are not there to test the limits of your hearing.  For example, the highest frequency they test is 8 kHz.
> 
> You seem to have no use for audio research or actual experience with the topic.


You must have crappy doctors if yours only test up to 8khz. Our government tests are beyond 20khz as they want to find out if we have hearing loss. My last test took over 4 hours.


----------



## castleofargh

the most basic hearing test at my previous work done every 3 years was up to 8khz too. it's the stuff I've seen all my life with the crappy closed back on ear headphone which has clamp of death to try and isolate a little from external noises. usually with rather poor success in my experience. I've had cars passing down the street probably 20dB louder than the test tone I was supposed to identify. not that it's relevant anyway, they do it because they have a legal obligation to do so, but if you end up with massive hearing loss, they stop being doctors and instantly become defense lawyers. "do you own a lawn mower? do you go to concerts? do you have a car? are your kids noisy? well that's why you have hearing loss, and absolutely not because you spend 8hours a day next to a V16 engine without the right to ear plugs. now that we've established that your job is very safe for your ears, have a good day and be more careful when around kids your ears are precious."  or something along those lines...


----------



## chef8489

castleofargh said:


> the most basic hearing test at my previous work done every 3 years was up to 8khz too. it's the stuff I've seen all my life with the crappy closed back on ear headphone which has clamp of death to try and isolate a little from external noises. usually with rather poor success in my experience. I've had cars passing down the street probably 20dB louder than the test tone I was supposed to identify. not that it's relevant anyway, they do it because they have a legal obligation to do so, but if you end up with massive hearing loss, they stop being doctors and instantly become defense lawyers. "do you own a lawn mower? do you go to concerts? do you have a car? are your kids noisy? well that's why you have hearing loss, and absolutely not because you spend 8hours a day next to a V16 engine without the right to ear plugs. now that we've established that your job is very safe for your ears, have a good day and be more careful when around kids your ears are precious."  or something along those lines...


Ours are done with in ears then they switch to bone conducted tests half way through. It is extremely intensive and long. Have to take several breaks during it.


----------



## amirm

chef8489 said:


> You must have crappy doctors if yours only test up to 8khz. Our government tests are beyond 20khz as they want to find out if we have hearing loss. My last test took over 4 hours.


Which government is that?  Is there a link to the test protocol?  Do you have your results you can post?

The tests here last 15 to 20 minutes.  I doubt very much anyone would want to sit through a 4 hour test without any indication of disease.  

Fatigue is a big problem when tests lasts more than 20 to 30 minutes.  So I question whether there are valid results in anything that would last 4 hours.

As to doctors being crappy, that is not the case.  Their focus is in people hearing and understanding others.  20 kHz is not needed for that.


----------



## bigshot

I'm sure if you called an audiologist and told him the kind of test you would like to have done, he could accommodate it.


----------



## amirm

bigshot said:


> I'm sure if you called an audiologist and told him the kind of test you would like to have done, he could accommodate it.


Nope.  I asked.  They said you had to go the hospital to have the full frequency range test done.  And that they would be reluctant to run it on me since it is reserved for cancer patients and such.

The entire test setup, equipment and training is different.

I trust you have not had the test done to say that?


----------



## chef8489

amirm said:


> Which government is that?  Is there a link to the test protocol?  Do you have your results you can post?
> 
> The tests here last 15 to 20 minutes.  I doubt very much anyone would want to sit through a 4 hour test without any indication of disease.
> 
> ...


That would be the Us government. I would have to request those results as we do not get copies. They are kept in our medical records and they are government property. Part of the reason they are so indepth is for compensation and pension claims for service connection disabilities later on.


----------



## TheSonicTruth

chef8489 said:


> That would be the Us government. I would have to request those results as we do not get copies. They are kept in our medical records and they are government property. Part of the reason they are so indepth is for compensation and pension claims for service connection disabilities later on.



"_That would be the Us government. I would 
have to request those results as we do not 
get copies._"

"We do not get copies" - of your own hearing test results?

What is this, the United States of America or communist-era Russia?!


----------



## chef8489

TheSonicTruth said:


> "_That would be the Us government. I would
> have to request those results as we do not
> get copies._"
> 
> ...


No we do not get copies of our tests as they go in our records. They go over the tests, but the records do not belong to us. We can submit official requests for copies. If our records are restricted due to security clearances and service locations that makes things more complicated.


----------



## TheSonicTruth

chef8489 said:


> No we do not get copies of our tests as they go in our records. They go over the tests, but the records do not belong to us. We can submit official requests for copies. If our records are restricted due to security clearances and service locations that makes things more complicated.




That's RIDICULOUS. This is a hearing test, albeit a more thorough one than we civvies get, not classified intel or top-secret weapons design here!


----------



## chef8489

TheSonicTruth said:


> That's RIDICULOUS. This is a hearing test, albeit a more thorough one than we civvies get, not classified intel or top-secret weapons design here!


I get you. Like i said i can submit a request to get copies and see. It has been about a year or two since my last one. I need to check when it was.


----------



## TheSonicTruth

chef8489 said:


> I get you. Like i said i can submit a request to get copies and see. It has been about a year or two since my last one. I need to check when it was.



I'm not ridiculing or mad at you, I'm just questioning a system that would withold such mundane data from you.


----------



## colonelkernel8

TheSonicTruth said:


> That's RIDICULOUS. This is a hearing test, albeit a more thorough one than we civvies get, not classified intel or top-secret weapons design here!


You see, Chef is a weapon.


----------



## chef8489

colonelkernel8 said:


> You see, Chef is a weapon.


I was when i was serving in SF Now i am just an instructor and a chef.


----------



## TheSonicTruth

chef8489 said:


> I was when i was serving in SF Now i am just an instructor and a chef.



Like Seagal in "Under Siege"! lol


----------



## colonelkernel8

chef8489 said:


> I was when i was serving in SF Now i am just an instructor and a chef.


And an R&B singer perhaps?


----------



## castleofargh

chef8489 said:


> I was when i was serving in SF Now i am just an instructor and a chef.


I imagine it's only the billionth time someone makes that joke, but you've baited us hard with this "I'm just a cook" post. ^_^


----------



## chef8489

castleofargh said:


> I imagine it's only the billionth time someone makes that joke, but you've baited us hard with this "I'm just a cook" post. ^_^


My x made it all the time. No worries.


----------



## chef8489

colonelkernel8 said:


> And an R&B singer perhaps?


No i dont sing for crap.


----------



## colonelkernel8

chef8489 said:


> No i dont sing for crap.


I'm just joking with you guy. Like "Chef" from "South Park".


----------



## chef8489

colonelkernel8 said:


> I'm just joking with you guy. Like "Chef" from "South Park".


Ah lol did not catch that. Was too busy making chocolate salty balls.


----------



## TheSonicTruth

chef8489 said:


> No i dont sing for crap.



Hey I resemble that statement! I like R&B, among many genres.


----------



## roweder (Jul 6, 2018)

This article makes it sound to me that when using 24 bit audio instead of 16 bit, the range of loudness that the audio file can describe becomes greater.  Why couldn't it instead increase the number of digital increments describing the difference between say -4db and -3db?

additionally, why couldn't the dynamic range of 16 bit audio be expanded by reducing the number of digital increments per 1db change? 

Is the difference in db between the integer value that the 16 or 24 bits convert to a fixed value? 




To me it seems wasteful to allow a 16 bit audio file to have a noise floor ~60db below the quietest noise in the music (assuming a song with 36db of dynamic range).  Why can't the lowest noise or output voltage (-36db) be set to 0000 0000 0000 0001 and the highest noise (0db) to 1111 1111 1111 1111?


I apologize in advance for posting to such an old forum post.  I yearn for a greater understanding of this subject. 

Thanks in advance!


----------



## 71 dB

roweder said:


> This article makes it sound to me that when using 24 bit audio instead of 16 bit, the range of loudness that the audio file can describe becomes greater.  Why couldn't it instead increase the number of digital increments describing the difference between say -4db and -3db?
> 
> additionally, why couldn't the dynamic range of 16 bit audio be expanded by reducing the number of digital increments per 1db change?
> 
> ...


For simplicity PCM audio is linear. Each bit increases dynamic range by 6 dB. There is no need to increace dynamic range of 16 bit audio (16*6 dB = 96 dB). With dither it is more than enough (perceptual dynamic range up to 120 dB!). 24 bit (24*6 dB = 144 dB) is good in music production, but not needed in consumer audio. Even at studio all of that 24 bit dynamic range can't be used, because all electrical devices have higher background noise level. 

Remember, half of the 16 bits is used for negative values of the signal, so the maximum amplitude is 2^15 = 32768 = 0 dBFS. Then 23198 = -3 dBFS and 20675 = -4 dBFS. So, there are 2523 "increments"  between -4 dBFS and -3 dBFS. This number of increments isn't that interesting. The level of quantization noise/dither is what matters. 

There is µ-law/A-law 8 bit audio (companding algorithm) used in digital telecommunication that does what you propose. It results in quantization noise that fluctuates with the signal. The louder signal, the louder noise and vice versa. This was an early form of perceptual audio encoding.


----------



## pinnahertz

71 dB said:


> For simplicity PCM audio is linear. Each bit increases dynamic range by 6 dB. There is no need to increace dynamic range of 16 bit audio (16*6 dB = 96 dB). With dither it is more than enough (perceptual dynamic range up to 120 dB!). 24 bit (24*6 dB = 144 dB) is good in music production, but not needed in consumer audio. Even at studio all of that 24 bit dynamic range can't be used, because all electrical devices have higher background noise level.
> 
> Remember, half of the 16 bits is used for negative values of the signal, so the maximum amplitude is 2^15 = 32768 = 0 dBFS. Then 23198 = -3 dBFS and 20675 = -4 dBFS. So, there are 2523 "increments"  between -4 dBFS and -3 dBFS. This number of increments isn't that interesting. The level of quantization noise/dither is what matters.
> 
> There is µ-law/A-law 8 bit audio (companding algorithm) used in digital telecommunication that does what you propose. It results in quantization noise that fluctuates with the signal. The louder signal, the louder noise and vice versa. This was an early form of perceptual audio encoding.


Good explanation.  PCM is linear.  

µ-law/A-law are not "perceptually coded" at all, just companded, which is why they don't work very well for high quality audio.


----------



## gregorio (Jul 7, 2018)

roweder said:


> [1] This article makes it sound to me that when using 24 bit audio instead of 16 bit, the range of loudness that the audio file can describe becomes greater.
> [2] Why couldn't it instead increase the number of digital increments describing the difference between say -4db and -3db?
> [3] additionally, why couldn't the dynamic range of 16 bit audio be expanded by reducing the number of digital increments per 1db change?
> [3a] Is the difference in db between the integer value that the 16 or 24 bits convert to a fixed value?
> ...



1. Correct.

2. That IS exactly what it does! The basic concept (or mental image) of how digital audio works is actually pretty simple, once you've got your head around it. The difficult part about understanding the basics is "getting your head around it" in the first place, because it's based on some relatively complex maths (proposed 90 years ago and proven 70 years ago) and can be somewhat counter-intuitive just using layman's terms. In other words, without a high level of education in maths, there are certain aspects of how digital audio works that you just have to accept. With this in mind, I'll try to explain using your examples and terminology in order to help you "get your head around it". At the moment, your difficulty appears to be that you are thinking about the "range of loudness" and the "number of digital increments" as two different things, that it's possible to use a given number of bits to either describe the "range of loudness" or describe the "number of digital increments (between say -4dB and -3dB)". What you're going to need to do, is get your head around the fact that in effect they are the same thing, that the end result in digital audio of increasing "the number of digital increments" is a larger "range of loudness". You have to remember that all sound is just a sine wave (or combination of sine waves) and that the dB value of a sine wave/s constantly varies with time, at every instant in time the dB value is different. Intuition would therefore suggest that to get a perfect measurement/description of a sine wave we would need an infinite number of measuring points (in time) and an infinite number of measurement bits/values (to describe the infinite number of dB values). Obviously that's impossible and in effect our intuition is letting us down because that's not what digital audio attempts to do, digital audio approaches the problem from a different angle and that's why we have to dump our intuition and "get our head around" that different angle. Instead of trying to get a perfect measurement/description of a sine wave in the first place, digital audio takes a different approach, it works on the principle of deliberately taking imperfect measurements to start with and then using maths to predict and effectively correct the those imperfections/errors. In effect, the difference between the actual dB values of our sine wave and our measured/assigned digital values (the imperfections/errors) is converted into white noise. In other words, what we end up with is a PERFECT description of our sine wave/s plus noise, and this is true no matter how many digital bits/values we have available, even with only one bit (two values)! With only two digital values to represent an infinite number of actual dB values obviously we're going to get a huge amount of imperfection/error, what that means is we'll end-up with a perfect description of our sine wave/s and a huge amount of noise. As we increase our number of digital bits/values, so we decrease the amount of imperfections/errors and therefore we end up with (as always) a perfect description of our sine wave/s, plus a decreasing amount of noise. With this in mind, try reading the OP again, it will make a lot more sense.

3. Hopefully, after reading the above and then the OP again, you can answer your subsequent questions yourself but just in case: Reducing the "number of digital increments per 1dB change" reduces the accuracy of the values stored in those "increments" or put the other way around, increases the amount of imperfection/error (the amount of noise we end up with in addition to our perfect sine wave/s) and therefore decreases our dynamic range.
3a. I'm not sure I understand the question. The digital integer values stored are fixed values but what comes out of your DAC is not, it is a virtually perfect reconstruction of the (constantly varying) originally digitised sine wave/s plus an amount of noise, defined by the number of bits or by the analogue noise floor of your particular DAC.
3b. In a sense you're absolutely correct but the idea with 16bit audio is that the digital noise floor is always below any of the other noise floors present, such as: The noise floor of your reproduction equipment (amp, speakers or HPs for example), the noise floor of your listening environment and the noise floor of the recording itself (the recording venue, microphones, etc.). While it's true that many pieces of popular music only have a dynamic range of around 36dB or so and therefore 7bits would be sufficient, other types of music (and other audio content in general) can have a dynamic range significantly greater, up to around 60dB or even 70dB or so in a few cases.
3c. I suppose in theory you could but what would you gain from doing this? You can't get any more of a perfect description of the sine waves because it's already perfect and the digital noise floor with 16bit is already below audibility. All you would do is effectively eliminate the quiet parts of other pieces of music/audio content.

4. No problem, that's why I started this thread in the first place. Please be aware that my explanation above is a simplification, it doesn't take into account "noise-shaping" and some other considerations but those other considerations aren't going to make much sense unless you've got your head around the basics first.

G


----------



## castleofargh

roweder said:


> This article makes it sound to me that when using 24 bit audio instead of 16 bit, the range of loudness that the audio file can describe becomes greater.  Why couldn't it instead increase the number of digital increments describing the difference between say -4db and -3db?
> 
> additionally, why couldn't the dynamic range of 16 bit audio be expanded by reducing the number of digital increments per 1db change?
> 
> ...


to add the noob view to what the others said.
PCM is cool because for a given sample, you can easily correlate with an output voltage for the analog signal coming out of the DAC. let's say the first bit can be 0 or 1V, the second bit would code for 0 or 0.5V(half the voltage, giving us -6dB). the third bit codes for half the previous value, and so on. 
so one way to look at this makes you and I realize that more bits only let us code quieter and quieter signals. when we add more bits beyond 16, we're pretty much making the background noise having a much better resolution ^_^. that doesn't seem like the best use of bits yet in some ways it is.
here are other ways to look at this and draw different conclusions:
for example, if we wish to improve the resolution of the musical content you can take my example and think about expressing the signal with 3 bits, then look at what more bits would do(I'm not bothering with how we'd need to code for negative values too on a sine wave, just so that the example is dead simple). 
   let's say we want the code for 0.4V. with 3 bits 0.4V could be coded using only the second bit to get 0.5V.  that approximation isn't great. to increase the resolution between those first bits like you're suggesting, is actually what extra bits do to PCM. when you add an extra bit you can now code for 1V, 0.5V, 0.25V, and 0.125V and turn any combination ON anytime. now we can express our 0.4V as 0.25+0.125=0.375 instead of getting 0.5V with our 3bit code. and the more bits, the easier it becomes to zero in on the desired value.
so in practice extra bits are already increasing the resolution within the audio content like you suggest to do. 

another way to look at it is with the additive properties of waves. we can visualize the perfect audio signal, and say that every variation from it is simply another sound being added to the perfect one. the obvious idea for this would be one signal as music and any variation from that being noise. if you increase the resolution so that you can code for a quieter noise, the result is going to look closer to the perfect music signal. it's like turning down the second source of sound. what's left is a cleaner first source. in that respect, just having a low enough quantization noise is giving the music higher resolution. 
even if we only cared about not hearing that noise, we would probably wish to keep maybe 11 or 12bit. something like 36dB(about 6bits) wouldn't do for audibly clean background. you want to be sure that the quantization noise(from the value of the lower bit) is so far below the music that we won't notice it. including when the music is not stuck at 0dB all the time. all in all when you start considering pretty extreme examples, and using replay gain or other digital attenuation like changing the volume on the computer, if you still wish for the quantification noise to go unnoticed, you probably won't end up too far from 16bit in a PCM system. 


now what I talked about is the correlation between PCM and voltage amplitude of the signal coming out of the DAC. because before and after do correlate, so we can still talk as if a DAC was an perfect R2R design, and get the right results. but in truth most DACs nowadays don't work that way and instead change the amplitude a great many times between each sample. so in practice modern DACs are not using the PCM coding of the file, not as is anyway.


----------



## 71 dB

pinnahertz said:


> µ-law/A-law are not "perceptually coded" at all, just companded, which is why they don't work very well for high quality audio.



You just NEED to discredit people all the time

Not the way for example mp3s are, but the coding allows increased perceptual quality for speech for which it was developped. The dynamic range of speech exceeds linear 8 bit and µ-law/A-law coding allow bigger dynamic range to be fit to 8 bit at the expence of increased quantization noise which is less harmful than lack of dynamic range. Hence better perceptual sound quality = perceptual coding of a kind even if totally different from perceptual coding methods developped for music/high quality audio.

You know I am right with this so just let it be, ok?


----------



## roweder

Let me preface my reply by saying I am very surprised right now, this is the greatest number of intelligent responses I have ever gotten from posting in a forum before, EVER.  Even posting in forums about circuits, I haven't had responses like these before.  Thank you!



castleofargh said:


> to add the noob view to what the others said.
> PCM is cool because for a given sample, you can easily correlate with an output voltage for the analog signal coming out of the DAC. let's say the first bit can be 0 or 1V, the second bit would code for 0 or 0.5V(half the voltage, giving us -6dB). the third bit codes for half the previous value, and so on.
> so one way to look at this makes you and I realize that more bits only let us code quieter and quieter signals. when we add more bits beyond 16, we're pretty much making the background noise having a much better resolution ^_^. that doesn't seem like the best use of bits yet in some ways it is.



I think this was the explanation that finally cracked the code for me.  Let me take your example and use it to explain what I was thinking:

Lets use the 2 bit audio, we have 11=1.5V, 10=1V, 01=0.5V, and 00=0V.  My thoughts were, why couldn't you code your audio file so that... wait, is audio file where audiophile came from???.... code your audio file so that 00 is 0.2V instead.  That is what I was sort of thinking.  Because the way it was explained, I was thinking that the last bits were being "assigned" smaller sound levels as more bits were added to the front of the bit value.  But what you have explained has finally gotten me to think of this in what I think is the correct way?  That adding more bits simply makes those last bits quieter, and there's no way around that.  0000 0000 0000 0000 = 0V, and 0000 0000 0000 0000 0000 0000 = 0V, and that couldn't be changed.  Now I am understanding this.  I will read OP's post again now

from the original post:  "So, 24bit does add more 'resolution' compared to 16bit but this added resolution doesn't mean higher quality, it just means we can encode a larger dynamic range. " and "The only difference between 16bit and 24bit is 48dB of dynamic range (8bits x 6dB = 48dB) and *nothing else*."

Now I do understand why 24 bit can, and always will encode a larger dynamic range than 16 bit, but I'm not so clear on why the added resolution does not mean higher quality.  I get that with dither 16 bit can record an essentially perfect waveform with some noise, but wouldn't 24 bit with dither still be better than 16 bit with dither?  

With your replies and re-reading of OP's post, I've been able to understand why adding bit depth does increase the dynamic range and that wouldn't change.  

But does it not also increase the definition with which each note is described?  When I was learning A to D stuff for chemical instrumentation, we were essentially taught that more bits are always better, because you get closer and closer to describing the true value of your analogue voltage using bits.  Is this simply not true past a certain point in an audio signal?  Is this another example of diminishing returns?  Even if the difference is imperceptible to most listeners most of the time, is it not possible that say a uniquely distorted ~14kHz note could come through your system more similar to the way the artist intended using 24bit vs using 16 bit?  Even if you couldn't notice a difference in quality?  Say, maybe you don't really know which distortion is higher quality, but maybe if you knew the original sound, you could notice?







P.S. I have learned up to calculus 3 (3D calc), and differential equations, and I learned about the Nyquest frequency and how it works when learning about instrumentation for chemical spectroscopy & spectrometry.  If this qualifies me to handle higher level audio understanding, please do throw it at me!



P.P.S. What are your thoughts on this:


----------



## gregorio

roweder said:


> Now I do understand why 24 bit can, and always will encode a larger dynamic range than 16 bit, but I'm not so clear on why the added resolution does not mean higher quality. I get that with dither *16 bit can record an essentially perfect waveform* with some noise, but wouldn't 24 bit with dither still be better than 16 bit with dither?



No, 16bit is already perfect there is no "better" than perfect. In fact 1bit is perfect, just with a lot more noise. This is the basic tenet of the Nyquist theory presented 90 years ago and is what digital audio is based on. The added "resolution" of 24bit is what causes the larger dynamic range. I think the difficulty here might be that the term "resolution" in digital audio is marketed as meaning more or higher "quality" but that's not true, more "resolution" means more dynamic range, not more quality, because more quality cannot exist.

G


----------



## TheSonicTruth

castleofargh said:


> to add the noob view to what the others said.
> PCM is cool because for a given sample, you can easily correlate with an output voltage for the analog signal coming out of the DAC. let's say the first bit can be 0 or 1V, the second bit would code for 0 or 0.5V(half the voltage, giving us -6dB). the third bit codes for half the previous value, and so on.
> so one way to look at this makes you and I realize that more bits only let us code quieter and quieter signals. when we add more bits beyond 16, we're pretty much making the background noise having a much better resolution ^_^. that doesn't seem like the best use of bits yet in some ways it is.
> here are other ways to look at this and draw different conclusions:
> ...



Or, one could compress, limit, and make-up gain the audio to fit nicely into the MSB!  That tharez 'yoozin all the bits'  for yah! lol


----------



## ILikeMusic

gregorio said:


> No, 16bit is already perfect there is no "better" than perfect. In fact 1bit is perfect, just with a lot more noise.



I think I understand the application of Nyquist as it relates to sampling rate, but I'm having a little difficulty understanding the '1 bit' statement as it relates to bit depth. Recording amplitude in one bit would allow you to know if signal was present or not but I don't see how it could do any more than that, and certainly not result in a recognizable copy of the original modulated sine wave. What am I missing?


----------



## pinnahertz

71 dB said:


> You just NEED to discredit people all the time
> 
> Not the way for example mp3s are, but the coding allows increased perceptual quality for speech for which it was developped. The dynamic range of speech exceeds linear 8 bit and µ-law/A-law coding allow bigger dynamic range to be fit to 8 bit at the expence of increased quantization noise which is less harmful than lack of dynamic range. Hence better perceptual sound quality = perceptual coding of a kind even if totally different from perceptual coding methods developped for music/high quality audio.
> 
> You know I am right with this so just let it be, ok?


Here’s wha perceptual codinhg is:
https://en.m.wikipedia.org/wiki/Perceptual_audio_coder

µ-law/A-law Is not based on perception at all.  So I guess I don’t know you’re right about this after all.

I’m not discrediting anyone, I’m adding accurate information.


----------



## gregorio (Jul 7, 2018)

ILikeMusic said:


> Recording amplitude in one bit would allow you to know if signal was present or not but I don't see how it could do any more than that, and certainly not result in a recognizable copy of the original modulated sine wave. What am I missing?



Essentially with only 1 bit/2 values to play with, you would only be able to encode if the amplitude of the signal being digitized is increasing or decreasing. Obviously, this is highly imperfect or to put it another way, we have a great deal of error (resulting in a perfect waveform and a great deal of noise). However, using aggressive noise-shaped dither, all that noise can be placed into the ultrasonic frequency range, assuming a high sample rate and therefore a large freq range to redistribute that huge amount of noise. BTW, this isn't just theory, Sony actually produced such a digital audio format/product nearly 20 years ago, it's called SACD.

G


----------



## castleofargh

roweder said:


> Lets use the 2 bit audio, we have 11=1.5V, 10=1V, 01=0.5V, and 00=0V. My thoughts were, why couldn't you code your audio file so that... wait, is audio file where audiophile came from???.... code your audio file so that 00 is 0.2V instead. That is what I was sort of thinking.



from a "language" perspective, of course we could assign any value to anything we want and start wherever we like. but we have to consider the practical application. how would we go to implement that in the DAC? no signal set at 0.2V, does it mean we have DC voltage on every silent passage?
in practice if a DAC can output 2V max that value will be affect as 0dB(loudest signal) and we go down from there in a 6dB increment for each bit value, and as low as the DAC can. I'm not sure how we could do differently. if we took a song where the lowest signal is at -36dB, and somehow 0dB was affected to the max output and -36dB as the LSB value, that would result in quantization noise at -36dB which really wouldn't improve fidelity. 




ILikeMusic said:


> I think I understand the application of Nyquist as it relates to sampling rate, but I'm having a little difficulty understanding the '1 bit' statement as it relates to bit depth. Recording amplitude in one bit would allow you to know if signal was present or not but I don't see how it could do any more than that, and certainly not result in a recognizable copy of the original modulated sine wave. What am I missing?


the obvious example of what he means is DSD. you code in 1bit but you can achieve an equivalent to 24/96 or even higher without too much difficulty. it's one of the best examples to show that whatever we have it's in the end the right accurate signal, and some noise. when we have a way to push the noise around, what's left is the proper accurate signal. 
of course if you just code stuff in 1bit and reconstruct them like that, you just get noisy crap. it can't work on its own as a one step does all, but then again few digital designs do.


----------



## 71 dB (Jul 7, 2018)

pinnahertz said:


> Here’s wha perceptual codinhg is:
> https://en.m.wikipedia.org/wiki/Perceptual_audio_coder
> 
> µ-law/A-law Is not based on perception at all.  So I guess I don’t know you’re right about this after all.
> ...



It's semantics bro! It is based on _perception_, because µ-law/A-law coding increases intellibility of speech. What is inaccurate about that?

Is it my fault µ-law/A-law is not called what it is for historical reasons? Later much more advanced coders don't change things.

_*"μ-law encoding effectively reduced the dynamic range of the signal, thereby increasing the coding efficiency while biasing the signal in a way that results in a signal-to-distortion ratio that is greater than that obtained by linear encoding for a given number of bits. This is an early form of perceptual audio encoding."*_

https://en.wikipedia.org/wiki/Μ-law_algorithm


----------



## ILikeMusic (Jul 7, 2018)

gregorio said:


> Obviously, this is highly imperfect or to put it another way, we have a great deal of error (*resulting in a perfect waveform* and a great deal of noise).


The 'perfect waveform' is the part I'm missing, at least in the amplitude domain. My understanding is that in terms of _frequency, _sample rate needs to be only twice that of the highest frequency you intend to encode since that will provide sufficient sample points to fit a sine function that will allow exact reproduction of the input, hence a perfect waveform in the frequency domain. However, with regard to _bit depth_ you are attempting to quantize amplitude and there will always be some error. I understand how this error becomes broadband noise but I don't understand how we reconstruct a _perfect_ reproduction of amplitude in the reproduced signal. Is the idea that dither provides sufficient randomization to eliminate all error down to a mathematical certainty?


----------



## pinnahertz

71 dB said:


> It's semantics bro! It is based on _perception_, because µ-law/A-law coding increases intellibility of speech. What is inaccurate about that?


 Did your read the link?


71 dB said:


> Is it my fault µ-law/A-law is not called what it is for historical reasons? Later much more advanced coders don't change things.
> 
> _*"μ-law encoding effectively reduced the dynamic range of the signal, thereby increasing the coding efficiency while biasing the signal in a way that results in a signal-to-distortion ratio that is greater than that obtained by linear encoding for a given number of bits. This is an early form of perceptual audio encoding."*_
> 
> https://en.wikipedia.org/wiki/Μ-law_algorithm


I completely disagree with that based on the link that defines what perceptual coding is.  "an early form" is not the same as the actual thing defined completely differently.


----------



## old tech

ILikeMusic said:


> The 'perfect waveform' is the part I'm missing, at least in the amplitude domain. My understanding is that in terms of _frequency, _sample rate needs to be only twice that of the highest frequency you intend to encode since that will provide sufficient sample points to fit a sine function that will allow exact reproduction of the input, hence a perfect waveform in the frequency domain. However, with regard to _bit depth_ you are attempting to quantize amplitude and there will always be some error. I understand how this error becomes broadband noise but I don't understand how we reconstruct a _perfect_ reproduction of amplitude in the reproduced signal. Is the idea that dither provides sufficient randomization to eliminate all error down to a mathematical certainty?


I believe, and I'm no expert here so others may chime in, that dither just adds white noise to the signal which decorrelates the amplitude errors.  So the errors are still there but it now randomised at the cost of a higher noise floor which sounds like tape hiss (but, at least with 16bits, you are unlikely to hear the hiss unless it is a quiet passage played at insane loud levels).  The main point is that there is no perfection even in the natural audio world, let alone audio recording/playback technology.  With analog recordings/playback the quantiitasion like imperfections manisfests itself as hiss, which is the sound of the random errors, with digital there is no random errors, they are precise errors so all dither does is randomises the errors in a similar way which analog media does through its inherent imperfections.


----------



## old tech

roweder said:


> Let me preface my reply by saying I am very surprised right now, this is the greatest number of intelligent responses I have ever gotten from posting in a forum before, EVER.  Even posting in forums about circuits, I haven't had responses like these before.  Thank you!
> 
> Now I do understand why 24 bit can, and always will encode a larger dynamic range than 16 bit, but I'm not so clear on why the added resolution does not mean higher quality.  I get that with dither 16 bit can record an essentially perfect waveform with some noise, but wouldn't 24 bit with dither still be better than 16 bit with dither?


While Greg explained it well, I find many people understand this concept a bit better after reading this article from Ian Sheppard.

http://productionadvice.co.uk/no-stair-steps-in-digital-audio/


----------



## TheSonicTruth

old tech said:


> While Greg explained it well, I find many people understand this concept a bit better after reading this article from Ian Sheppard.
> 
> http://productionadvice.co.uk/no-stair-steps-in-digital-audio/




I prefer this guy's presentation of the matter over that of the Wolf in Shep's clothing(!):  



Skip ahead to 5:58 for the nitty-gritty on "stair steps" in digital sound.


----------



## pinnahertz

old tech said:


> 1. I believe, and I'm no expert here so others may chime in, that dither just adds white noise to the signal which decorrelates the amplitude errors.  So the errors are still there but it now randomised at the cost of a higher noise floor which sounds like tape hiss (but, at least with 16bits, you are unlikely to hear the hiss unless it is a quiet passage played at insane loud levels).  2. The main point is that there is no perfection even in the natural audio world, let alone audio recording/playback technology.  3. With analog recordings/playback the quantiitasion like imperfections manisfests itself as hiss, which is the sound of the random errors, with digital there is no random errors, they are precise errors so all dither does is randomises the errors in a similar way which analog media does through its inherent imperfections.


Dither also involves noise-shaping, the result of which does not raise the audible noise floor.  Dither adds a small amount of randomization so that signal levels below the LSB can be recorded, but it doesn't need to be in the most audible spectrum to work, so noise-shaping moves noise power into the portion of the spectrum that is least audible.  There are several types of noise found in analog tape, including the basic tape hiss which is not shaped of course, but also other types of signal-dependent noise.  Dither does not sound like tape noise.

2. The concept of "perfection" definitely depends on point of view.  Even on-line definitions have a range from the absolute to the practical.  I favor the practical, which means there are many examples of perfection, even in audio.  But it seems a silly point to debate.

3. There is no quantization in analog recording.  The imperfections caused by the media and method are all consequences of the physical nature of magnetics and physics.  They happen to distort, modulate, and add noise to the desired signal.  Tape his is not the sound of random errors, it's caused by the size of the magnetic particles involved.  If you recorded no signal, not even an erase signal, you'd still have tape hiss.  It's not an error, it's the noise floor. Small, but important difference.  Theoretical digital has no random errors, but practical digital always does in the form of Least Significant Bit jitter, that's the built-in dithering of the LSB.  The errors therefore are not precise, but rather always slightly dithered. Dithering further randomizes using a signal that doesn't raise the apparent noise floor.


----------



## old tech

pinnahertz said:


> Dither also involves noise-shaping, the result of which does not raise the audible noise floor.  Dither adds a small amount of randomization so that signal levels below the LSB can be recorded, but it doesn't need to be in the most audible spectrum to work, so noise-shaping moves noise power into the portion of the spectrum that is least audible.  There are several types of noise found in analog tape, including the basic tape hiss which is not shaped of course, but also other types of signal-dependent noise.  Dither does not sound like tape noise.
> 
> 2. The concept of "perfection" definitely depends on point of view.  Even on-line definitions have a range from the absolute to the practical.  I favor the practical, which means there are many examples of perfection, even in audio.  But it seems a silly point to debate.
> 
> 3. There is no quantization in analog recording.  The imperfections caused by the media and method are all consequences of the physical nature of magnetics and physics.  They happen to distort, modulate, and add noise to the desired signal.  Tape his is not the sound of random errors, it's caused by the size of the magnetic particles involved.  If you recorded no signal, not even an erase signal, you'd still have tape hiss.  It's not an error, it's the noise floor. Small, but important difference.  Theoretical digital has no random errors, but practical digital always does in the form of Least Significant Bit jitter, that's the built-in dithering of the LSB.  The errors therefore are not precise, but rather always slightly dithered. Dithering further randomizes using a signal that doesn't raise the apparent noise floor.


Thanks for the information, I'm always learning something.  While dither itself does not sound like tape hiss, I always thought that the effect it has on the noise floor (ie the noise addition) does.  That is certainly how it sound like to me when, for example, 8 bits is dithered and it is how many others describe it.  The hiss you hear from vinyl records, when isolated from other noises of vinyl playback sounds like tape hiss too.  I can hear this on my records that were sourced from a digital recording, so it can't be a recording of tape hiss from the master.

Another question, and this is more 'absolute than practical', I thought quantitisation exists in all audio, even in the natural world but is randomised by natural errors or man-made errors.  The paper from from St Andrews Uni demonstrates, for example, quantitisation effects on vinyl records though it is randomised by tracking errors from stylus in the groove, and the movement of the actual vinyl molecules pressed against the stylus.  Air molecules have the same effect in the natural world.


----------



## 71 dB

pinnahertz said:


> Did your read the link?



Yes.



pinnahertz said:


> I completely disagree with that based on the link that defines what perceptual coding is.  "an early form" is not the same as the actual thing defined completely differently.



You're splitting hairs here. Early baroque (say as Johann Rosenmüller or Heinrich Scheidemann) is a form of baroque. Nobody is saying Rosenmüller is the same as Handel, but both are baroque music.


----------



## pinnahertz

old tech said:


> Thanks for the information, I'm always learning something.  While dither itself does not sound like tape hiss, I always thought that the effect it has on the noise floor (ie the noise addition) does.  That is certainly how it sound like to me when, for example, 8 bits is dithered and it is how many others describe it.  The hiss you hear from vinyl records, when isolated from other noises of vinyl playback sounds like tape hiss too.  I can hear this on my records that were sourced from a digital recording, so it can't be a recording of tape hiss from the master.
> 
> Another question, and this is more 'absolute than practical', I thought quantitisation exists in all audio, even in the natural world but is randomised by natural errors or man-made errors.  The paper from from St Andrews Uni demonstrates, for example, quantitisation effects on vinyl records though it is randomised by tracking errors from stylus in the groove, and the movement of the actual vinyl molecules pressed against the stylus.  Air molecules have the same effect in the natural world.



I must take exception with the at least some of the analysis in that "paper", especially if the conclusion is to be quantization in vinyl being comparable at all to digital.  

The paper mentions the thickness of a layer of a crystalline carbon (diamond) as an example, then goes on:

_"Each carbon atom has an effective diameter of around half a nanometre so the thickness of each layer will be approximately 0·5 nm. The position of the stylus is determined by resting on top of the uppermost layers of atoms. Hence we can see that the stylus position will be roughly quantised by the finite thickness of the atomic layers. When playing a sinewave whose peak size is 8 microns the movement of the stylus would take place in 1 nm steps. Instead of smoothly varying, the stylus offset would therefore always adopt one of the set of available levels,
	

	
	
		
		

		
			





, where m is an integer and 
	

	
	
		
		

		
		
	


	




 is the thickness of the atomic layers. The effect is to divide the 
	

	
	
		
		

		
		
	


	




microns swing of a 0 dB 1 kHz sinewave into 32,000 steps — j*ust as if the signal had passed through an ADC! "*_

The problem with the conclusion here is that we'd have to have a stylus that contacts only a single atom at a time to define a specific quantization level that would be comparable in nature to digital quantization, which is not the case, far from it.   The stylus contacts a large number of carbon atoms at any moment, and the number changes continuously.  

He's correct that technically (a very specific point here) any system that takes a large sampling of data and reduces it to a smaller set is quantization, so technically, analog recording qualifies, but specifically, it's nothing like what happens in a digital system which assigns specific and repeatable values to the quantized data.  So his concluding statement is technically correct, but specifically wrong.  It's not at all like the signal was passed through an ADC, other than to match the very general definition of quantization.

I find it pointless to micro-analyze and system like this because apart from the highly specific definitions, the results are completely different, with analog adding quite a number of additional distortions and signals that have nothing to do with its reduced data set at all.  It's clearly "spun" to make a point, yet the point is, in the end, lost because of the results.


----------



## pinnahertz

71 dB said:


> Yes.


And yet you are still arguing...



71 dB said:


> You're splitting hairs here. Early baroque (say as Johann Rosenmüller or Heinrich Scheidemann) is a form of baroque. Nobody is saying Rosenmüller is the same as Handel, but both are baroque music.


Agreed with your example, but µ-law/A-law have none of the things that differentiate perceptual coding.  More like comparing Handel to Cage.  Both are music, but that's pretty much where it ends.


----------



## 71 dB

pinnahertz said:


> 1. And yet you are still arguing...
> 
> 2. Agreed with your example, but µ-law/A-law have none of the things that differentiate perceptual coding.  More like comparing Handel to Cage.  Both are music, but that's pretty much where it ends.



1. Unfortunately.
2. µ-law/A-law is for speech, telecommunication. Later perceptual coding methods are for high quality audio/music. Both are perceptual coding even if completely different.


----------



## castleofargh

it's not using a form of psychoacoustic coding. but as a codec for speech, it's hard to argue that perception wasn't a relevant factor. everybody won





oh you bunch of highly educated old kids


----------



## pinnahertz

71 dB said:


> 1. Unfortunately.
> 2. µ-law/A-law is for speech, telecommunication. Later perceptual coding methods are for high quality audio/music. Both are perceptual coding even if completely different.


I’m having extreme difficulty caring any less about this.


----------



## bigshot

It can't be over until someone cries.


----------



## danadam

ILikeMusic said:


> The 'perfect waveform' is the part I'm missing, at least in the amplitude domain. My understanding is that in terms of _frequency, _sample rate needs to be only twice that of the highest frequency you intend to encode since that will provide sufficient sample points to fit a sine function that will allow exact reproduction of the input, hence a perfect waveform in the frequency domain. However, with regard to _bit depth_ you are attempting to quantize amplitude and there will always be some error. I understand how this error becomes broadband noise but I don't understand how we reconstruct a _perfect_ reproduction of amplitude in the reproduced signal. Is the idea that dither provides sufficient randomization to eliminate all error down to a mathematical certainty?


(the way I understand it) At some point in ADC you have samples with perfect amplitude that you need to quantize (at least conceptually):

```
input[n] -> [ quantization ] -> output[n]
```
What happens in "quantization" box? We modify sample value so that it can be represented digitally. We can write:

```
input[n] + q[n] = output[n]
```
So you can see that the output, our digitalized signal, *is*: input which is *perfect* plus some error/noise. To your question "how we reconstruct a _perfect_ reproduction of amplitude", we don't do anything specific. Perfect input is "baked in" in the output. So you just play the output and you get perfect input with noise.

If you were asking how we get perfect input without the noise (maybe you did?), then that is not possible, as far as I know.


----------



## ILikeMusic

Thanks all, got it now I think. My problem was that I was still looking at amplitude sampling as being recorded and reproduced as some fixed temporal event (sigh, stairsteps I guess, even though I thought I knew that wasn't the case) when just as in the frequency domain this isn't actually the case. The actual process is non-intuitive and difficult to visualize without a clear explanation and as a result is probably understood only by a small fraction of a percentage of those professing opinions concerning 'hi-res' audio. This is a truly great thread, thanks.

My next question, why does copper wire sound warmer than silver?


----------



## TheSonicTruth (Jul 8, 2018)

ILikeMusic said:


> Thanks all, got it now I think. My problem was that I was still looking at amplitude sampling as being recorded and reproduced as some fixed temporal event (sigh, stairsteps I guess, even though I thought I knew that wasn't the case) when just as in the frequency domain this isn't actually the case. The actual process is non-intuitive and difficult to visualize without a clear explanation and as a result is probably understood only by a small fraction of a percentage of those professing opinions concerning 'hi-res' audio. This is a truly great thread, thanks.
> 
> My next question, why does copper wire sound warmer than silver?



Sure does - if you add a couple dB between 200-300Hz at a low-Q and gently roll off the top, in the master!


----------



## gregorio (Jul 9, 2018)

ILikeMusic said:


> [1] The 'perfect waveform' is the part I'm missing, at least in the amplitude domain...Thanks all, got it now I think. My problem was that I was still looking at amplitude sampling as being recorded and reproduced as some fixed temporal event (sigh, stairsteps I guess, even though I thought I knew that wasn't the case) when just as in the frequency domain this isn't actually the case.
> [2] The actual process is non-intuitive and difficult to visualize without a clear explanation and as a result is probably understood only by a small fraction of a percentage of those professing opinions concerning 'hi-res' audio.



1. Maybe the difficulty you were/are having is that you were trying to separate the issue into two different "domains" (amplitude and frequency)? While it's sometimes useful to do this for the sake of "vizualization" in reality they are not separate/different domains, they're exactly the same thing. A sine wave (for example) is effectively defined as: An increasing amplitude until a "peak" is reached, then a decreasing amplitude until the "trough" is reached and then an increasing amplitude again until the starting point is reached. We call this a "cycle" and frequency is simply the number of cycles per second. In other words, Frequency = Amplitude (over time).

2. As I mentioned in my first response to you, it's all a question of getting your head around it (or "vizualing" as you put it) and that requires an explanation which works for you personally. I'm not sure that "the actual process is non-intuitive", I think it depends on the "visualization" you have to start with. For example, if you take someone who's never thought about how digital audio works, who's effectively a "blank canvas" with no preconceived "vizualization" then I don't think the process is counter-intuitive but it's more difficult and more counter-intuitive if you do have preconceived notions of how it works (the "stairstep" notion for instance). Many audiophiles for example, seem to have the notion that digital audio is effectively analogue audio but with digital data, IE. Analogue audio creates an "analogy" of actual sound waves using an electrical current and digital audio creates an "analogy" of actual sound waves using digital data. This view/"vizualisation" is incorrect and leads to a bunch of further incorrect assumptions, such as; more data (bits or sample points) results in digital audio data which is a closer/higher resolution "analogy" to the actual sound waves. In reality though, digital audio is effectively just a sequence of data points which allows the sound waves to be "reconstructed" through the application of some mathematical processes, digital audio data is NOT analogous to the sound waves.



old tech said:


> [1] I believe, and I'm no expert here so others may chime in, that dither just adds white noise to the signal which decorrelates the amplitude errors.
> [1a] So the errors are still there but it now randomised at the cost of a higher noise floor which sounds like tape hiss ...



1. You're not exactly wrong but not exactly correct either. The way you appear to be looking at dither leads to some incorrect conclusions/assumptions. Instead of thinking about dither in terms of actual white noise, try more in terms of what it actually is, a mathematical function. Standard dither is usually abbreviated to TDPF, a Triangular Probability Density Function, which is a rather off-putting term for the layman but can be thought of as: A sort of mathematical equation which randomises errors, the end result effectively being; that ALL the error is converted into white noise.
1a. By looking at dither this way, you can hopefully see that your statement is incorrect: Firstly, the errors are not "still there", the errors are completely gone, they've been converted into white noise and Secondly, the "noise" is not higher, it's the same. The difference is that with dither we end up with a constant low level amount of white noise, while without dither we end up a non-constant amount of signal distortion but in both cases we end up with the same overall "amount". This is of course the logical conclusion using this view, as all we're doing is converting the error into white noise. ... Dither is a prerequisite of digital audio, without it the conditions required to achieve the "Sampling Theorem" cannot be met, in much the same way as not applying an anti-alias filter at half the sampling rate fails to meet the required conditions. Dither is therefore always automatically applied during the quantisation process, as is an anti-alias filter, both are intrinsic to the process of digital audio. In other words, dither does not raise the noise floor, it's what defines the noise floor in the first place!

We've also now got what's called noise-shaped dither, which became commercially available in the early 1990's, in response to the growing requirement of "re-quantisation". Re-quantisation became a requirement when high-end digital recording and mixing moved beyond 16bit (initially to 20bit) and therefore needed to be re-quantised down to 16bit for consumer distribution. Without another round of dither, the re-quantisation process would introduce "truncation error", which is effectively a slightly more severe form of quantisation error. Noise-shaped dither was introduced to effectively maintain the 20bit dynamic range but in a 16bit file format. As pinnahertz effectively stated, our resultant white noise is "shaped", it's no longer "white", it's concentrated in areas where our hearing is least sensitive and is therefore inaudible.

BTW, all the above is not exactly correct or incorrect either! It's just another way of looking at the issue, a way which avoids some incorrect conclusions/assumptions.



71 dB said:


> [1] Hence better perceptual sound quality = perceptual coding of a kind even if totally different from perceptual coding methods developped for music/high quality audio.
> [2] You're splitting hairs here.



1. Throughout your argument with pinnahertz you seem to have missed the fact that "perceptual coding" has a specific and well defined meaning. You are incorrectly equating better perceptual sound quality with perceptual coding. "Perceptual coding" at least partially relies on "auditory masking" and reducing the amount of data by removing masked frequencies. On the other hand, "Better perceptual sound quality" can be achieved in numerous ways which do not rely on or even directly involve "auditory masking". A simple EQ or filter, noise reduction, compression, expansion or other processes can all produce better perceptual sound quality but are NOT "perceptual coding".

2. No, he wasn't. It's an important distinction with wide ranging ramifications. Plus, if we're going to accept your definition of "perceptual coding", what new/different term are we going to use for actual perceptual coding? It seems to be a bit of a trend, someone makes an incorrect statement of fact and when called out on it, responds with "you're splitting hairs", "just because I can't find the right words doesn't mean I'm wrong" or "you just like to argue for argument sake". I'm not sure if this type of response is simply an attempt to deflect or minimise the fact they've been caught making-up (or recounting) incorrect facts or whether it's because they really don't understand "science" or enough about "sound" to appreciate why made-up, incorrect statements of fact are so important.

G


----------



## drtechno

Just to cook your noodle a bit @pinnahertz , I have three types of dither, besides the 4 variations of uv22hr, just saying....


----------



## drtechno

danadam said:


> If you were asking how we get perfect input without the noise (maybe you did?), then that is not possible, as far as I know.



Well, to have "perfect" input would mean the ADC would have to have an infinite bandwidth with a virtual ground that is finite so that the signal to noise ratio would be infinite, all harmonics are captured.  That is your "perfect" signal recording.

To have your perfect DAC, it would require an infinite slew rate, thus reproducing the signal and its harmonics that was recorded and the final line stage should continue this high slew rate at the same time be able to drive the transmission line without voltage reduction (due to current fold-over commonly in semiconductors).


----------



## gregorio

drtechno said:


> Well, to have "perfect" input would mean the ADC would have to have an infinite bandwidth ...



I presume you have some proof that the Nyquist/Shannon Theorem is incorrect? Or, are you saying that musical instruments and sounds produce harmonics of an infinite bandwidth and that microphones and mic pre-amps have an infinite bandwidth? If so, again, some proof or reliable evidence please. This is the sound science forum and if you are going to make claims which contradict the known science, then you MUST provide reliable evidence! Same for your claims of slew rate.

G


----------



## 71 dB

gregorio said:


> 1. Throughout your argument with pinnahertz you seem to have missed the fact that "perceptual coding" has a specific and well defined meaning. You are incorrectly equating better perceptual sound quality with perceptual coding. "Perceptual coding" at least partially relies on "auditory masking" and reducing the amount of data by removing masked frequencies. On the other hand, "Better perceptual sound quality" can be achieved in numerous ways which do not rely on or even directly involve "auditory masking". A simple EQ or filter, noise reduction, compression, expansion or other processes can all produce better perceptual sound quality but are NOT "perceptual coding".
> 
> 2. No, he wasn't. It's an important distinction with wide ranging ramifications. Plus, if we're going to accept your definition of "perceptual coding", what new/different term are we going to use for actual perceptual coding? It seems to be a bit of a trend, someone makes an incorrect statement of fact and when called out on it, responds with "you're splitting hairs", "just because I can't find the right words doesn't mean I'm wrong" or "you just like to argue for argument sake". I'm not sure if this type of response is simply an attempt to deflect or minimise the fact they've been caught making-up (or recounting) incorrect facts or whether it's because they really don't understand "science" or enough about "sound" to appreciate why made-up, incorrect statements of fact are so important.
> 
> G



µ-law/A-law allows coding larger dynamic range in just 8 bits (data reduction) using auditory masking (loud sounds mask quieter sounds). Wikipedia calls it an early perceptual coding method, so take your anger to them.


----------



## gregorio

71 dB said:


> µ-law/A-law allows coding larger dynamic range in just 8 bits (data reduction) using auditory masking (loud sounds mask quieter sounds).



No, it does not use "auditory masking" (either frequency masking or temporal masking), it just uses compression/expansion. It's nowhere near as sophisticated as perceptual coding, which analyses about 1000 samples at a time, divides the frequency spectrum into 500 or so frequency bands, compares the results with a psycho-acoustic model and discards those bands which fall under the masking threshold. This is all completely different to just simple audio compression, there is no audio compression in perceptual coding!

G


----------



## TheSonicTruth (Jul 9, 2018)

gregorio said:


> No, it does not use "auditory masking" (either frequency masking or temporal masking), it just uses compression/expansion. It's nowhere near as sophisticated as perceptual coding, which analyses about 1000 samples at a time, divides the frequency spectrum into 500 or so frequency bands, compares the results with a psycho-acoustic model and discards those bands which fall under the masking threshold. This is all completely different to just simple audio compression, there is no audio compression in perceptual coding!
> 
> G



Gregorio wrote: "there is no audio compression in perceptual coding!"

Finally, something we BOTH have been trying to hammer through thick skulls:  Dynamic and lossy data compression are two different frickn' things!  lol!

I personally use the term 'data-reduction' when referring to lossy/lossless codecs.    MP3 for example = lossy data reduction.  NOT 'lossy compression'.  Prevents loads of confusion, of which there is plenty of in this thing we love called digital audio!


----------



## drtechno

gregorio said:


> I presume you have some proof that the Nyquist/Shannon Theorem is incorrect? Or, are you saying that musical instruments and sounds produce harmonics of an infinite bandwidth and that microphones and mic pre-amps have an infinite bandwidth? If so, again, some proof or reliable evidence please. This is the sound science forum and if you are going to make claims which contradict the known science, then you MUST provide reliable evidence! Same for your claims of slew rate.
> 
> G


First off Nyquest has nothing to do with the actual physical working of the musical instrument. Because I can change my strings on my guitar from D'adario silk-nsteel to nylon or martin steel stings and get a different set of harmonics from the instrument. Sampling rates for digital never coincided with any real analog standard but itself Nyquist never included all of the harmonics of the either. ADC don't have linear signal to noise ratios either,  as well as bandwidth is different depending on how the ADC ic is assembled in the circuit. Its a shame the ic mfgs don't really publish the chart of Vref impedance to signal to noise and bandwidth. otherwise I would post one. But all what I said is common engineering knowledge on what happens there and what variables effect the track and hold circuit, and its settling time. About Slew rate with DACs: This is why high speed op amps are integrated into the DAC ic to overcome the old design issues of the I to V RC network (that generates a loss before amplification).

I will, just for you, look at google, and see what things pop up:

Look at pg18: http://www.delftek.com/wp-content/uploads/2012/04/National_ABCs_of_ADCs.pdf 
Here is a glossary of terms that apply to ADCs and DACs: https://www.maximintegrated.com/en/app-notes/index.mvp/id/641


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## bigshot (Jul 9, 2018)

drtechno said:


> Well, to have "perfect" input would mean the ADC would have to have an infinite bandwidth with a virtual ground that is finite so that the signal to noise ratio would be infinite, all harmonics are captured.  That is your "perfect" signal recording.



Perfect means everything humans can hear perfectly reproduced. Bats and dogs might need a different kind of DAC than we do.

Inaudible harmonics are inaudible.


----------



## gregorio

drtechno said:


> [1] First off Nyquest has nothing to do with the actual physical working of the musical instrument. Because I can change my strings on my guitar from D'adario silk-nsteel to nylon or martin steel stings and get a different set of harmonics from the instrument. Sampling rates for digital never coincided with any real analog standard but itself Nyquist never included all of the harmonics of the either.
> [2] ADC don't have linear signal to noise ratios either ... [2a] as well as bandwidth is different depending on how the ADC ic is assembled in the circuit.



1. Correct, the Nyquist/Shannon Theorem has nothing to do with physical musical instruments or any sounds/harmonics, it covers ALL instruments/sounds/harmonics. So, are you disputing the Nyquist/Shannon Theorem or not? If not, then you must be saying that musical instruments produce an infinite number of harmonics, please provide some supporting evidence for that claim.

2. What has that got to do with anything?
2a. Not to any significance, it depends on the characteristics of the anti-alias filters and of course the Nyquist point of a particular sample rate but again, what has this got to do with your assertion that ADCs have to have infinite bandwidth?

G


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## ILikeMusic

Harmonic energy within the Nyquist limit should be recorded and reproduced accurately and harmonic energy above the limit (assuming the typical 20khz after filtering) would be inaudible so it wouldn't be relevant. Or as bigshot so succinctly put it, 'inaudible harmonics are inaudible', so why would it even be desirable to have infinite bandwidth?


----------



## 71 dB

gregorio said:


> No, it does not use "auditory masking" (either frequency masking or temporal masking), it just uses compression/expansion. It's nowhere near as sophisticated as perceptual coding, which analyses about 1000 samples at a time, divides the frequency spectrum into 500 or so frequency bands, compares the results with a psycho-acoustic model and discards those bands which fall under the masking threshold. This is all completely different to just simple audio compression, there is no audio compression in perceptual coding!
> 
> G


Quantization noise fluctuates with signal level. Signal masks noise.


----------



## pinnahertz

71 dB said:


> µ-law/A-law allows coding larger dynamic range in just 8 bits (data reduction) using auditory masking (loud sounds mask quieter sounds). Wikipedia calls it an early perceptual coding method, so take your anger to them.


Do you know who writes the Wiki articles?  Anyone!  Do you know who can edit them?  Anyone!


----------



## pinnahertz

71 dB said:


> Quantization noise fluctuates with signal level. Signal masks noise.


It's incidental, not intentional.  If you define perceptual coding by only including the fact that masking occurs, then every recording system uses an early form of perceptual coding.  That means RIAA is perceptual coding, Tape EQ is perceptual coding, heck the Edison wax cylinder employed an early form of perceptual coding.  All of that is complete nonsense.

The part of coding that is specifically designed to exploit masking, the, "perceptual" part, is what's missing.

You need to work on your definitions, that's all I'm saying.


----------



## SoundAndMotion

pinnahertz said:


> Do you know who writes the Wiki articles?  Anyone!  Do you know who can edit them?  Anyone!


Do you know who writes forum posts?

How would you define "perceptual coding"? I found a nice definition that seems good to me: "Perceptual Coding": Lossy compression that takes advantage of limitations in human perception. In perceptual coding, audio data is selectively removed based on how unlikely it is that a listener will notice the removal.

Do you know where the log relationship for A-law and µ-law come from? Look up Weber-Fechner for a clue.


----------



## pinnahertz

SoundAndMotion said:


> Do you know who writes forum posts?


It's how Wikipedia works: Anyone...even you!....could write an article.  And anyone...even me!....could edit it and correct it.  It's the entire principle behind the site.


SoundAndMotion said:


> How would you define "perceptual coding"? I found a nice definition that seems good to me: "Perceptual Coding": Lossy compression that takes advantage of limitations in human perception. In perceptual coding, audio data is selectively removed based on how unlikely it is that a listener will notice the removal.


Not bad.  And by that definition, µ-law is not perceptual coding because it does not specificaly base it's action on the likelihood of a listener hearing what's been removed.


----------



## 71 dB

pinnahertz said:


> Do you know who writes the Wiki articles?  Anyone!  Do you know who can edit them?  Anyone!



Well, go and correct the mistakes then! You use a lot of energy correcting me, one person while anyone can read Wikipedia.


----------



## 71 dB

pinnahertz said:


> It's incidental, not intentional.  If you define perceptual coding by only including the fact that masking occurs, then every recording system uses an early form of perceptual coding.  That means RIAA is perceptual coding, Tape EQ is perceptual coding, heck the Edison wax cylinder employed an early form of perceptual coding.  All of that is complete nonsense.
> 
> The part of coding that is specifically designed to exploit masking, the, "perceptual" part, is what's missing.
> 
> You need to work on your definitions, that's all I'm saying.


Why would any sound quality reduction be _intentional _? All of it is incidental, but in a controlled way. We know how harmful it is from perceptual point of view. Perceptual coding uses data reduction and you don't have that with analog formats. Even if wax cylinders were perceptual coding so what? Changes nothing.


----------



## gregorio

71 dB said:


> [1] Quantization noise fluctuates with signal level. Signal masks noise.
> [2] Perceptual coding uses data reduction and you don't have that with analog formats. [2a] Even if wax cylinders were perceptual coding so what? Changes nothing.



1. That's great, it means there's no need whatsoever for dither because quantisation noise is apparently masked. All these decades of dither and dither development wasted ... do you think we should tell someone? 

2. So now, in effect, you're contradicting yourself because µ-law/A-law are analogue processes!
2a. By your definition, wax cylinders are perceptual coding, vinyl with RIAA are perceptual coding, cassettes with bias are perceptual coding, CDs with emphasis are perceptual encoding, despite your claim in point #1, dither is perceptual coding and therefore pretty much all digital audio uses perceptual coding and every mix of every piece of commercial audio has EQ, mic placement, compression or a number of other factors/processes applied to improve perception. In other words, by your definition, everything uses perceptual coding. So, please answer these two questions: A. What is the point of the term "perceptual coding" if it means the same as "everything" and B. What new/different term should we use for actual perceptual coding?

You're following the exact same path you always follow. You make some incorrect claim/assertion of fact which is refuted and then instead of holding your hand up and admitting it or just dropping it, you defend it to the death for post after post, make more and more ridiculous assertions to support your position, even to the point of contradicting yourself, and dig a deeper and deeper hole for yourself until eventually you back yourself into a logical cul de sac. At which point you usually post something along the lines that you feel like a useless human being. My last question of this post is therefore: Why follow that same path, why put yourself through all this time after time?

G


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## SoundAndMotion

Wow. You missed both my points. I must have been unclear.



pinnahertz said:


> It's how Wikipedia works: Anyone...even you!....could write an article.  And anyone...even me!....could edit it and correct it.  It's the entire principle behind the site.


I know how Wikipedia works. I was not trying to elevate Wikipedia articles. I was pointing out that forum posts (like yours.... and mine!!!) are also written by _anyone_. The big difference is that I can't edit yours and vice-versa. So you or I or anyone can say whatever we want (subject to TOS) and it can't be edited/corrected (other than mods)



pinnahertz said:


> Not bad.  And by that definition, µ-law is not perceptual coding because it does not specificaly base it's action on the likelihood of a listener hearing what's been removed.


Hmmm, µ-Law *is* perceptual coding because it *does* _take advantage of limitations of human perception_ (logarithmic scaling (Weber-Fechner)) and it does _selectively remove data based on how unlikely it is that a listener will *notice* the removal_ (in the form of speech intelligibility). "Notice" not "hear", and notice is usage dependent.


----------



## gregorio

SoundAndMotion said:


> Hmmm, µ-Law *is* perceptual coding because it *does* _take advantage of limitations of human perception_ (logarithmic scaling (Weber-Fechner)) and it does _selectively remove data based on how unlikely it is that a listener will *notice* the removal_ (in the form of speech intelligibility). "Notice" not "hear", and notice is usage dependent.



You're missing the point. Using the definition of "taking advantage of limitations of human perception" covers everything! Music takes advantage of the limitations of human perception; the perception of relationships between note pitches (Harmony and melody) and/or the perception of relationships between the timing of events (rhythm and beats), so just about all music, recorded or not is, by this definition, "perceptual coding" and in fact, so is all commercial audio (TV, film, radio) because in addition; EQ, compression and various other effects are always employed to take advantage of the limitation of human perception. In practice, the term "perceptual coding" specifically means the use of a perceptual model of frequency and timing masking (auditory masking) used as a reference for removing data. It's therefore a term which only applies to certain lossy audio codecs rather than being a meaningless term because it applies to everything.

G


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## SoundAndMotion (Jul 10, 2018)

gregorio said:


> You're missing the point. Using the definition of "taking advantage of limitations of human perception" covers everything! Music takes advantage of the limitations of human perception;


No. I shouldn’t have abbreviated the quote. Perceptual Coding is lossy compression that takes advantage of limitations in human perception. In perceptual coding, audio data is selectively removed (compressed) based on how unlikely it is that a listener will notice the removal.
Music is not lossy compression.



gregorio said:


> In practice, the term "perceptual coding" specifically means the use of a perceptual model of frequency and timing masking (auditory masking) used as a reference for removing data. It's therefore a term which only applies to certain lossy audio codecs rather than being a meaningless term because it applies to everything.


That is true for modern codecs, but it is not a general rule/definition.



gregorio said:


> 1. Maybe the difficulty you were/are having is that you were trying to separate the issue into two different "domains" (amplitude and frequency)? While it's sometimes useful to do this for the sake of "vizualization" in reality they are not separate/different domains, they're exactly the same thing. A sine wave (for example) is effectively defined as: An increasing amplitude until a "peak" is reached, then a decreasing amplitude until the "trough" is reached and then an increasing amplitude again until the starting point is reached. We call this a "cycle" and frequency is simply the number of cycles per second. In other words, Frequency = Amplitude (over time).


In your exchange with ILoveMusic, there are several ideas that are unclear, misleading, garbled or incorrect. Not all of that from you(!) in the above quote(!!), but rather in the entire exchange.
If you’re so inclined, perhaps you can comment on the following facts:

1. Time domain (amplitude vs. time)  and frequency domain (amplitude or power vs. frequency) are indeed _different_ domains. The _same_ information is represented in _different_ forms. An example of the difference is it allows one to use a point-by-point product in one domain instead of convolution in the other. The analog signal from a microphone, the signal on an analog interconnect, the output of a DAC or the signal on the speaker wire from an analog amplifier are all voltage amplitude vs. time (time domain). Frequency information is not available unless one transforms the signal, using a spectrum analyzer for analog data or a Fourier transform for digital data. The fact that the unit of frequency is Hertz, equivalent to cycles/second, and the word second implies you have time is meaningless.
2. 16 bit in not perfect. If I have an original signal, convert it to 16 bit resolution, and use that to create a reproduced signal, the original and reproduced will not be identical. That is, subtracting the two does not give all zeroes. If the intention is to say that the imperfection is not audible, that is different from saying it is perfect. Yes, I know BigShot is itching to say “in the context of this forum, inaudible is perfect”, but “16bit is already perfect there is no "better" than perfect” is misleading.
3. 1 bit delta-sigma coding (used in DSD and SACD) is not the same as 1 bit LPCM coding. Usually, talking about 16 bits implies linear pulse code modulation.
4. Shannon-Nyquist tells us we need to sample at _greater than_ twice the highest frequency of interest, *not* _greater than or equal to_ twice. Twice the highest frequency is inadequate.


----------



## bigshot

For the purposes of listening to recorded music in the home with human ears....


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## SoundAndMotion (Jul 10, 2018)

bigshot said:


> For the purposes of listening to recorded music in the home with human ears....


Right on cue...
Thanks! 
That is the context you mean. I think in the context of Shannon-Nyquist, "perfect" has a different meaning 
Saying "the imperfections of 16-bit are inaudible" is different from saying "16-bits is perfect".


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## TheSonicTruth (Jul 10, 2018)

SoundAndMotion said:


> Right on cue...
> Thanks!
> That is the context you mean. I think in the context of Shannon-Nyquist, "perfect" has a different meaning
> Saying "the imperfections of 16-bit are inaudible" is different from saying "16-bits is perfect".



That is why, since the early 1990s I guess, production has moved from 16/44.1/-96 to 24, 32-float/44.1, 88.2, 96, 192/-144 to -infinity dynamic range, instead of remaining at the same specs as Redbook, as it did through most of the eighties.

Every time processing is done, in the digital domain, quantization occurs, and thus dither must be applied.  Producing(recording, mixing, mastering), and delivering all at 16/44.1/-96 could result in an effective 6bit/44.1/-36 PUDDING as delivered on CD!


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## ILikeMusic (Jul 10, 2018)

Help me out again here. Is this just another argument about whether Redbook specs are adequate to serve the limits of human hearing, or are there ways in which a digitally reproduced signal is non-perfect with respect to the input, even in an absolute sense? (assuming you remain within bandwidth and dynamic range limitations.)


----------



## SoundAndMotion

ILikeMusic said:


> In what ways is a digitally reproduced signal non-perfect with respect to the input, even in an absolute sense? (assuming you remain within bandwidth and dynamic range limitations.)


If you remain within the bandwidth limits, the resolution of a 16-bit digital signal has a _limit_ on its dynamic range, that the original signal _might not have_. Keep in mind that many here can and should point out that if each stage of your analog electronic chain before you digitize doesn't have better than 16-bit resolution, the noise from your analog chain will be in the dynamic range of the digital signal. Many will also like to challenge the audibility of whatever 16-bits can't capture.


ILikeMusic said:


> Or is this just another argument about whether Redbook specs are adequate to serve the limits of human hearing?


No, it is an argument about whether 16-bits can be used to "perfectly" reconstruct the signal. That is, whether there is _*no*_ difference between original and reproduced. Audibility is not part of that statement.


----------



## ILikeMusic (Jul 10, 2018)

Well, ignoring for the moment that any errors beyond audibility are irrelevant in the real world, are you saying that in musical performances it's possible (meaning in any reasonable and non-pedantic sense) to exceed a 96 dB dynamic range?


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## SoundAndMotion (Jul 10, 2018)

ILikeMusic said:


> So in musical performances it's possible (meaning in any reasonable and non-pedantic sense) to exceed a 96 dB dynamic range?


Is a live acoustic performance, yes and no. In one that you might attend, probably no. If you allow deafeningly loud peaks, like real cannons around the audience for the 1812 Overture, perhaps yes. If the whole performance is in an anechoic chamber, and you hold your breath, possibly yes.
If it is amplified, possibly yes.
In a studio, it is possible, but...
I doubt it happens often, if at all.

_(I have a can of worms here labelled, "Most people don't understand how to calculate the dynamic range of human hearing". I hope no one opens it, because the short answer is "it is silly to try to find a single number for it")_


----------



## ILikeMusic

Not to mention the minute likelihood of any system actually being capable of reproducing such a recording (or at least for the sake of your hearing, you hope.)


----------



## bigshot

A dynamic range in music that broad would be uncomfortable to listen to under any conditions.


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## bigshot (Jul 10, 2018)

SoundAndMotion said:


> I think in the context of Shannon-Nyquist, "perfect" has a different meaning
> Saying "the imperfections of 16-bit are inaudible" is different from saying "16-bits is perfect".



No that isn't different. Nyquist says that waveforms are perfectly reconstructed within the bandwidth of the sampling rate. It doesn't claim to be perfect beyond the range of its sampling rate. If you want super audible frequencies to be reproduced perfectly, you just up the sampling rate.


----------



## TheSonicTruth

bigshot said:


> No that isn't different. Nyquist says that waveforms are perfectly reconstructed within the bandwidth of the sampling rate. It doesn't claim to be perfect beyond the range of its sampling rate. If you want super audible frequencies to be reproduced perfectly, you just up the sampling rate.




But bit-depth(8, 16, 24, etc) _does_ determine the amount of quantization error/noise.  So even if conditions(_within the bandwidth of the sampling rate_) are met, there will always be 'noise'.  48dB down from full-scale for 8bit, 96dB down from full-scale for 16bit, and so on.


----------



## SoundAndMotion

bigshot said:


> No that isn't different. Nyquist says that waveforms are perfectly reconstructed within the bandwidth of the sampling rate. It doesn't claim to be perfect beyond the range of its sampling rate. If you want super audible frequencies to be reproduced perfectly, you just up the sampling rate.


It _is_ different because of 16 bits, as SonicTruth said, _not_ because of bandwidth/sampling rate issues.


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## SoundAndMotion

bigshot said:


> A dynamic range in music that broad would be uncomfortable to listen to under any conditions.


96dB dynamic range? Under any conditions? You're simply wrong. Upon what do you base your statement?


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## TheSonicTruth (Jul 10, 2018)

SoundAndMotion said:


> 96dB dynamic range? Under any conditions? You're simply wrong. Upon what do you base your statement?




I *think* he meant a RECORDED piece of music with that amount of dynamic range.

96dB total dynamic range is like reserve horsepower - you'll likely only ever use or experience, in a recording, only 4 to 40dB of that.  Or like a 100W per channel stereo amp: You may never use more than 30W per channel of it, but thank 'God it's there!'(Rick Harrison, LifeLock)  lol!


----------



## 71 dB

gregorio said:


> 1. That's great, it means there's no need whatsoever for dither because quantisation noise is apparently masked. All these decades of dither and dither development wasted ... do you think we should tell someone?
> 
> 2. So now, in effect, you're contradicting yourself because µ-law/A-law are analogue processes!
> 2a. By your definition, wax cylinders are perceptual coding, vinyl with RIAA are perceptual coding, cassettes with bias are perceptual coding, CDs with emphasis are perceptual encoding, despite your claim in point #1, dither is perceptual coding and therefore pretty much all digital audio uses perceptual coding and every mix of every piece of commercial audio has EQ, mic placement, compression or a number of other factors/processes applied to improve perception. In other words, by your definition, everything uses perceptual coding. So, please answer these two questions: A. What is the point of the term "perceptual coding" if it means the same as "everything" and B. What new/different term should we use for actual perceptual coding?
> ...


1. When have I said the quantization noise in µ-law is inaudible? It is partly masked by the signal, of course! Large dynamic range is more important than quantization noise with speech intelligibility. Why do you guys try SO HARD to find errors in what I say? Even using these silly strawmen.

2. Doesn't matter. The end result is 8 bit digital audio.

2a. CDs with emphasis are kind of perceptual.

Whatever!!!!!!!!


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## 71 dB

If µ-law is not perceptual coding, then WHAT IS IT? Somebody better correct Wikipedia!


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## 71 dB

I'm fed up fighting here! What is the harm of calling µ-law early perceptual coding? WHAT IS THE FUCVKINFG HARM!!!!????=


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## 71 dB

I AMSO ANGEYHG


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## 71 dB

I AM SO FRUSTRATED!!!! ANGRYT!!! DAMN FORUM!!! DAMN BOYS!! MEN!! STUPID!!


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## bfreedma

71 dB said:


> I AM SO FRUSTRATED!!!! ANGRYT!!! DAMN FORUM!!! DAMN BOYS!! MEN!! STUPID!!



Can you please dial down the drama.


----------



## bigshot




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## 71 dB

bfreedma said:


> Can you please dial down the drama.



Already feeling better so yes, I can. I'm done with µ-law.


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## gregorio

SoundAndMotion said:


> [1] In perceptual coding, audio data is selectively removed (compressed) based on how unlikely it is that a listener will notice the removal. ...That [definition] is true for modern codecs, but it is not a general rule/definition.
> [2] In your exchange with ILoveMusic, there are several ideas that are unclear, misleading, garbled or incorrect. Not all of that from you(!) in the above quote(!!), but rather in the entire exchange.
> If you’re so inclined, perhaps you can comment on the following facts:



1. That definition doesn't work though. For example, reducing a 24bit signal down to 16bit with the application of noise-shaped dither would also qualify as "perceptual coding" and therefore most CDs produced in the last 20 years or so are "perceptual coding". In modern usage, the term "perceptual coding" is used specifically as I stated and does not include redbook CDs.

2. I made it clear a number of times that I was not attempting to be absolutely accurate, instead, I was looking for a view or "visualization" which would help him understand some of the fundamental basics. For example, in that same post I stated "_BTW, all the above is not exactly correct or incorrect either! It's just another way of looking at the issue, a way which avoids some incorrect conclusions/assumptions._" with this in mind, I'll address your points:


SoundAndMotion said:


> 1. ... The analog signal from a microphone, the signal on an analog interconnect, the output of a DAC or the signal on the speaker wire from an analog amplifier are all voltage amplitude vs. time (time domain). Frequency information is not available unless one transforms the signal, using a spectrum analyzer for analog data or a Fourier transform for digital data.
> 2. 16 bit in not perfect. [2a] If I have an original signal, convert it to 16 bit resolution, and use that to create a reproduced signal, the original and reproduced will not be identical. That is, subtracting the two does not give all zeroes.
> 3. 1 bit delta-sigma coding (used in DSD and SACD) is not the same as 1 bit LPCM coding. Usually, talking about 16 bits implies linear pulse code modulation.
> 4. Shannon-Nyquist tells us we need to sample at _greater than_ twice the highest frequency of interest, *not* _greater than or equal to_ twice. Twice the highest frequency is inadequate.


1. That was exactly my point. All that is actually measured/stored in the digital domain is amplitude vs. time. Frequency is derived from that "amplitude vs time" information, so effectively there is only one domain "amplitude vs time".
2. 16bit is perfect, any bit depth it's perfect and to suggest otherwise requires disproving the Nyquist/Shannon Theorem. However ..
2a. Two points: Firstly, if you convert your original signal to 16bits and then reproduce that signal you are going to need an Analogue to Digital Converter and a Digital to Analogue Converter, both of which, by definition, are not just digital devices but analogue devices as well. If you compare an input signal to an output signal you are therefore not only evaluating 16bit digital but 16bit plus two analogue stages, neither of which can be absolutely perfect. Secondly and in addition, while 16bit is perfect and Nyquist/Shannon and the Sampling theory are true/correct, that does NOT mean that the theory is perfectly implemented in ADCs and DACs, there is of course variations of implementations between different makes and models. In fact, some audiophile DACs not only didn't implement the theory perfectly, they deliberately broke the theory; Filter-less NOS Dacs for example. If we're going to be precise about it, no DAC implements the theory absolutely perfectly, a perfect Sinc function is impractical for example. In practise, even very cheap implementations can get surprisingly close to perfect.
3. Both (DSD and PCM) follow the same basic rules of digital audio theory. If we're going to get into the fine detail though, then the situation is rather complex and the lines between delta-sigma and PCM become blurred. For example, for 25 years or more all professional PCM ADCs (as far as I'm aware) actually digitize the input signal using a form of delta-sigma encoding, typically a handful of bits with sample rates in the many megahertz range and then decimate down to the user defined PCM sample rate and bit depth.
4. Again, yes if we're going to be precise about it but for the sake of discussion and a simple "view", 2x sample rate is acceptable. In effect, the difference we're talking about is a Nyquist Point for CD (44.1kS/s) of 22.04kHz instead of 22.05kHz.

G


----------



## bigshot

We were discussing Nyquist and reproducing upper harmonic frequencies. That's sampling rate, not bit rate.

Assuming the normal room tone of a room being somewhere above 30dB, in order for a 96dB dynamic range to be audible above the room tone all the way down to -96dB, you would need to be into the volume range range pushing past the threshold of pain. I sure don't listen to music that way,

Redbook is already overkill.


----------



## gregorio

TheSonicTruth said:


> [1] That is why, since the early 1990s I guess, production has moved from 16/44.1/-96 to 24, 32-float/44.1, 88.2, 96, 192/-144 to -infinity dynamic range, instead of remaining at the same specs as Redbook, as it did through most of the eighties.
> [2] Every time processing is done, in the digital domain, quantization occurs, and thus dither must be applied.



1. Not really. In the '80's the vast majority of production (mixing) was still analogue, even though the recording was often digital. In the 1990's recording progressed from 16bit to 24bit, not because 24bit had more accuracy/sound quality than 16bit but simply because it provided far more headroom, making multi-channel recordings much easier.

2. No, that's not correct. Yes, every time processing is applied in the digital domain quantisation error occurs but dither does NOT need to be applied. In the '90's, commercial digital mix environments were usually 48bit fixed or 32 float. Many hundreds of processing steps would be required for the cumulative quantisation error to get even near audibility and therefore dither is not typically required. Today (and for quite a few years), commercial mix environments are virtually all 64bit float and the quantisation error is so tiny, you'd need to sum together thousands of processing steps to even get within the theoretical limits of 24bit!



SoundAndMotion said:


> [1] But bit-depth(8, 16, 24, etc) _does_ determine the amount of quantization error/noise. So even if conditions(_within the bandwidth of the sampling rate_) are met, there will always be 'noise'. 48dB down from full-scale for 8bit, 96dB down from full-scale for 16bit, and so on.
> [2] 96dB dynamic range? Under any conditions? You're simply wrong. Upon what do you base your statement?



1. Mmmm, that's not exactly correct. If it were correct, 1bit (SACD) would be un-listenable with only 6dB of dynamic range. Of course, SACD is very listenable, the noise is effectively a great deal further down than -6dB, due to noise-shaped dither. Therefore:

2. Even in most top class commercial recording studios a 96dB dynamic range is typically more than uncomfortable. With a studio noise floor of say 20dB, a 96dB range above that would put the peaks at 116dB which is not far off the threshold of pain, let alone "comfortable". However "96dB dynamic range" is rather arbitrary, referring to the point above, the dynamic range of CD/16bit is not 96dB, it's effectively about 120dB, due to noise-shaped dither. Can you think of any conditions where a dynamic range of 120dB would be comfortable? In practice, virtually all commercial audio is kept within a 60dB dynamic range, as more than that would be uncomfortable for the majority of consumers.

G


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## TheSonicTruth (Jul 10, 2018)

gregorio said:


> 1. Not really. In the '80's the vast majority of production (mixing) was still analogue, even though the recording was often digital. In the 1990's recording progressed from 16bit to 24bit, not because 24bit had more accuracy/sound quality than 16bit but simply because it provided far more headroom, making multi-channel recordings much easier.
> 
> 2. No, that's not correct. Yes, every time processing is applied in the digital domain quantisation error occurs but dither does NOT need to be applied. In the '90's, commercial digital mix environments were usually 48bit fixed or 32 float. Many hundreds of processing steps would be required for the cumulative quantisation error to get even near audibility and therefore dither is not typically required. Today (and for quite a few years), commercial mix environments are virtually all 64bit float and the quantisation error is so tiny, you'd need to sum together thousands of processing steps to even get within the theoretical limits of 24bit!
> 
> ...



Man you just love saying "No", "Not correct".  Maybe I just got the timelines a little wrong??

Well, the information I posted, that you claim is "not correct", is being retransmitted all over the web, on my Facebook pages and blogs.  And guess what Gregorio: I'm getting people to change both how they buy music and what they listen to it on!

They, like I, are tired of having our chains jerked.


----------



## castleofargh

most of us understand that there isn't much of anything perfect going on anywhere from recording down to playback output. I understand how some find weird that @gregorio  would insist on some perfect aspects of digital audio so often while it's obvious that he himself recognizes the limits of a practical application as he just did. but here is why I believe he does that and is right to do so(maybe I'm wrong, he'll tell me^_^). to me it's about an important aspect of digital audio that fails to get across when we stop at visualizing the signal as sampled dots on a graph over time. digital audio is more than that, it's supported by wave properties.
for example, we know that we don't have the ability to create missing data out of nothing. we can add stuff but it won't be the previously lost data. if I delete half of my post(real delete, so no CTRL+Z), there is no simple math trick to reconstruct it or part of it. with that in mind, the intuitive idea of a quantification to N bits, could be that all the data below the least significant bit is lost. be it the quieter signals, or the extra amplitude accuracy for all the music(same thing). I know how easy it is to think that way. but that intuitive idea is false, and it's easy to demonstrate. based on what we know about data, we shouldn't be able to retrieve information below the LSB if that information was truly lost/cut out/discarded. and yet we do just that all the time with noise shaping. we have defined a lower limit with the bit depth encoding, and then we go retrieve information the least significant bit of that code shouldn't be able to quantify. 
that's why even though it can seem like a small distinction in most conversations, or even as zeal in favor of digital audio,  the mental model where the signal is perfectly captured, plus some noise, is the better simplified model. that way, when we move the noise around and the original signal appears in the audible frequency range, it actually makes sense instead of feeling like it's witchcraft. and again DSD is the living extreme example of that fact. we encode a 1bit signal, yet we can use that to reconstruct stuff with better than 100dB of dynamic/SNR in the audible frequency range. and adding even more samples allows, after some fooling around, to retrieve even lower levels. showing that the bit depth wasn't in the first place putting any strict limit of how accurate the amplitude of the captured signal can be. the signal was really there all along plus some noise. 


a counter example to that would be to filter out a 60khz signal and encode at 44.1khz. no matter how many bits we'd use to record, we wouldn't reconstruct that 60khz tone. the data this time has effectively been discarded. which agrees with the sampling theorem. 44.1khz sampling isn't able to completely capture a 60khz signal. and the band limiting makes very sure of that. like my half deleted post, this is a concrete examples of what happens with effectively lost data. it's pretty much lost. ^_^

so IMO all this is making a strong case in favor of looking at quantization noise, as noise over a fully captured signal within the allowed frequency range. we can't really capture anything perfectly or reconstruct perfectly in the real world, but there is meaning in avoiding the assumption that we're just dealing with quantization errors below which the signal is forever lost/discarded.



does this make sense, or did I just enter the twilight zone in my own head?


----------



## bigshot

Even without dithering, there is more than enough oomph in redbook to reproduce recorded music perfectly for the purposes of listening to music in the home. I think there is a test of that in one of the videos in my sig. How many times does the lily need to be gilded?


----------



## jgazal

pinnahertz said:


> (...) Anyone...even you!....could write an article. (...)





castleofargh said:


> most of us understand that there isn't much of anything perfect going on anywhere from recording down to playback output.
> (...)
> does this make sense, or did I just enter the twilight zone in my own head?



I identify with "writing articles" from the "twilight zone in my own head"!


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## SoundAndMotion (Jul 11, 2018)

gregorio said:


> That definition doesn't work though. For example, reducing a 24bit signal down to 16bit with the application of noise-shaped dither would also qualify as "perceptual coding" and therefore most CDs produced in the last 20 years or so are "perceptual coding". In modern usage, the term "perceptual coding" is used specifically as I stated and does not include redbook CDs.


Well, when I found it here, I thought it caught the gist of the idea, but you have indeed found a loophole. But your loophole does not only use the "perceptual" (noise-shaped dither) part of the definition, your argument includes the idea that RBCDs use "lossy compression". It it were 2 in the morning in the dorms, the pizza just arrived, and we just reloaded the bong or grabbed a six-pack, I'd be up for arguing for hours whether down-sampling and truncation to create CDs constitutes "lossy compression". But I moved out of the dorms 40 years ago and haven't "smoked" in nearly as long, so I'm not interested in this idea.



gregorio said:


> I made it clear a number of times that I was not attempting to be absolutely accurate, instead, I was looking for a view or "visualization" which would help him understand some of the fundamental basics. For example, in that same post I stated "BTW, all the above is not exactly correct or incorrect either! It's just another way of looking at the issue, a way which avoids some incorrect conclusions/assumptions."


Cool, that's a neat trick. Indeed, I didn't read that part. If you put that in your sig, you never have to worry about being wrong again. You just say "Dude, read my sig". Sweet.
As for helping "him understand some of the fundamental basics", how can you explain the need for anti-aliasing filters, the idea of frequency folding, the need for anti-imaging (reconstruction) filters, even really fundamental stuff like bandwidth and frequency response without the frequency domain?



gregorio said:


> 1. That was exactly my point. All that is actually measured/stored in the digital domain is amplitude vs. time.
> 1a.Frequency is derived from that "amplitude vs time" information, so effectively there is only one domain "amplitude vs time".


1a. I don't even get how you can make this mistake. Freq. and time domains are 2 domains. There are complete textbooks on freq. domain analysis, that don't work in the time domain, and vice versa. I mentioned convolution vs. point-by-point multiply... they are different.
1. Yay! I'm so glad you have this brand new perspective that corrects so very many of your previous posts, e.g.


gregorio said:


> Since the dawn of audio recording right up to the present day we measure/convert and store just two properties, frequency and amplitude.


 among so many dozens more over the years. We don't measure/convert frequency. We measure/convert amplitude as a function of the independent variable, time.



gregorio said:


> 2. 16bit is perfect, any bit depth it's perfect and to suggest otherwise requires disproving the Nyquist/Shannon Theorem. However ..
> 2a. Two points: Firstly, if you convert your original signal to 16bits and then reproduce that signal you are going to need an Analogue to Digital Converter and a Digital to Analogue Converter, both of which, by definition, are not just digital devices but analogue devices as well. If you compare an input signal to an output signal you are therefore not only evaluating 16bit digital but 16bit plus two analogue stages, neither of which can be absolutely perfect.


2. No, BigShot is right. Shannon-Nyquist deals with sample rate and bandwidth, not bit depth. (@bigshot-I never claimed Nyquist deals with bit depth). You have mistaken the complete lack of bit depth in Shannon-Nyquist to mean bit depth is irrelevant. That is not what it means. Sampling theory deals with periodically taking samples... but garbage-in, garbage-out... the quality (resolution) of the sample determines the quality of the reconstruction. Perfect reconstruction requires perfect samples. I think BigShot may want to add: "But good enough is good enough". Perhaps a dictionary is needed to agree on the meaning of "perfect"! ...?
2a. When talking about the math and theory of Shannon-Nyquist, we are talking analytic, not noisy world. IF the original signal is analog (which I didn't say), then talking about analog electronics can be "straight wire with gain". I was actually thinking of a double- or extended-precision float signal. Convert it to 16-bit integer (scaling properly, of course), and then convert back to the same floats, dither, filter, scale as needed, and the signal will be different. The difference is 5 or 6 orders of magnitude down, but different. If you do the same with 24-bit, the difference will more than 7 orders of magnitude down.
But you say any bit depth is perfect... 1 or 2 bit LPCM will really suck...

Need a break, will answer the rest of your post later.


----------



## gregorio (Jul 11, 2018)

castleofargh said:


> I understand how some find weird that @gregorio would insist on some perfect aspects of digital audio so often while it's obvious that he himself recognizes the limits of a practical application as he just did. but here is why I believe he does that and is right to do so (maybe I'm wrong, he'll tell me^_^) ...



You're not wrong at all. In fact you've done a very good job of explaining why "I do that". I would like to expand it with a couple of points, which admittedly might add confusion rather than reduce it:

1. Shannon stated: "_If a function x(t) contains no frequencies higher than B hertz, it is* completely determined* by giving its ordinates at a series of points spaced 1/(2B) seconds apart._" (emphasis is mine) - We have to remember that this statement isn't just an assumption, opinion, idea or suggestion, Shannon proved it mathematically. Furthermore, there's been no hint in the 70 years since he published his proof that it's in anyway incorrect, in fact quite the opposite, his proof forms the basis for Information Theory and therefore of all digital technology. Hence why Shannon is often described as "the father of the digital age". In short, Shannon's statement is logically* incontestable*, effectively it's an unquestionable absolute truth/fact. It's vital to fully appreciate this, Shannon's statement does not require any additional conditions in order to be true, it is true, period!! For example, note that it does not mention bit depth, that's because it does not need to, Shannon's statement is correct as it stands, period (IE. Regardless of bit depth). In effect, all "bit depth" does is define the amount of noise which accompanies our "complete determination" but the "complete determination" is always there and to suggest otherwise effectively means disproving what Shannon has already proven.

2. In addition to the previous point, another reason for "visualising" the issue as: "A signal that is perfectly captured (completely determined) plus some noise", is that this "noise" is a unique case. This noise is generated by/the result of a mathematical process, so within that process this noise can be manipulated independently from the signal, it can be "Shaped". All other noise cannot be "shaped", it's already part of the signal (for example, noise floor of the recording venue or thermal noise from the mic's and other analogue equipment in the recording chain) and cannot be manipulated independently.



TheSonicTruth said:


> [1] Man you just love saying "No", "Not correct". [1a] Maybe I just got the timelines a little wrong??
> [2] Well, the information I posted, that you claim is "not correct", is being retransmitted all over the web, on my Facebook pages and blogs.
> [2a] And guess what Gregorio: I'm getting people to change both how they buy music and what they listen to it on!
> [3] They, like I, are tired of having our chains jerked.



1. That is NOT true! I do not love saying "no" or "not correct" and you obviously don't like being told you are incorrect/wrong, so why don't you do us BOTH a favour and not post incorrect information/assertions in the first place??
1a. No, it wasn't just the timeline, it was your whole "guess". Except in a tiny number of special cases, production could not "_remaining at the same specs as Redbook, as it did through most of the eighties_" because production has never been "at the same specs as redbook", not during the '80's or any other time. Production was analogue and when it switched to digital it was at higher than 16bit.

2. Good choice, Facebook probably is the best place for your information, due to it's reputation for disseminating "Fake News".
2a. Great! BTW, when you've got say about twenty million people "to change" be sure to let the major record labels know!

3. I don't like having my chains jerked either but for me, "fake news" IS having my chains jerked. I realise though that many people seem to prefer "fake news" to the actual facts, presumably because it doesn't take much effort or intellectual ability. So I hope "they, like you" are very happy together.

G


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## SoundAndMotion (Jul 11, 2018)

Aha, no response to my post... understood. Being embarrassed is not conducive to answering. No point for me to continue on that post...



gregorio said:


> In short, Shannon's statement is logically* incontestable*, effectively it's an unquestionable absolute truth/fact. It's vital to fully appreciate this, Shannon's statement does not require any additional conditions in order to be true, it is true, period!! For example, note that it does not mention bit depth, that's because it does not need to, Shannon's statement is correct as it stands, period (IE. Regardless of bit depth). In effect, all "bit depth" does is define the amount of noise which accompanies our "complete determination" but the "complete determination" is always there and to suggest otherwise effectively means disproving what Shannon has already proven.



Not sure, to whom that's directed. I'm not contesting Shannon; I'm contesting your understanding of several issues related to digital audio, including Shannon (my underline in your text). "Complete determination" plus a bunch of noise destroys the "complete determination", so it's wrong. It is not perfect.
Should we make it interesting?
-I will make a signal with a bandwidth less than 20kHz and duration of 1 second.
-You tell me exactly how you want me to prepare it, consistent with a simple test of your contention that bit depth doesn't alter the ability for "complete determination". Anything you want, ending with me sending you a file that is encoded at 4-bits and sampled at 44.1kHz. I will document each step with an intermediate, modified data file.
-You recreate my signal, showing a "complete determination" of my signal. If some of the important features of my signal are buried in noise missing, that is not a success.
-We post the results here.
-If you do succeed... what should we do? Have a bet? You succeed, I send you money; you don't succeed, you send me money? Or we send to a charity of the winner's choice? Or? How much? 10, 100, 1000? Euros, dollars, pounds?

I'm confident. Are you?

Edit: strikethrough buried in noise; replaced with "missing"


----------



## 71 dB

SoundAndMotion said:


> 2. No, BigShot is right. Shannon-Nyquist deals with sample rate and bandwidth, not bit depth. (@bigshot-I never claimed Nyquist deals with bit depth). You have mistaken the complete lack of bit depth in Shannon-Nyquist to mean bit depth is irrelevant. That is not what it means. Sampling theory deals with periodically taking samples... but garbage-in, garbage-out... the quality (resolution) of the sample determines the quality of the reconstruction. Perfect reconstruction requires perfect samples. I think BigShot may want to add: "But good enough is good enough". Perhaps a dictionary is needed to agree on the meaning of "perfect"! ...?.



Output = original signal + noise. How do you define signal and noise? Original signal is what you put in (input signal) and noise is everything else. So, by definition all inaccuracies of the output IS noise while the signal part is perfect. That's why bit depth doesn't make the signal any less perfect. Lower bit depth just increases inaccuracies => increases noise.

Original signal is bandlimited => Shannon-Nyquist applies to it
Noise is bandlimited => Shannon-Nyquist applies to it
Output signal = original signal + noise is bamdlimited => Shannon-Nyquist applies to it

Output signal is such a bandlimited signal that is able to be fitted into the bit depth while being as close to the original signal as possible. So in the quantization process noise was added to the input signal so that the sum becomes output signal and can be stored to the bit depth available. The signal is perfect bandlimited itself all the time. Only noise is added to it. When the bit depth is large enough, this noise is below the threshold of hearing and only signal is heard, and that signal is perfect (in theory, in practice ADCs and DACs are not perfect and can introduce audible errors, especially during the early years of digital audio).


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## RRod (Jul 11, 2018)

Doesn't seem to me that greg is trying to say that Q(x) = Q(x + n), if Q is a quantifier and n is noise we add intentionally. I read this as "x + n still has x in it, and we can choose n so that we can hear x audibly perfect after quantization."

*edit: @71 dB was on it


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## 71 dB

SoundAndMotion said:


> -I will make a signal with a bandwidth less than 20kHz and duration of 1 second.
> -You tell me exactly how you want me to prepare it, consistent with a simple test of your contention that bit depth doesn't alter the ability for "complete determination". Anything you want, ending with me sending you a file that is encoded at 4-bits and sampled at 44.1kHz. I will document each step with an intermediate, modified data file.
> -You recreate my signal, showing a "complete determination" of my signal. If some of the important features of my signal are buried in noise missing, that is not a success.
> -We post the results here.
> ...


At 4-bit there is about 24 dB of dynamic range, so of course all the stuff below that is likely to be lost in the noise. So, what are these important features? However, it's perfect signal + really loud noise.


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## gregorio (Jul 11, 2018)

SoundAndMotion said:


> [1] But your loophole does not only use the "perceptual" (noise-shaped dither) part of the definition, your argument includes the idea that RBCDs use "lossy compression".
> [2] Cool, that's a neat trick. [2a] Indeed, I didn't read that part. [2b] If you put that in your sig, you never have to worry about being wrong again.



1. My argument includes the idea of selectively removing data, namely the last 8 LSBs, not specifically that RBCDs use "lossy compression". With a noise-shaped RBCD the data we're removing only represents unwanted noise, with lossy compression the data being removed doesn't only represent unwanted noise, it represents actual musical signals (which are masked according to a perceptual model). And BTW, that was just one loophole off the top of my head, I can think of others! 

2. Yes, it is but I can't take any credit for it. Creating an over-simplified "visualisation" (which isn't precisely accurate) to help someone understand some basic concept has been around for millennia, probably since humans first developed complex language skills.
2a. Or apparently several other parts/posts, where I made it abundantly clear I was talking in very simplified terms. My initial response (here) to this topic of the thread spelt it out in some detail.
2b. That is an idea but I haven't done it because I'm not always posting "over-simplified visualisations". Instead, I do it the other way around, I try to make it clear when I am posting in over-simplified terms. However, some level of simplification is usually necessary, because this is a forum of mostly laypeople and if I used accurate pro audio engineering terminology (as I would with other professional engineers) some/many would have trouble understanding/following.



SoundAndMotion said:


> 1a. I don't even get how you can make this mistake.
> 1. Yay! I'm so glad you have this brand new perspective that corrects so very many of your previous posts, e.g.
> among so many dozens more over the years. We don't measure/convert frequency. We measure/convert amplitude as a function of the independent variable, time. ...
> [2] As for helping "him understand some of the fundamental basics", how can you explain the need for anti-aliasing filters, the idea of frequency folding, the need for anti-imaging (reconstruction) filters, even really fundamental stuff like bandwidth and frequency response without the frequency domain?



1a. What mistake, quoting you? "_The analog signal from a microphone, the signal on an analog interconnect, the output of a DAC or the signal on the speaker wire from an analog amplifier are all voltage amplitude vs. time (time domain). Frequency information is not available unless one transforms the signal, using a spectrum analyzer for analog data or a Fourier transform for digital data._" - Are you now saying it's not "all voltage amplitude vs time"?
1. So you're saying that "amplitude vs time" does not contain frequency information? Presumably you're not saying that, otherwise the second part of your quote is incorrect. Sound waves can be defined by "Frequency + Amplitude" or "Amplitude vs Time" (and separated into the Frequency and Time domains as audio engineering text books do) both are interchangeable and valid, Frequency, by definition, contains time and "Amplitude vs Time" contains frequency. Looking at it one way can help with some explanations and looking at it the other way can help with other explanations. In other words, I do not "have this brand new perspective", I have various different perspectives of the same thing and none of them are new to me!

2. Again, that's the point, I was not trying to explain bandwidth, frequency response, anti alias/imaging, decimation or any other type of filter, just amplitude/bits.

Consistently in this line of "discussion" you are failing to account for the context of my responses or even that they are in fact responses!



SoundAndMotion said:


> 2. No, BigShot is right. Shannon-Nyquist deals with sample rate and bandwidth, not bit depth.
> 2a. I was actually thinking of a double- or extended-precision float signal. Convert it to 16-bit integer (scaling properly, of course), and then convert back to the same floats, dither, filter, scale as needed, and the signal will be different.
> [3] But you say any bit depth is perfect... 1 or 2 bit LPCM will really suck...



2. I believe I dealt with this in my additional point #1 in my previous message (response to castleofargh).
2a. Of course it will be different, as castleofargh explained, if you're going to remove a bunch of bits, the information in those bits are gone for good. Converting 16bit fixed back to 32 or 64bit float isn't ever going to give us that exact same data back. We can, with noise-shaped dither, expose more of the "completely determined" signal (beneath the noise floor imposed by 16bit with TDPF dither) so that in practical application the 16bit conversion sounds exactly the same as the 32/64 bit float original but the data will not be exactly the same (because we're adding noise-shaped dither).

3. It IS effectively perfect but with just one or two bits it would "really suck" because you'd barely be able to hear any of that perfect signal buried in the huge amount of quantisation error/noise. Of course, if you applied noise-shaped dither during the quantisation process then more of our perfect ("completely determined") signal would be exposed, as the quantisation noise is redistributed away from our range of hearing. This is the reason why SACD does not "really suck"!

G


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## SoundAndMotion

71 dB said:


> At 4-bit there is about 24 dB of dynamic range,


 More if noise shaping and dither are used, which I assume @gregorio would request.


71 dB said:


> So, what are these important features?


Some clue about my signal... any evidence that it has been perfectly reproduced... see below.


71 dB said:


> However, it's perfect signal + really loud noise.


That sounds like doing evoked potential recording in the brain, which I've done. You do signal averaging. Your stimulus produces a tiny signal deeply buried in a lot of noise, so you repeat the process many times. We recorded from the brain stem, with surface (skin) electrodes, and got pretty good results with 1000 repetitions. The noise averages lower and lower and the tiny signal pops out. The larger the noise compared to the signal, the more repetitions required.


----------



## gregorio

SoundAndMotion said:


> I'm not contesting Shannon ... "Complete determination" plus a bunch of noise destroys the "complete determination", so it's wrong. It is not perfect.



This is contradictory! If "complete determination" is "wrong" or "not perfect", then YOU ARE contesting Shannon! Shannon's statement DOES NOT require some addendum or qualifications to be true, it is proven true as it stands.



SoundAndMotion said:


> You tell me exactly how you want me to prepare it, consistent with a simple test of your contention that bit depth doesn't alter the ability for "complete determination". Anything you want, ending with me sending you a file that is encoded at 4-bits and sampled at 44.1kHz. I will document each step with an intermediate, modified data file.



You don't appear to understand what noise-shaped dither is!

How you would need to prepare the signal would be: An appropriate noise-shaped dither applied at the point of quantisation, to redistribute the quantisation noise/error outside the range of human hearing (thereby revealing the "complete determination"). "Appropriate noise-shaped dither" would mean an algorithm designed with a suitable amount of "shaping" aggressiveness specifically for 4bit quantisation and a sample rate which allows a large enough audio bandwidth to accommodate the redistribution of such a large amount of noise. With SACD we obviously have an even larger amount of noise to redistribute but with a sample rate of 2.8mHz there's plenty of (inaudible) audio bandwidth within which to redistribute that noise. Even so, Sony required SACD players to implement an analogue LPF (@ 50kHz) post reconstruction, to remove most of that HF noise energy, as it was easily enough to cause IMD in downstream equipment.

G


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## SoundAndMotion (Jul 11, 2018)

I really apologize, but I've grown very tired of this. Perhaps some sleep and some coffee will help... another time.
Some quickies:


gregorio said:


> 1. So you're saying that "amplitude vs time" does not contain frequency information?


I'm saying "amplitude vs. time" is signal information in the time domain and "amplitude(+phase) vs. frequency" is the same signal information in the frequency domain. You must do a mathematical transform to go back and forth. In a digital audio file, e.g. .wav, _there is no coding of the frequency content of the signal_, just amplitude vs. time. Saying frequency domain is the same as time domain is like saying flour and water IS bread. It's not, unless you transform it. Look at this data:




Give me some idea about the frequencies**. How many are there? It's a small number. What are the frequencies? They are simple, round numbers. If I sent you this file, you could easily TRANSFORM it and answer. Otherwise, to answer your question above, yes, there is no frequency information in the graph, WITHOUT A TRANSFORM of the data.


gregorio said:


> It IS effectively perfect but with just one or two bits it would "really suck" because you'd barely be able to hear any of that perfect signal buried in the huge amount of quantisation error/noise. Of course, if you applied noise-shaped dither during the quantisation process then more of our perfect ("completely determined") signal would be exposed, as the quantisation noise is redistributed away from our range of hearing. This is the reason why SACD does not "really suck"!


I'm getting tired and bored with your "it's perfect, but with imperfections" arguments. SACD and DSD are 1-bit "delta-sigma"!!!! There are more than 2 output values, there are 2 delta values and many sigma values. It is also sampled at a dramatically higher sample rate. It is not the same as 1 or 2 bit LPCM.

** If you want to cheat, I'll tell you exactly how:
See if NIH Image still exists, or use your own image processing program.
Read in the image and kill everything not blue.
Do an AND with a 1-pixel wide black and white vertical stripe pattern.
Get the program (NIH Image will) to spit out the coordinates of each point.
Stick the data in MATLAB, LabVIEW, Mathematica, Maple, Octave... heck even MS Excel.
Do a Fourier TRANSFORM, careful with the time scale!
Post the answer.


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## SoundAndMotion (Jul 11, 2018)

gregorio said:


> You don't appear to understand what noise-shaped dither is!


Not an experienced expert, but I understand it pretty well. I would use this algorithm:


> algorithm noise_shape {
> /*Input:
> b_orig, the original bit depth
> b_new, the new bit depth to which samples are to be quantized
> ...


If you have a better one, that's fine. I can upsample, apply TPDF (not TDPF), do noise shaping... whatever you want. BUT, in accordance with sampling theory, I'd only send 4-bit 44.1kHz, and I'd expect to see my signal in some form, not pure noise.

If you want to use the signal averaging I mentioned above. I could rerandomize the TPDF and send multiple files, to help you pull out my signal.


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## gregorio (Jul 11, 2018)

SoundAndMotion said:


> Not an experienced expert, but I understand it pretty well. ... BUT, in accordance with sampling theory, I'd only send 4-bit 44.1kHz, and I'd expect to see my signal in some form, not pure noise.



Again, you are contradicting yourself. If you "understand it pretty well", then you would realise that "in accordance with sampling theory" 4bit 44.1kS/s (or did you really mean 44.1kHz? See, two can play at that game!) is insufficient. If you want to use such a few number of bits, the quantisation noise obscuring the perfect (completely determined) signal is so great you need to extend the sample rate to provide extended audio bandwidth to redistribute that noise. I'm not sure how much use this will be as you're apparently trying to disprove even the very foundation of sampling theory (the Nyquist-Shannon Theorem) but just in case, you should look up the Gerzon-Craven Noise-Shaping Theory. As I haven't the time to go into detail, a good starting place would be the introductory Lipshitz and Vanderkooy extract "Pulse-Code Modulation - An Overview" - particularly sections two and three, which deal with dither/quantisation and noise-shaping respectively.

G

BTW thanks, amplitude vs time contains no information in respect of frequency and bread contains no flour or water, got it!


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## SoundAndMotion (Jul 11, 2018)

gregorio said:


> Again, you are contradicting yourself.



Karl Rove's Tactic #3? (Maybe you should switch to politics)
Let's see...


gregorio said:


> 1. Shannon stated: "If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart."
> ... (snip)
> Shannon's statement does not require any additional conditions in order to be true, it is true, period!! For example, note that it does not mention bit depth, that's because it does not need to, *Shannon's statement is correct as it stands*, period (IE. *Regardless of bit depth*).





gregorio said:


> *If you want to use such a few number of bits*, the quantisation noise obscuring the perfect (completely determined) signal is so great *you need to extend the sample rate* to provide extended audio bandwidth to redistribute that noise.


(My bold) Hmmm... You two should get your story straight.


gregorio said:


> If you "understand it pretty well", then you would realise that "in accordance with sampling theory" 4bit 44.1kS/s (or did you really mean 44.1kHz? See, two can play at that game!) is insufficient.


Yes, I did really mean 44.1kHz sample rate, like RBCD. That gives 44,100 samples per second, so 44.1kS/s. Saying it's insufficient contradicts your previous quote about Shannon. I'm tired of the game you two are playing.


gregorio said:


> BTW thanks, amplitude vs time contains no information in respect of frequency and bread contains no flour or water, got it!


Actually, you didn't "got it". I think all caps threw you off... "TRANSFORM" is the same as "transform". Now go back and read it and you'll get what I said, and you can stop misquoting me.

Gregorio, you seem pretty smart and a lot of the stuff you write probably helps a lot of people. But you have to understand the limits of your knowledge, acknowledge and correct when you are mistaken (don't change your view and act as though it was always so), and quit saying so often that your take is "the truth and incontrovertible".
We all make mistakes.
But our exchange on the last few pages is going nowhere.


----------



## bigshot

SoundAndMotion said:


> But our exchange on the last few pages is going nowhere.



You only JUST noticed that?! Why do you ask questions you don't want answers to?


----------



## SoundAndMotion (Jul 12, 2018)

bigshot said:


> Why do you ask questions you don't want answers to?


I do want the answers, but given the responses I got, I don't expect them.


----------



## SoundAndMotion

bigshot said:


> You only JUST noticed that?!


No...
If you READ the posts, you'll see my rising level of frustration with the responses. Been going on a while...

Anything meaningful to contribute?


----------



## rule42

Maybe someone could define a bit clearer as to what they mean by frequency domain. I get what Gregorio says about a simple relation between time and frequency but my understanding of signal processing in general seems to think that the frequency domain is defined as the set of data after a Fourier Transform is done, that involves complex numbers, negative frequencies and a whole bunch of maths way over my head, I think most image manipulation routines work on data in frequency domain like this then do an inverse with the signal still as was. Don't know so much about audio but have assumed a lot of the filters and the like work the same way.


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## SoundAndMotion (Jul 12, 2018)

rule42 said:


> ..my understanding of signal processing in general seems to think that the* frequency domain is defined as the set of data after a Fourier Transform is done*, that involves complex numbers, negative frequencies and a whole bunch of maths way over my head,


my bold

Yep, that's basically it. In the context of audio, you start with a data series of amplitude vs. time (time domain), do a FT and you do get complex values with negative frequencies, but you can fairly easily make it useful to you.

You discard the last (N/2 - 1) points (negative frequencies, N is number of points) and convert the complex values to polar form (most numerical packages have a simple command). If your time domain data is real (not complex, and in this case it is), then the negative frequencies duplicate all but 2 of the positive frequencies (well the complex conjugates, and counting zero as positive).

Then for each frequency, except the first (zero) and last (fs/2)(fs is the sampling frequency), you get the magnitude (modulus of complex value) and phase (argument of complex value). The first and last don't have phase. You must scale all values based on the number of data points.
You can plot this (it's called a Bode plot, and there are 2 graphs) and see the magnitude vs. frequency (commonly called spectrum or frequency response) and the phase vs. frequency. *This is the frequency domain.
*
It is a lossless process (other than numerical roundoff errors of your computer) and can be reversed.
Freq. domain data --> Inverse Fourier transform --> Time domain data (if you set up the data correctly)

A lot of analysis and manipulation (like filtering) can be done in either domain, but yes filtering is computationally and conceptually easier in the frequency domain.
Does this help, or was I too _complex_  ?


EDIT: many numerical packages do everything I just said for you and go straight from time domain data to frequency domain Bode plots! 
EDIT 2: Audacity will give you a spectrum plot from time domain data.


----------



## gregorio

SoundAndMotion said:


> [1] You two should get your story straight.
> [1a] Yes, I did really mean 44.1kHz sample rate, like RBCD. That gives 44,100 samples per second, so 44.1kS/s. Saying it's insufficient contradicts your previous quote about Shannon.
> [2] I'm tired of the game you two are playing.
> [3] We all make mistakes. ... [3a] But you have to understand the limits of your knowledge, acknowledge and correct when you are mistaken ... [3b] and quit saying so often that your take is "the truth and incontrovertible".
> [3c] But our exchange on the last few pages is going nowhere.



[1] Who "two"? Me and Claude Shannon, Nyquist and Shannon? Don't forget Stephen Hawking and the there's Wittaker, Kotelnikov and others who contributed or discovered the theorem independently. As for getting our "story straight": You are confusing YOUR inability to understand the story with the story not being straight. The "story" is (again!): The signal IS "completely determined" plus there is ALWAYS some amount of noise; in theory zero noise would require an infinite number of bits and in practise even with an infinite number of bits there is still some amount of noise, due to the laws of physics (Thermal noise for example)! The real issue then is one of how much fidelity: How much of that "completely determined" signal do we want to recover or conversely, how little noise do we want obscuring that "complete determination"? You want to recover a significant amount of the "complete determination" AND use a bit depth which introduces a great deal of quantisation noise. Digital Audio/Sampling Theory allows for this scenario, because quantisation noise is effectively separate from the "complete determination" (perfect signal). The (band limited) "complete determination" can therefore be recovered/exposed (to any arbitrary level above thermal noise limits), by moving/redistributing the quantisation noise away from the frequency band occupied by the "complete determination" (as defined by the Gerzon-Craven Noise-Shaping Theorem). It should be obvious therefore, that you need to provide enough audio bandwidth to accommodate both the band of frequencies occupied by the "completely determined" signal + the band of frequencies occupied by the (redistributed) quantisation noise. The large amount of noise introduced by 4bit quantisation would need to be redistributed over a large band of frequencies, the vast majority of which would need to be in the ultrasonic range and therefore a far higher sample rate would be required to accommodate that audio frequency band.

Why don't you read the EVIDENCE ALREADY PRESENTED? Specifically, page 13 of the Lipsh*tz-Vanderkooy extract linked previously where two examples are given, an example with just 1 bit and another with 8 bits. The latter for instance demonstrates the perfect ("completely artefact free") recovery of the "complete determination" of any signal within an audio band of 0-20kHz with a SNR of 120.4dB using just 8bits, plus a sample rate of 176.4kS/s to redistribute the quantisation/dither noise (to above 20kHz).

1a. Saying it is insufficient does NOT contradict my previous quote of Shannon, it ENTIRELY agrees with it!

2. The game us two are playing is called Science, if you're "tired" and want to play a different game you're in the wrong forum!

3. By "we all" do you mean everyone except you?
3a. But you don't have to?
3b. No, YOU quit saying it's "your take", it is NOT my "take", it's Shannon's "take", I'm quoting him exactly and directly, with nothing added or taken away.
3c. I did not say (Shannon's "take") was incontrovertible, I said it was "logically incontrovertible". The reason our exchange is going nowhere is because you are effectively illogically controverting it! You've presented no evidence al all, only an example which you don't believe is possible but which the evidence I've presented (Lipsh*tz-Vanderkooy extract and the Gerzon-Craven Theorem) indicates it's entirely possible, plus of course, SACD actually demonstrates it!

G


----------



## gregorio

rule42 said:


> I get what Gregorio says about a simple relation between time and frequency but my understanding of signal processing in general seems to think that the frequency domain is defined as the set of data after a Fourier Transform is done, that involves complex numbers, negative frequencies and a whole bunch of maths way over my head



Yep, it is pretty simple. Here is a graph of amplitude vs time:


 
As frequency is just cycles per second (CPS or more commonly Hz), then "amplitude vs time" inherently contains the information for the frequency. In the above example, we can follow the amplitude over time and see that a "cycle" completes in 1 millisecond, it therefore has a frequency of 1,000 cycles per second (or 1kHz). We could have done it the other way around; not given you the time (not labelled the x-axis) but informed you it was a 1kHz signal and then you could have followed the amplitude to where the cycle completes and inserted the time yourself (1 millisecond).

From the above we can see that "frequency" and "amplitude vs time" are interchangeable. However, in real life we're rarely dealing with a single frequency, we're usually dealing with a whole bunch of simultaneous frequencies. All those frequencies combine (sum) to form a single "complex waveform", rather than the simple sine wave above. All the frequency information is still there but it needs to be calculated, split out into it's constituent frequencies. This calculation is done by means of a "Fourier Transform" (FT) and the result is called the "Fourier Series", after this calculation (FT) we are in the "Frequency Domain". Below we see a complex wave form but after a Fourier Transform we see that it's actually still relatively simple, just two frequencies of different amplitudes summed together (500Hz and 800Hz). 



You seem to already know/understand the above but it might be useful for others.

G


----------



## rule42

@SoundAndMotion  and @gregorio . Thank you both for the replies. All nice and clear. No, about the right level of complexity for now before I delve into it a bit deeper. I like trying to follow examples through as a way of learning, so do you think that Octave would be the best free tool for that? I've not used Matlab but have some familiarity with Python from over 20 years ago (including Numpy and a little Scipy) and they don't seem too dissimilar. Or stick with Python of course.


----------



## bigshot

Frequency is dependent on time. You don't have one without the other. That is what sampling rate is all about. That should be obvious.


----------



## rule42

Yes, of course. I'm just trying to fit how the time and frequency domains work together in harmony. That should be obvious.


----------



## SoundAndMotion

rule42 said:


> @SoundAndMotion  and @gregorio . Thank you both for the replies. All nice and clear. No, about the right level of complexity for now before I delve into it a bit deeper. I like trying to follow examples through as a way of learning, so do you think that Octave would be the best free tool for that? I've not used Matlab but have some familiarity with Python from over 20 years ago (including Numpy and a little Scipy) and they don't seem too dissimilar. Or stick with Python of course.


MATLAB is extremely powerful, with a matching price tag. It is updated with new features every 6 months. It includes Simulink, which is peerless for general purpose modeling/simulation. There are some fairly recent, very nice GUI features, but it is primarily command-line. I currently use MATLAB/Simulink and LabVIEW.

Octave is a free MATLAB clone. It is a few years behind the latest version of MATLAB and has nothing like Simulink. There are minor differences from MATLAB, but with very little effort, you can use m-file scripts (for MATLAB). But it is extremely powerful and free. I used Octave a few years back; it is a great, powerful substitute for MATLAB!!

I've never used Python, other than tiny projects with my son. A few years back, a colleague tried to convince me to switch. With the scientific packages (NumPy, SciPy, Matplotlib, IPython and pandas), he said it was at least as powerful as MATLAB, and free. There has been a recent push to use free software for publicly-funded research. So R for stats and (scientific-)Python for experiment control and general purpose analysis. You can get it for nearly anything: obviously WinPC, Mac, Linux, but even iPad and RaspberryPi.

I strongly support and encourage learning this stuff by taking different types of signals and seeing what happens when you ... "play". As an in-depth guide you can use this nice book, but the example code is in BASIC. There are also books, where the examples are MATLAB (works for Octave too) or Python, or PM me if you want some quick tips or direction on particularly enlightening exercises. ...lots of help online...


----------



## yo2tup2

Good discussion


----------



## rule42

Great info that @SoundAndMotion . I'll download and have a play with Octave to see what I can do with it over the next few days. Looks great free software and will no doubt do way more than I want, which is 'play' as you describe, plus there seem to be loads of tutorial/examples online for it or Matlab ones that seem basically the same. Might PM you for pointers to nice examples in a few days time. That book you link is great too (well the first few chapters are nice and gentle so far). Many thanks.


----------



## SoundAndMotion

gregorio said:


> [1] Who "two"? Me and Claude Shannon, Nyquist and Shannon? Don't forget Stephen Hawking and the there's Wittaker, Kotelnikov and others who contributed or discovered the theorem independently. As for getting our "story straight": You are confusing YOUR inability to understand the story with the story not being straight. The "story" is (again!): The signal IS "completely determined" plus there is ALWAYS some amount of noise; in theory zero noise would require an infinite number of bits and in practise even with an infinite number of bits there is still some amount of noise, due to the laws of physics (Thermal noise for example)! The real issue then is one of how much fidelity: How much of that "completely determined" signal do we want to recover or conversely, how little noise do we want obscuring that "complete determination"? You want to recover a significant amount of the "complete determination" AND use a bit depth which introduces a great deal of quantisation noise. Digital Audio/Sampling Theory allows for this scenario, because quantisation noise is effectively separate from the "complete determination" (perfect signal). The (band limited) "complete determination" can therefore be recovered/exposed (to any arbitrary level above thermal noise limits), by moving/redistributing the quantisation noise away from the frequency band occupied by the "complete determination" (as defined by the Gerzon-Craven Noise-Shaping Theorem). It should be obvious therefore, that you need to provide enough audio bandwidth to accommodate both the band of frequencies occupied by the "completely determined" signal + the band of frequencies occupied by the (redistributed) quantisation noise. The large amount of noise introduced by 4bit quantisation would need to be redistributed over a large band of frequencies, the vast majority of which would need to be in the ultrasonic range and therefore a far higher sample rate would be required to accommodate that audio frequency band.
> 
> Why don't you read the EVIDENCE ALREADY PRESENTED? Specifically, page 13 of the Lipsh*tz-Vanderkooy extract linked previously where two examples are given, an example with just 1 bit and another with 8 bits. The latter for instance demonstrates the perfect ("completely artefact free") recovery of the "complete determination" of any signal within an audio band of 0-20kHz with a SNR of 120.4dB using just 8bits, plus a sample rate of 176.4kS/s to redistribute the quantisation/dither noise (to above 20kHz).
> 
> ...


(1 and 1a). Gregorio, you deserve a calm, thorough, civil (no insults) response to this part and some previous posts. The back and forth sniping of "yes it is" - "no it isn't" helps no one and reflects badly on both of us, IMHO. Rather than relying on "my take" or "your take" (see below), I'll try to mostly quote directly from published sources. I've read most of [Shannon, C. E. (1949). Communication in the presence of noise. _Proceedings of the IRE, 37(1)_, 10-21.] and I've skimmed [Shannon, C. E. (2001). A mathematical theory of communication. _ACM SIGMOBILE mobile computing and communications review, 5(1)_, 3-55.] and [Nyquist, H. (1928). Certain topics in telegraph transmission theory. _Transactions of the American Institute of Electrical Engineers, 47(2)_, 617-644.]
Interestingly, I was originally planning to skip your "reading assignment" [Lipshitz, S. P., & Vanderkooy, J. (2004). Pulse-Code Modulation--An Overview. _Journal of the Audio Engineering Society, 52(3)_, 200-215.], but I was curious how Gerzon-Craven differed from the algorithm I posted. I read it and it turns out *this is a really nice review. Thanks!* Most of my response to you will come from this. It answers pretty much everything. I'll write the calm, thorough, civil response when I finish reading and have a bit more time.

2. If I tell you I have GarageBand on my Mac, and my son and I have played with some of the noises he made with his e-guitar, and therefore I can lecture you on best practices and normal procedure in recording studios (your field), would you get angry, laugh, shake your head... ? I laughed.

3. FYI, "we all" means me, you and the other 7+ billion.

3a. Of course I do. But I'm human and imperfect, and although I admit I'm wrong more than anyone I know (not because it's frequent, but because I find it important), I don't always do so. I should though.

3b. Shannon died in 2001. The only source for "his take" is his own well-written words. When you tell me "what he meant", either directly related to his words, or worse "what he meant" from what he left out, that is "your take".

3c Again, I don't take issue with the Shannon quote w.r.t our discussion; I don't agree with your take regarding bit-depth.

BTW, after carefully reading Lipshitz & Vanderkooy, I realize I should not offer to dither or noise-shape, since those were not known and not included at the time Shannon, Nyquist, Whittaker et al. created the sampling theory stuff. Their work survives without dither/noise-shaping. But I'll dither (TPDF) and noise shape with my algorithm or any you provide anyway, if you want. (algorithm, not JAES theoretical paper; my membership in AES lapsed about 2 years ago). My signal will easily survive 44.1kHz sampling rate, as per Shannon et al., but won't survive 4-bit bit-depth.

LOL, I thought you were shy about L*i*p*s*h*1*t*z's name, but I see the "bad part" is automatically deleted.

Peace


----------



## 71 dB

SoundAndMotion said:


> LOL, I thought you were shy about L*i*p*s*h*1*t*z's name, but I see the "bad part" is automatically deleted.



This automatic deletion of the 4 letter words starting with f and s is so stupid. 
People are _attacked_ online all the time. If you are triggered seeing 4-letter words then better stay offline, becuase the internet is full of s**t.


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## SoundAndMotion (Jul 15, 2018)

@gregorio , is this you?

I never realized how hard your work is  Good job! 
(Have fun!)


LATE EDIT: My wife said this _can_ be seen as a jibe. That is _NOT_ the intent. It is an attempt to reduce tension, and a lighthearted way to say I respect your competence in your field... with perhaps a hope for reciprocation.


----------



## SoundAndMotion

rule42 said:


> @SoundAndMotion
> Many thanks.


You're welcome.
Good luck, but more important, have fun!!


----------



## TheSonicTruth

SoundAndMotion said:


> (1 and 1a). Gregorio, you deserve a calm, thorough, civil (no insults) response to this part and some previous posts. The back and forth sniping of "yes it is" - "no it isn't" helps no one and reflects badly on both of us, IMHO. Rather than relying on "my take" or "your take" (see below), I'll try to mostly quote directly from published sources. I've read most of [Shannon, C. E. (1949). Communication in the presence of noise. _Proceedings of the IRE, 37(1)_, 10-21.] and I've skimmed [Shannon, C. E. (2001). A mathematical theory of communication. _ACM SIGMOBILE mobile computing and communications review, 5(1)_, 3-55.] and [Nyquist, H. (1928). Certain topics in telegraph transmission theory. _Transactions of the American Institute of Electrical Engineers, 47(2)_, 617-644.]
> Interestingly, I was originally planning to skip your "reading assignment" [Lip****z, S. P., & Vanderkooy, J. (2004). Pulse-Code Modulation--An Overview. _Journal of the Audio Engineering Society, 52(3)_, 200-215.], but I was curious how Gerzon-Craven differed from the algorithm I posted. I read it and it turns out *this is a really nice review. Thanks!* Most of my response to you will come from this. It answers pretty much everything. I'll write the calm, thorough, civil response when I finish reading and have a bit more time.
> 
> 2. If I tell you I have GarageBand on my Mac, and my son and I have played with some of the noises he made with his e-guitar, and therefore I can lecture you on best practices and normal procedure in recording studios (your field), would you get angry, laugh, shake your head... ? I laughed.
> ...



Lipschitz.  

There, should duck the censor algorithms.


----------



## TheSonicTruth

SoundAndMotion said:


> @gregorio , is this you?
> 
> I never realized how hard your work is  Good job!
> (Have fun!)




Roseanne's cousin?


----------



## old tech

Hey @gregorio 

Mark Aldrep has (unconvincingly) critiqued your OP of this thread in his reply to one of his reader's comments.

http://www.realhd-audio.com/?p=6234


----------



## pinnahertz

old tech said:


> Hey @gregorio
> 
> Mark Aldrep has (unconvincingly) critiqued your OP of this thread in his reply to one of his reader's comments.
> 
> http://www.realhd-audio.com/?p=6234


Waldrep's entire justification for his work in HD audio is contained in this sentence from that thread, "I prefer to match the real world and let listeners strive to play it back — without compromise."  He states this without reference to exactly what the "real world' actually is, but also freely admits elsewhere that he's not sure what matters, or even what about HD is audible, so he just strives to capture as much as possible.  If you read more of his writings you find he freely acknowledges that both 24 bits and high sampling rates above 96kHz do not have much justification, but since verification and validation are so darn difficult, he opts for the "safe" route.  Yet, it's all tempered by his statement, "However, it is true that recordings and systems with high-resolution, 24-bit capability are very rare. The run of the mill 24-bit downloads you get online don’t need 24-bits."  And that's true even of his recordings.

Obviously, capturing "more" than you think you'll ever need is not necessarily a bad thing, but insisting on a verifiable and repeatable audible improvement because of it is without much substance.  Yet, Waldrep's company is also perhaps the only one that faithfully supplies real HD audio throughout the entire production chain, from capture to release, including the use of microphones with actual response above 20kHz.  So he does have some of the best test material.  What I find is a far bigger factor in the quality of his recordings is they are just well done (mostly), and are still well done regardless of actual resolution.


----------



## gregorio

SoundAndMotion said:


> [1] The only source for "his take" is his own well-written words. When you tell me "what he meant", either directly related to his words, or worse "what he meant" from what he left out, that is "your take".
> 3c Again, I don't take issue with the Shannon quote w.r.t our discussion; I don't agree with your take regarding bit-depth.
> [2] I realize I should not offer to dither or noise-shape, since those were not known and not included at the time Shannon, Nyquist, Whittaker et al. created the sampling theory stuff. [2a] Their work survives without dither/noise-shaping.
> [3] My signal will easily survive 44.1kHz sampling rate, as per Shannon et al., but won't survive 4-bit bit-depth.



1. I am NOT telling you what he meant, I am quoting him directly.
3c. There is no "take" on bit depth, Shannon's statement is proven true, as it stands!

2. Shannon, Nyquist et al., did NOT create sampling theory stuff, they created the Sampling Theorem upon which Sampling Theory is based. 
2a. Yes it does, because it is the foundation of Sampling Theory. However, their work was not the end of the development of Sampling Theory.

3. Yes it will, otherwise you are saying Shannon was wrong. Let's put the issue the other way around, let's say Shannon was somehow wrong, that his statement/proof is incorrect and in fact a signal is not "completely determined" with a more than x2 sampling rate, it also requires a certain bit depth (an infinite bit depth in theory but we'll ignore that for now) and therefore with 4 bits Shannon's Sampling Theorem fails; the signal is not "completely determined" and "won't survive 4-bit bit-depth". How then does Noise-Shaped Dither work and why does it even exist? How can noise-shaped dither reveal the signal down to some arbitrary level if the signal has not "survived" (is not "completely determined")?



old tech said:


> Mark Aldrep has (unconvincingly) critiqued your OP of this thread in his reply to one of his reader's comments.



I agree with @pinnahertz but I'd go a step further, systems such as he (Waldrep) are NOT "very rare", they're non-existent! He also makes some other beginner mistakes or more likely some other omissions, for example he consistently ignores noise-shaped dither, which give 16bit in practise a dynamic range of 120dB. This makes a nonsense of his statement "A larger dynamic range — matching the capabilities of human hearing without compromise — DOES MEAN HIGHER QUALITY!" - because a dynamic range of 120dB significantly EXCEEDS "the capabilities of human hearing"! Also, what in the "real world" of music requires 120dB dynamic range?

Mark Waldrep does make some excellent recordings, he's also generally a good source of accurate information and has eloquently debunked all manner of audiophile myths regarding audio formats, with the exception of the audio format he sells! As pinnahertz also stated, even though the benefits of Hi-Rez are inaudible, at least he actually makes/sells what he is purporting to sell!

G


----------



## TheTrace

Almost 10 years of arguments guys, come on we're almost there! 

Like a big happy family.


----------



## TheSonicTruth

TheTrace said:


> Almost 10 years of arguments guys, come on we're almost there!
> 
> Like a big happy family.



And even the 'experts' aren't perfect!


----------



## old tech

The discussion around bit depth and sampling rates has been quite interesting.  Being a somewhat novice, there is one conceptual aspect of the noise shaping which I don’t understand too well.

I follow the argument that 8bits and 4bits or even 1bit can fully reveal the analog signal, but with more noise.  I also follow the logic that if we wanted say 4bits to have noise levels beneath the human hearing threshold (ie the same as 24bits), we would need a higher sampling rate than 44.1 to move the noise into the ultrasonic range.

What I don’t quite get is how then noise shaping works with 16/44.  Firstly, I don’t get why noise shaping is required with 16bits when its noise floor is already around -96db.  Secondly, where does the noise shaped signal go if the band width is limited to 22.05khz?


----------



## TheSonicTruth

old tech said:


> The discussion around bit depth and sampling rates has been quite interesting.  Being a somewhat novice, there is one conceptual aspect of the noise shaping which I don’t understand too well.
> 
> I follow the argument that 8bits and 4bits or even 1bit can fully reveal the analog signal, but with more noise.  I also follow the logic that if we wanted say 4bits to have noise levels beneath the human hearing threshold (ie the same as 24bits), we would need a higher sampling rate than 44.1 to move the noise into the ultrasonic range.
> 
> What I don’t quite get is how then noise shaping works with 16/44.  Firstly, I don’t get why noise shaping is required with 16bits when its noise floor is already around -96db.  Secondly, where does the noise shaped signal go if the band width is limited to 22.05khz?





old tech said:


> The discussion around bit depth and sampling rates has been quite interesting.  Being a somewhat novice, there is one conceptual aspect of the noise shaping which I don’t understand too well.
> 
> I follow the argument that 8bits and 4bits or even 1bit can fully reveal the analog signal, but with more noise.  I also follow the logic that if we wanted say 4bits to have noise levels beneath the human hearing threshold (ie the same as 24bits), we would need a higher sampling rate than 44.1 to move the noise into the ultrasonic range.
> 
> What I don’t quite get is how then noise shaping works with 16/44.  Firstly, I don’t get why noise shaping is required with 16bits when its noise floor is already around -96db.  Secondly, where does the noise shaped signal go if the band width is limited to 22.05khz?




Scroll ahead to 8:42 in this video, where he gets into dithering:


----------



## old tech

I understand dithering and the points made by Monty.  The question I have is more around noise shaped dither, particularly moving the energy into the higher frequencies when the band width is limited to 22.05 khz.

I appreciate that even so, being limited to 22khz that our hearing is less sensitive at higher frequencies doesn't that the extra energy have an effect?  If not, why would noise shapes 8bits require a higher bandwidth than 22khz?  More fundamentally, why is noise shaping beneficial at all for 16bits?


----------



## TheSonicTruth (Jul 17, 2018)

old tech said:


> I understand dithering and the points made by Monty.  The question I have is more around noise shaped dither, particularly moving the energy into the higher frequencies when the band width is limited to 22.05 khz.
> 
> I appreciate that even so, being limited to 22khz that our hearing is less sensitive at higher frequencies doesn't that the extra energy have an effect?  If not, why would noise shapes 8bits require a higher bandwidth than 22khz?  More fundamentally, why is noise shaping beneficial at all for 16bits?



14:09 in that video.  And listen to what he said at 17:13!


----------



## gregorio (Jul 18, 2018)

old tech said:


> What I don’t quite get is how then noise shaping works with 16/44. [1] Firstly, I don’t get why noise shaping is required with 16bits when its noise floor is already around -96db. [2] Secondly, where does the noise shaped signal go if the band width is limited to 22.05khz?
> [3] ... why would noise shapes 8bits require a higher bandwidth than 22khz?



1. TBH, in most cases it isn't. If I explain it in a little more detail perhaps that would help: With 16bit, the noise floor with (standard, TPDF) dither is about -92dB, as dither typically uses 1 LSB and 16 bit un-dithered would actually have a noise floor of -98.08dB (16 x 6.02dB + 1.76dB). The vast majority of music has a dynamic range of around 48dB or less. Popular/Non-acoustic genres will hit near 0dB numerous times, overall be relatively loud and require a relatively low output level on playback. The dither noise floor at -92dB is going to be 100 times or more below the noise floor of the recording and therefore, even at loud playback volumes the dither noise floor is going to be completely inaudible. Even with classical and jazz, a -92dB noise floor is going to be completely inaudible in the vast majority of cases. However, there are a potential set of circumstances where it *could* be audible. For example, the 1812 Overture (Tchaikovsky) has cannons near the end and it could be that they produce transient peaks say 18dB above any other peak in the rest of the overture. If you wanted to playback such a recording so that the rest of the overture (excluding the cannons) sounded roughly the same volume as other classical recordings, then you'd have to increase your playback level by say 18dB and our dither noise floor would therefore be 18dB higher (effectively at -74dBFS). Assuming you have a high quality playback system (capable of +18dBSPL louder than normal), normally listen quite loudly and have a listening environment with a low noise floor, then potentially the dither noise floor could become audible. The 1812 overture is an obvious example but there are other examples which are not so obvious. For example, a hard hit on an orchestral bass drum produces a large amount of energy. It's not obvious because much of that energy is below 50Hz, where our hearing is insensitive and therefore it doesn't sound particularly loud but we could have peaks up to as much as about 12dB higher than normal. All the above requires a quite extreme set of circumstances and only applies to a tiny number of recordings, because most recordings with such unusual peaks would have those peaks reduced (compressed/limited), so the recording is suitable for consumers with good equipment/listening environments rather than only for those with excellent equipment/environments. Having said all this, it's been standard mastering practise to apply noise-shaped dither to all 16bit releases for the last 20+ years, as it only takes about a minute to apply and then you're covered, regardless of ANY music and playback scenario.

2. That's not entirely fixed, it depends on the noise-shaping algorithm and there's sometimes some user (mastering engineer) adjustment available. In general though, the shaped dither noise starts ramping up from around 10kHz and is at it's peak by about 17kHz, this deliberately coincides with human hearing; we're most sensitive at around 3kHz and have a roll-off in sensitivity starting around 5-7kHz and a steeper roll-off around 12-14kHz. With noise-shaping we don't get less dither noise, we get the same amount (or typically slightly more), so as far as RMS dither noise is concerned we've got a dither noise floor of say -90dB, but that noise is outside the range of hearing sensitivity, giving us a perceived noise floor around -120dB. This graph of noise-shaping might help you visualise the situation:






The X-Axis is frequency and the Y-Axis represents relative dB. So with 16bit the "0dB" line represents about -96dB and the fairly flat blue (ID=99) line covering it represents standard (TPDF) dither. The other "IDs" represent different user selectable noise-shaping algorithms. You'll notice a couple of things: A. We're not actually loosing any noise energy, just redistributing it. As we reduce it from one area of freqs, we must increase it elsewhere so we end up with the same amount of RMS noise energy (exactly how this works is laid out in the Gerzon-Craven Noise-Shaping Theorem). B. That from about 600Hz upwards the shaping curve is roughly an inverse of the Fletcher-Munson equal loudness curves. For example, ID=16 (the strongest noise-shaping) gives us about -26dB less noise at 3kHz, IE. A noise floor of about -122dB with 16bit (-96dB - 26dB) but by around 17kHz we've got about 30dB more noise, a noise floor of about -66dB (-96dB + 30dB) However, assuming perfect hearing, our sensitivity is down by about 50-60dB at 17kHz and down by over 100dB at 20kHz (where our redistributed noise peaks at around +36dB). Additionally, our sensitivity rolls-off in the lower freqs starting from around 800Hz. Therefore, as far as human hearing is concerned, the noise-shaped dither noise floor of ID=16 would never sound higher than about -122dB.

3. Using my explanation and graph from point #2, let's substitute 16bit with 8bit. Our ID=99 (0dB) line now represents -48dB. ID=16 therefore represents a perceptual noise floor of -74dB (26dB lower than -48dB), while peak noise (at around 20kHz) would be at about -12dB (-48dB + about 36dB @ 20khz). However, let's say for illustration purposes that we want a perceptual noise floor of say -92dB (roughly the same as standard dithered 16bit). First of all, we're going to need another algorithm, one that is 18dB more aggressive than ID=16, so that at peak hearing sensitivity (about 3kHz) it is removing 44dB of noise instead of about 26dB (-48dB - 44dB = -92dB). Unfortunately though, this means that the peak noise level (at about 20kHz) is likewise going to be about 18dB higher than ID=16: -48dB + 36dB + 18dB = +6dBFS, which is impossible. The solution would be to increase the sample rate, say double it to 88.2kS/s. With ID=16 the highest redistributed noise levels cover a 5kHz freq band (17kHz to 22kHz). With a sample rate of 88.2kS/s we could spread that same amount of redistributed noise energy over a much larger frequency band, a 27kHz band (5kHz + the additional 22.05kHz) and thereby significantly lower it's level. ..... From all this, I hope you can see that the lower the noise floor we wish to achieve the more dither noise therefore has to be redistributed and in addition, the fewer bits we have to play with, the higher the dither noise we've got to start with. Hence why SACD, with just one bit plus a desired noise floor of about -120dB, needs a sample rate of 2.8 megahertz, to redistribute the massive amount of resultant noise.

BTW, in the example I quoted previously (Lipsh*tz & Vanderkooy), they achieved a noise floor of -120.4dB with 8 bits and they didn't follow the Fletcher-Munson curve, they simply reduced the noise by 72dB throughout the freq band of 0hz-20kHz. Obviously that would result in a lot of noise needing to be redistributed (and all of it above 20kHz), so they used a sample rate of 176.4kS/s (44.1kS/s x 4) with the redistributed noise (at -19dBFS) occupying the 20kHz-88.2kHz audio band.

G

PS. I'm not sure how easy my explanation is to understand?


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## old tech

Thanks for the comprehensive explanation Greg.

And yes, it was easy to understand and answers the the three questions.


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## amirm

old tech said:


> I appreciate that even so, being limited to 22khz that our hearing is less sensitive at higher frequencies doesn't that the extra energy have an effect?


It can.  Your speakers and amplifiers will happily amplifier all that noise you have stuffed in the high frequencies 22 kHz.  Tweeters can resonate in that frequency and distortion (intermodulate) into more audible band.  And your amplifier can also oscillate.  After all, real music doesn't have such high ultrasonic content so equipment is not designed necessarily to handle such a situation.

This is why for noise shaping you want to have higher sample rate so that you can spread the shaped dither power over a wider area.

The simple answer to all of this is not to attempt to stuff down high-resolution audio into 16/44.1 when that format is essentially gone from many of our lives.  If the source is 24 bits, release it that.  If it is higher sample rate, leave that alone.  I can do the conversion myself if I want, thank you very much.  Or you as the distributor can offer both versions.  Don't force me into a spinning disc format when I am not spinning anything....


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## pinnahertz

amirm said:


> It can.  Your speakers and amplifiers will happily amplifier all that noise you have stuffed in the high frequencies 22 kHz.  Tweeters can resonate in that frequency and distortion (intermodulate) into more audible band.  And your amplifier can also oscillate.  After all, real music doesn't have such high ultrasonic content so equipment is not designed necessarily to handle such a situation.
> 
> This is why for noise shaping you want to have higher sample rate so that you can spread the shaped dither power over a wider area.
> 
> The simple answer to all of this is not to attempt to stuff down high-resolution audio into 16/44.1 when that format is essentially gone from many of our lives.  If the source is 24 bits, release it that.  If it is higher sample rate, leave that alone.  I can do the conversion myself if I want, thank you very much.  Or you as the distributor can offer both versions.  Don't force me into a spinning disc format when I am not spinning anything....


Now, realistically if you don't mind... have you ever...even once...known of an amplifier that broke into oscillation because of -70dBFS 20kHz noise-shaped dither that wouldn't have oscillated far more readily with a test signal, and been rejected in QC?  Or an amplifier that couldn't deliver 20kHz to a nominal load at even 50% power? 

The way IMD typically works is the intermod products that are produced are lower in amplitude than either of the intermodulating signals.  So if we had some noise-shaped dither at -70dBFS at 20kHz, and it intermodulated with some audio signal at  -70dBFS @ 17kHz to the level of 10%, that would put the 3kHz intermod product at -90dBFS, and it wouldn't be a discrete tone but rather a bit of noise. And that's worst case, because if you raise the 17kHz audio signal the resulting intermod will drop lower.   How is this a problem?

I'm not saying you're wrong, but rather trying to keep this in the real world.  It would be sad if someone saw this and said "Geez, dither is horrible!  It'll ruin my amp and tweeters!"  Nothing could be further from the truth.  You can be both correct but unrealistic.

As to releasing all that glorious 24bit audio, talk to Apple, Spotify, Amazon, Google, eMusic, and Napster - the top half dozen online music retailers - who are releasing only 16/44.  Or the top source for music streaming - YouTube - releasing in 16/44.  I agree, they could move to something higher if phones, smart speakers, and DMPs could handle it, but there's not much movement in that direction.  Just as there's even less recorded music where any of this would matter one tiny wit.


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## bigshot

Too much is never enough.


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## TheSonicTruth

bigshot said:


> Too much is never enough.



Spoken like an industry insider!  

Yes, 128bit/512k sampling may be beneficial during production, but 16/44.1 is FINE for delivery, in physical or download form.


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## gregorio

amirm said:


> [1] This is why for noise shaping you want to have higher sample rate so that you can spread the shaped dither power over a wider area.
> 
> [2] The simple answer to all of this is not to attempt to stuff down high-resolution audio into 16/44.1 when that format is essentially gone from many of our lives.  If the source is 24 bits, release it that.



1. What "shaped dither power"? Pinnahertz mentioned a peak shaped dither noise floor at -70dB, the worst case scenario I presented above (ID=16) would result in a shaped dither peak @ 22kHz of about -60dB. In practice though, why would a mastering engineer ever choose such an aggressive noise-shaping algorithm? What music in the real world ever requires 122dB of dynamic range? In practice we would use a far less aggressive alogorithm, typically somewhere around the equivalent of ID=11 or ID=12, which would give us a perceptual noise floor of about -110dB and peak shaped dither noise at about -84dB. Any consumer equipment which suffers from IMD caused by "shaped dither power" at -84dB is going to suffer from IMD when playing back the vast majority of recordings, as they virtually all have musical signals and/or recording noise floors at levels higher than this. A higher sample rate to accommodate the redistribution of "shaped dither power" is therefore only an issue when we're dealing with the large amounts of dither power generated by very few bits. For example, the theoretical example of 8 bit above or the practical example of 1 bit (SACD).

2. The source is never 24bits, so that's the end of that "if"! Commercial mix environments are pretty much always 64bit float and as we can't distribute 64bit files we ALWAYS have to dither/truncate ("stuff down"). So it isn't a question of 24bit without dither/truncation or 16bit with dither/truncation, it's simply a question of which bit depth we dither/truncate ("stuff down") to. In practice, 16/44.1 IS high resolution, there is NO audibly higher resolution!

G


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## bigshot

Amirm thinks that -120dB is the proper place for a noise floor. He claims that anything less is clearly audible (if he cranks the volume up on tiny samples of fade out).


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## pinnahertz

bigshot said:


> Amirm thinks that -120dB is the proper place for a noise floor. He claims that anything less is clearly audible (if he cranks the volume up on tiny samples of fade out).


Hope he doesn’t try that playing vinyl.


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## 71 dB

amirm said:


> Or you as the distributor can offer both versions.



Well, some distributors do offer many versions. Look here: https://autechre.bleepstores.com/release/98703-autechre-nts-sessions-1-4

They offer:

- Vinyl
- CD
- 24/44.1 wav
- 16/44.1 wav
- 16/44.1 FLAC
- 320 kbps mp3

I pre-ordered the CD set and I was able to download any version given. I chose 16/44.1 FLAC, because I don't need the "extra" dynamic range of 24 bit files for anything (assuming there actually is musical information below 16 bits which is possible as this is totally computer-generated music). I guess even the 320 kbps mp3s would have been totally transparent to _my_ ears, but I played it safe.


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## RRod

The ones I like lately are places that offer "studio" 24/96 and then "studio +!@11" 24/192, which I guess is twice as studio-er. They still take a back-seat to any place that deals in DXD, who format-whore like nobody's bizniss.


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## bigshot (Jul 21, 2018)

The funny thing is that the music most HD audio nuts are listening to at these astronomical bitrates is stuff that was considered ephemeral rock music back in the 60s and 70s, so it was recorded quick and on the cheap in the first place. Just today I had a guy on another forum tell me that he didn't understand how digital audio worked and didn't care. He just uses his ears. But whenever he talks about music, it's always in a high data rate format. I think his ears must tell him to pick big numbers.


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## 71 dB

bigshot said:


> I think his ears must tell him to pick big numbers.



Not his ears of course, but his mind.


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## castleofargh

admittedly, the less you understand how digital audio works, the more likely you are to imagine that bigger is always better. the mistake at least seems logical. the guy who can't visualize audio not being analogue is always going to wish to fill in those "holes" in the music. or have smaller stair steps, or whatever faulty way he uses to try and force digital into a faulty analog representation. 
if I knew nothing about digital audio, I believe I would have such ideas.


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## bigshot

castleofargh said:


> admittedly, the less you understand how digital audio works, the more likely you are to imagine that bigger is always better.



It also helps to not believe that digital sound isn't "perfect". When you believe that everything is flawed in some way, then any kind of improvement of those flaws must result in better sound. It's a carry-over from the analogue era where every format and component added its own noise, coloration and distortion. The theory back then was to strip back to just the essentials, split components into as many specialized individual boxes as possible, and avoid anything that alters the signal, even if it is correcting imbalances. None of those things are necessary any more.


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## Oncisounds

Hi, bit late to this whole party here but a very interesting article! However I have a question if I may and they might sound very silly so I apologise in advance however; 

You said that the greater the bit depth the less quantisation errors right? The correction for quantisation errors is to introduce dither which in turn introduces noise into the audio recording in order to get a true a reading of the waveform as possible, this means the less bit depth the more dither and in turn the more noise. With that I figured the higher bit depth the better not necessarily for dynamic range but more so for less noise being introduced. I figured noise would be present in audio at all dB or is it only at the noise floor that this noise is present? 

Sorry again if it sounds like a silly question still trying to understand this all.


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## castleofargh

Oncisounds said:


> Hi, bit late to this whole party here but a very interesting article! However I have a question if I may and they might sound very silly so I apologise in advance however;
> 
> You said that the greater the bit depth the less quantisation errors right? The correction for quantisation errors is to introduce dither which in turn introduces noise into the audio recording in order to get a true a reading of the waveform as possible, this means the less bit depth the more dither and in turn the more noise. With that I figured the higher bit depth the better not necessarily for dynamic range but more so for less noise being introduced. I figured noise would be present in audio at all dB or is it only at the noise floor that this noise is present?
> 
> Sorry again if it sounds like a silly question still trying to understand this all.


the noise from quantization is determined by the amount between the correct value and the approximation due to quantization, meaning it's going to be quiet noise in the least significant bit. with dither it's going to depend on the type of dither used. but in general, withing the audio band we're still talking about the least significant bits or even lower than that with noise shaping. 

obviously, getting higher bit depth is objectively good for the noise floor. or it would be if quantization noise was among the loudest noises you'd get in your audio chain and listening environment. which might not be the case all that often. you'll need pretty clean gears, very quiet room, and to listen to music loudly, just to possibly make quantization noise the loudest noise of all. and even then, there is no guaranty that you'll detect it. a lot of abx tests between 16 and 24bit suggest we usually don't.  
the relevance of increasing bit depth is really a matter of context. as far as audio playback is concerned, the noise floor of 16bit is already pretty cool.


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## 71 dB

Oncisounds said:


> Hi, bit late to this whole party here but a very interesting article! However I have a question if I may and they might sound very silly so I apologise in advance however;
> 
> You said that the greater the bit depth the less quantisation errors right? The correction for quantisation errors is to introduce dither which in turn introduces noise into the audio recording in order to get a true a reading of the waveform as possible, this means the less bit depth the more dither and in turn the more noise. With that I figured the higher bit depth the better not necessarily for dynamic range but more so for less noise being introduced. I figured noise would be present in audio at all dB or is it only at the noise floor that this noise is present?
> 
> Sorry again if it sounds like a silly question still trying to understand this all.



There is nothing wrong asking when you don't know! 

Yes, the less bit depth the more dither noise you need, but the dither noise is constant in level. It doesn't matter if there's silence in the music or if the music is blasting off Dither is always the same. It's like having a fan blowing air while listening to music. The noise from the fan is constant. However, in 16 bit audio dither is so quiet you can't hear it. Dither becomes much more important when we go under 16 bits (8 bits for example). In those cases a very crappy badly granulating distortied sound if replaced with undistorted sound with heavy noise which is much better.


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## Oncisounds

71 dB said:


> There is nothing wrong asking when you don't know!
> 
> Yes, the less bit depth the more dither noise you need, but the dither noise is constant in level. It doesn't matter if there's silence in the music or if the music is blasting off Dither is always the same. It's like having a fan blowing air while listening to music. The noise from the fan is constant. However, in 16 bit audio dither is so quiet you can't hear it. Dither becomes much more important when we go under 16 bits (8 bits for example). In those cases a very crappy badly granulating distortied sound if replaced with undistorted sound with heavy noise which is much better.



Ahh okay that makes sense, see in my head the less quantisation the less dither you would need to correct for it hence the less noise in the audio interfering. But now I know it's a constant and always present but basically inaudible it kinda makes a lot of sense, same sort od argument as refresh rate in monitors etc.


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## gregorio (Oct 6, 2018)

Oncisounds said:


> Ahh okay that makes sense, see in my head the less quantisation the less dither you would need to correct for it hence the less noise in the audio interfering.



Yep, that is correct. The point/facts you were missing was how relevant and audible that dither noise is. When we record something, the recording venue always has noise, the mics used always add noise, the mixing and mastering processes add noise and then when you reproduce the recording, your amp adds noise, your HPs or speakers add noise and your listening environment has noise. All these sources of noise add different amounts of noise, for example a good amp will generally add very little noise, usually less than 16bit dither noise but in most other cases dither noise is lower or much lower than any one of the other noise sources, let alone all those other noise sources combined. As mentioned, it is sometimes possible to hear dither noise but the conditions required to do so are not realistic. First you have to pick a recording that has particularly low levels of recording, mixing and mastering noise, your system and playback environment have to have very low levels of noise and then you have to play that recording at very loud levels. Even if you have such a recording/s and reproduction system/environment, then the human ear adjusts itself for low noise and for comfort sake one would listen at a lower level than normal rather than much higher than normal. In other words, unless you manufacture the unreasonable conditions specifically to hear dither noise of 16bit, it is always going to be inaudible and irrelevant.

Hope this makes sense?

G


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## Oncisounds

gregorio said:


> Yep, that is correct. The point/facts you were missing was how relevant and audible that dither noise is. When we record something, the recording venue always has noise, the mics used always add noise, the mixing and mastering processes add noise and then when you reproduce the recording, your amp adds noise, your HPs or speakers add noise and your listening environment has noise. All these sources of noise add different amounts of noise, for example a good amp will generally add very little noise, usually less than 16bit dither noise but in most other cases dither noise is lower or much lower than any one of the other noise sources, let alone all those other noise sources combined. As mentioned, it is sometimes possible to hear dither noise but the conditions required to do so are not realistic. First you have to pick a recording that has particularly low levels of recording, mixing and mastering noise, your system and playback environment have to have very low levels of noise and then you have to play that recording at very loud levels. Even if you have such a recording/s and reproduction system/environment, then the human ear adjusts itself for low noise and for comfort sake one would listen at a lower level than normal rather than much higher than normal. In other words, unless you manufacture the unreasonable conditions specifically to hear dither noise of 16bit, it is always going to be inaudible and irrelevant.
> 
> Hope this makes sense?
> 
> G




Yeah it makes perfect sense I figured that would be the case, hence why the noise floor is so high, as that's the levels where it would start to become apparent. But thank you for expaining it all, it makes a lot of sense that noise from all aspects of recording to processing to digital output will cause noise of similar magnitude if not even more so than that of the dither. 

Thanks guys!


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## Haohua

thank you for the knowledge.


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## PanzerIV

Wow this is one of the most interesting read I've done. I might be almost 10 years late but this is still relevant info that anybody should know. At least now I know my ears aren't wrong as I was like "What I can't hear any tiny difference AT ALL between regular files and those higher bit super rare and expensive Hi-Res albums?!" but noticing any difference would only have been placebo effect. When you have in $CAD so remove 33% value, an headphone of 2700$ (*Audeze LCD3*), a 2000$ DAC (*NAD M51*), a 1600$ headphone amp (*Woo WA6-SE*) and 550$ cables (*Audioquest NRG-Y3 + Mackenzie RCA*), you should have stuff good enough to hear a difference if there's one as it's already better than what 99.9% of people have at home!

I remember reading somewhere here, that recording techniques at the studio was faaaar more important than the actual Bit/Khz numbers and I totally believe that. It's pretty much imo like those cheap 200$ phones that say it records at 1080p blablabla or huge MegaPixel but with absolutely crapty and tiny integrated lens. Trust me I could record even in 720p or take a picture at 3x lower MP than ur phone with my DSLR camera and professional lens with super good optical glass, and it's gonna look infinitely better. As like anything in life, quality over quantity! People really don't realize that 1920x1080 is only a tiny *2.07MP*! Then they think their 40MP cellphone is gonna be Nasa quality lol.


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## Slaphead

PanzerIV said:


> Wow this is one of the most interesting read I've done. I might be almost 10 years late but this is still relevant info that anybody should know. At least now I know my ears aren't wrong as I was like "What I can't hear any tiny difference AT ALL between regular files and those higher bit super rare and expensive Hi-Res albums?!" but noticing any difference would only have been placebo effect. When you have in $CAD so remove 33% value, an headphone of 2700$ (*Audeze LCD3*), a 2000$ DAC (*NAD M51*), a 1600$ headphone amp (*Woo WA6-SE*) and 550$ cables (*Audioquest NRG-Y3 + Mackenzie RCA*), you should have stuff good enough to hear a difference if there's one as it's already better than what 99.9% of people have at home!
> 
> I remember reading somewhere here, that recording techniques at the studio was faaaar more important than the actual Bit/Khz numbers and I totally believe that. It's pretty much imo like those cheap 200$ phones that say it records at 1080p blablabla or huge MegaPixel but with absolutely crapty and tiny integrated lens. Trust me I could record even in 720p or take a picture at 3x lower MP than ur phone with my DSLR camera and professional lens with super good optical glass, and it's gonna look infinitely better. As like anything in life, quality over quantity! People really don't realize that 1920x1080 is only a tiny *2.07MP*! Then they think their 40MP cellphone is gonna be Nasa quality lol.



Now try the same thing with varying levels of MP3 and AAC compressed audio against the original 16/44 - I'll bet you'll get to higher levels of compression before you hear an obvious difference than you thought you would - although I will say that some people are more sensitive to the artefacts that compression brings than others are. 

When I say that I don't mean hearing ability, it's just that I believe the standard psychoacoustic model used in compression fits better to some people than it does to others. The psychoacoustic model used in compression is a one size fits all approach, and thus there will be people that lie outside of that which model dictates.

You're absolutely right about recording techniques - you'd be lucky to get to 60dB (or 10 bits) of dynamic range in even the best setup studio. However that's not to say that hi-res audio isn't useful - in fact it's essential in the production phase where the music is mixed and then ultimately mastered as hi-res audio gives you the headroom in which to apply levels, EQ and other effects without introducing audible quantisation errors. However for end point delivery 16/44 is all that's needed, in fact it's more than is needed.


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## PanzerIV

Slaphead said:


> Now try the same thing with varying levels of MP3 compressed audio against the original 16/44 - I'll bet you'll get to higher levels of compression before you hear an obvious difference than you thought you would - although I will say that some people are more sensitive to the artefacts that compression brings than others are.


100% Agree with what you said that I took off from this quote, as yes I can totally understand this extra headroom is needed for the mixing phase, but not to us final user listening at it.

I'm just not entirely sure to understand though what you meant in that quote above. Did you mean that I could be surprised how low the bitrate of an MP3 needs to be BEFORE I finally start noticing artifacts? To be honest, I've never noticed even the tiniest difference between (*CBR 320*) and (*VBR 0*) but as I found the idea of having much more efficiency by having a file size lower with 0% loss of quality, I converted all CBR320 albums I downloaded or bought, into VBR0. However if I'm lucky enough to get a Flac file, now that I got the equipment to listen at it, I'll just keep them that way as these days storage cost absolutely nothing. It's sad though that Flac file are still very rare and you either must buy the album and sometime they still don't offer Flac, or go on specialized websites. I would never spend money into buying MP3 albums, that's for sure!


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## gregorio

PanzerIV said:


> [1] ... but noticing any difference would only have been placebo effect. ... I remember reading somewhere here, that recording techniques at the studio was faaaar more important than the actual Bit/Khz numbers and I totally believe that.
> [2] It's pretty much imo like those cheap 200$ phones that say it records at 1080p blablabla or huge MegaPixel but with absolutely crapty and tiny integrated lens.



1. To be pedantically accurate, noticing any difference may not ONLY be due placebo effect.  The reason for this is that your second sentence is true, recording, mixing and mastering are far more important and there are quite a few cases where the CD version of a recording has different mastering than the hi-res version and therefore it can be relatively easy to tell the difference in some cases. As there is no intrinsic audible difference between CD and hi-res, there is the temptation to reduce the quality of the CD version and justify the higher price of the hi-res version. In some cases where audiophiles believe they're noticing a difference, it is as you say placebo effect but in other cases they maybe noticing a real, actual difference but it's between different masters, not the audio file format.

2. We have to be a bit wary of analogies with video and audio. HD in video (even 720p) is technically AND VISUALLY higher definition than standard definition. When greater than 16/44.1 audio became available, the audiophile marketing world jumped on the "HD" video bandwagon, but in order for the term "HD" (or "Hi-res") to have any meaning they had to call CD (16/44.1) "standard definition". This was effectively a lie/false marketing because CD is NOT anywhere near equivalent to standard definition video. Relative to the capabilities of human vision and human rearing, CD is in fact higher resolution/definition than even the latest 4K HDR video specification! Once we have a video specification that exceeds ALL the capabilities of human vision, then higher definition than that would be pointless and that's the situation we have with CD audio.



Slaphead said:


> [1] You're absolutely right about recording techniques - you'd be lucky to get to 60dB (or 10 bits) of dynamic range in even the best setup studio.
> [2] However that's not to say that hi-res audio isn't useful - in fact it's essential in the production phase where the music is mixed and then ultimately mastered as hi-res audio gives you the headroom in which to apply levels, EQ and other effects without introducing audible quantisation errors.



1. That's not always the case, some of the best studios can manage a dynamic range of over 90dB. It's VERY expensive though, you're looking at double-shell construction and other expensive isolation strategies/treatments to get the noise floor of the studio very low and then a particularly good sound system capable of high output levels (and low noise/distortion).

2. We have to be careful here and make the distinction between the "mix environment" and the "audio file format" absolutely CLEAR. Modern professional mix environments are typically 64bit (float), all processing ("levels, EQ and other effects") occurs at this bit depth and therefore even cumulative quantisation errors are way below our ability to even reproduce, let alone hear. However, this mix environment is completely INDEPENDENT of the bit depth of the audio file format we load into it. Whether we load a 16bit file or 24bit file into the mix environment is completely irrelevant as the processing (and quantisation errors) will all be at the 64bit level. So as far as headroom for processing during mixing and mastering is concerned, it's EXACTLY THE SAME with 16bit and 24bit audio files. The ONLY place where a 24bit file format can make any difference is for additional headroom during recording.

G


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## TheSonicTruth (Dec 28, 2018)

gregorio said:


> 1. To be pedantically accurate, noticing any difference may not ONLY be due placebo effect.  The reason for this is that your second sentence is true, recording, mixing and mastering are far more important and there are quite a few cases where the CD version of a recording has different mastering than the hi-res version and therefore it can be relatively easy to tell the difference in some cases. As there is no intrinsic audible difference between CD and hi-res, there is the temptation to reduce the quality of the CD version and justify the higher price of the hi-res version. In some cases where audiophiles believe they're noticing a difference, it is as you say placebo effect but in other cases they maybe noticing a real, actual difference but it's between different masters, not the audio file format.
> 
> 2. We have to be a bit wary of analogies with video and audio. HD in video (even 720p) is technically AND VISUALLY higher definition than standard definition. When greater than 16/44.1 audio became available, the audiophile marketing world jumped on the "HD" video bandwagon, but in order for the term "HD" (or "Hi-res") to have any meaning they had to call CD (16/44.1) "standard definition". This was effectively a lie/false marketing because CD is NOT anywhere near equivalent to standard definition video. Relative to the capabilities of human vision and human rearing, CD is in fact higher resolution/definition than even the latest 4K HDR video specification! Once we have a video specification that exceeds ALL the capabilities of human vision, then higher definition than that would be pointless and that's the situation we have with CD audio.
> 
> ...



"_1. To be pedantically accurate, noticing any difference may not ONLY 
be due placebo effect. The reason for this is that your second sentence 
is true, recording, mixing and mastering are far more important and there 
are quite a few cases where the CD version of a recording has *different 
mastering than the hi-res version* and therefore it can be relatively easy to 
tell the difference in some cases. As there is no intrinsic audible difference 
between CD and hi-res, there is the temptation to reduce the quality of the 
CD version and justify the higher price of the hi-res version.*.."*_



I'm genuinely surprised that, as a widely regarded mastering engineer, you would admit that, Mr. C! I've been saying that for years, about composition, performance, recording, and mastering all being more critical than specific format.


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## bfreedma

TheSonicTruth said:


> "_1. To be pedantically accurate, noticing any difference may not ONLY
> be due placebo effect. The reason for this is that your second sentence
> is true, recording, mixing and mastering are far more important and there
> are quite a few cases where the CD version of a recording has *different
> ...




I’m surprised that you’re surprised given that no one has ever argued that point in this forum.  Different mastering of CD vs. SACD and it’s potential audibility has been consistently discussed here.  I certainly don’t recall @gregorio ever stating that when properly utilized, format was important, let alone more critical than the music, recording, and mastering.


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## TheSonicTruth

bfreedma said:


> I’m surprised that you’re surprised given that no one has ever argued that point in this forum.  Different mastering of CD vs. SACD and it’s potential audibility has been consistently discussed here.  I certainly don’t recall @gregorio ever stating that when properly utilized, format was important, let alone more critical than the music, recording, and mastering.



But it is when different mastering is used to set  one format apart from another that is dishonety in marketing.  Not impugning Gregorio's work in any way, just a general statement.


----------



## gregorio

TheSonicTruth said:


> I'm genuinely surprised that, as a widely regarded mastering engineer, you would admit that, Mr. C!



Firstly, I'm not a "widely regarded" mastering engineer. I've worked with some top mastering engineers and I always advise my clients to go to one of them but despite this advice, I have a handful of clients who insist I master for them. Having said this, I work mostly in Film these days and in the TV/Film world, the mix engineer is also effectively the mastering engineer. Secondly, it's hardly news, much earlier in this thread there were two instances of actual record labels contributing to this thread; Linn Records was one and you can see the exchange that started around post #401. I believe the other was a representative of HD Tracks (although it might have been another audiophile label) but it was many years ago and I can't find the exchange. Linn for example put up what they called a "real world" CD vs "Hi-res" listening challenge. The two example files were easy to tell apart and that was because they were in fact different masters! When challenged on this subterfuge Linn admitted they were different masters but effectively stated that because they always required the mastering engineer to create a different master for the CD version, that's why they called it a "real world" challenge. HDTracks (if that's who it was) not only admitted that their CD versions were different but admitted that they required their CD versions be mastered with more compression applied. The explanation was (paraphrasing from memory) that they expected CDs to be purchased by consumers who were going to rip them to a lossy format for their iPods and therefore more compression/louder would be appreciated, while the "hi-res" version was less compressed and designed more for audiophile consumption. I asked why they didn't sell both different versions on CD (16/44.1), why was the audiophile version only available in the higher bit depth/sample rate format (for which they charged substantially more), even though there would be no audible difference and that it actually cost more to produce the cheaper CD version? The answer, maybe unsurprisingly, was nothing at all, there was no response or any further contribution to the thread.

It was clear that both those audiophile labels saw this thread as a typical audiophile marketing opportunity but once they realised their usual nonsense wasn't going to work in this sub-forum, they promptly disappeared! Bare in mind this was nearly a decade ago and was just two small audiophile labels.

G


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## bigshot

Different mastering can be better, or it can be worse. It all depends on how good of a job they do and what the intended purpose is. And it isn't just mastering that might be different between CD and SACD... it might even be completely remixed. A lot of SACDs that have multichannel mixes will put the 2 channel fold down of the 5.1 in the stereo track. You can't tell much by just trusting the numbers. You have to talk to people who have listened to the release to know if it's worth getting.


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## TheSonicTruth (Dec 28, 2018)

gregorio said:


> Firstly, I'm not a "widely regarded" mastering engineer. I've worked with some top mastering engineers and I always advise my clients to go to one of them but despite this advice, I have a handful of clients who insist I master for them. Having said this, I work mostly in Film these days and in the TV/Film world, the mix engineer is also effectively the mastering engineer. Secondly, it's hardly news, much earlier in this thread there were two instances of actual record labels contributing to this thread; Linn Records was one and you can see the exchange that started around post #401. I believe the other was a representative of HD Tracks (although it might have been another audiophile label) but it was many years ago and I can't find the exchange. Linn for example put up what they called a "real world" CD vs "Hi-res" listening challenge. The two example files were easy to tell apart and that was because they were in fact different masters! When challenged on this subterfuge Linn admitted they were different masters but effectively stated that because they always required the mastering engineer to create a different master for the CD version, that's why they called it a "real world" challenge. HDTracks (if that's who it was) not only admitted that their CD versions were different but admitted that they required their CD versions be mastered with more compression applied. The explanation was (paraphrasing from memory) that they expected CDs to be purchased by consumers who were going to rip them to a lossy format for their iPods and therefore more compression/louder would be appreciated, while the "hi-res" version was less compressed and designed more for audiophile consumption. I asked why they didn't sell both different versions on CD (16/44.1), why was the audiophile version only available in the higher bit depth/sample rate format (for which they charged substantially more), even though there would be no audible difference and that it actually cost more to produce the cheaper CD version? The answer, maybe unsurprisingly, was nothing at all, there was no response or any further contribution to the thread.
> 
> It was clear that both those audiophile labels saw this thread as a typical audiophile marketing opportunity but once they realised their usual nonsense wasn't going to work in this sub-forum, they promptly disappeared! Bare in mind this was nearly a decade ago and was just two small audiophile labels.
> 
> G



WOW.  What insight into that whole HDTracks thing!  Looks like I'm not missing anything by spending hard-earned dough on stuff with more DR compression put on a higher-res format.  I wouldn't mind hearing the same stereo masters used, I.E., for my 1985 Thriller CD, unaltered, on a higher-res format, just to know that there really wouldn't be that much of a difference.  Of course, that's wandering into territory already covered in that thread "...Claims and Myths", lol!


----------



## bigshot

If the mastering was the same, there would be no difference between SACD and CD. If you want a test to make sure, just buy a Pentatone SACD disc and rip the CD layer and to a line level matched, direct A/B switched blind test. That will tell you exactly how irrelevant format is to sound fidelity. Pentatone is one of the few labels that uses the same mastering on both layers.


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## old tech

bigshot said:


> If the mastering was the same, there would be no difference between SACD and CD. If you want a test to make sure, just buy a Pentatone SACD disc and rip the CD layer and to a line level matched, direct A/B switched blind test. That will tell you exactly how irrelevant format is to sound fidelity. Pentatone is one of the few labels that uses the same mastering on both layers.


While I don't doubt that Pentatone use the same mastering on both layers, it is a mystery to me as to why.

I sort of can understand having different masterings on either layer, either for a deception in favour of the SACD or for practical purposes so a more compressed CD layer to sound better in noisier environments like listening in a car.

What I don't understand is why would someone who has invested in a SACD player, the higher price of a SACD (and who presumably believes it results in sound quality improvements) would want a CD layer?


----------



## bigshot

When I was comparing SACD to CD, I found that particularly with rock albums they would put older masterings on the CD layer and the newer ones on the SACD layer. In one case I found a song that had a totally different mix on the two layers. The redbook layer is generally at a lower level too. I think they do this to enhance the perception of quality of the SACD layer. If both sounded the same, what would be the point of buying an SACD? They stack the deck a bit by hobbling the CD layer.

Pentatone is a classical label that sells only hybrid SACDs. None of their recordings are available as CDs. I would bet a majority of their customers don't even have an SACD player. They buy the discs to play them as normal CDs. This makes them motivated to not hobble the CD layer. That's my theory at least.


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## stonesfan129

24-bit and DSD make sense for archival and mastering.  I was never able to hear an audible difference after I converted them to CD-quality WAV files and did a blind test.  In fact, I can't even hear any difference between 256k AAC files and CD-quality tracks.  Maybe my hearing just sucks but if the difference is there I don't hear it or just don't know how to listen for it.  I'm on a 1st-gen FiiO X1 and Sennheiser HD598SE headphones.


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## 71 dB

bigshot said:


> (1) When I was comparing SACD to CD, I found that particularly with rock albums they would put older masterings on the CD layer and the newer ones on the SACD layer. In one case I found a song that had a totally different mix on the two layers. The redbook layer is generally at a lower level too.
> (2) I think they do this to enhance the perception of quality of the SACD layer.
> (3) If both sounded the same, what would be the point of buying an SACD?
> (4) Pentatone is a classical label that sells only hybrid SACDs. None of their recordings are available as CDs. I would bet a majority of their customers don't even have an SACD player. They buy the discs to play them as normal CDs. This makes them motivated to not hobble the CD layer. That's my theory at least.



(1) So you have two masterings on one disc…
(2) Hateful if this is the case.
(3) Multichannel support is the only point of SACD, because for stereo sound CD is all we need.
(4) I don't have Pentatone's releases, but I have SACDs from BIS, CPO + some other labels. I have never noticed any "hobbling" with the CD layer. It's classical music and the rules are probably very different from rock.


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## ALRAINBOW

Regardless of meaning how does the same track sound 24/16 ? For me it's obvious that redbook missed the Boat at using 16 . Any same track at 24 shows a much lower noise floor 
It plays from blacker background. 
Even 24/88.2 is miles better.  
The lower noise floor maybe due to a much higher band width some 256 more in dynamics. 
So asdide from math does anyone hear this improvement ?


----------



## stonesfan129

ALRAINBOW said:


> Regardless of meaning how does the same track sound 24/16 ? For me it's obvious that redbook missed the Boat at using 16 . Any same track at 24 shows a much lower noise floor
> It plays from blacker background.
> Even 24/88.2 is miles better.
> The lower noise floor maybe due to a much higher band width some 256 more in dynamics.
> So asdide from math does anyone hear this improvement ?



I don't.  I have never seen any AES paper proving there is an audible difference in fact.  I thought 16-bits already had such a low noise floor no one could hear it.  If I hear a difference in the 24-bit version (which a lot of Mastered For iTunes AAC files are sourced from), it's because it's a different mastering than the CD.


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## ALRAINBOW

Just try it forget papers ok I know what you mean and I am not nuts or hear like a bat lol. 
Any same tracks in 24 bit depth blows away 16 even at 44.1 or 88.1 bit res 
Many cd titles were done at 24/88.2 as well a shame it was not the red book standard. Try then post if I have to I'll do a share link for you. 
Formats matter too but this is another discussion


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## castleofargh

ALRAINBOW said:


> Regardless of meaning how does the same track sound 24/16 ? For me it's obvious that redbook missed the Boat at using 16 . Any same track at 24 shows a much lower noise floor
> It plays from blacker background.
> Even 24/88.2 is miles better.
> The lower noise floor maybe due to a much higher band width some 256 more in dynamics.
> So asdide from math does anyone hear this improvement ?


the thing with noise is that there is always some. so common sense makes us look for a recording medium that will have self noise at lower level than the most of the other loud noises that will reach us. 24bit is overkill even for ideal playback in the Batcave. for starters, our audio gears don't resolve 24bit. but that's only the start of how irrational we are when we imagine that 24bit albums offer 24bit of resolution. it's not like the band is going to play and sing at 160dB SPL just to make sure what's recorded doesn't contain ambient noise from the studio, when a singer records a gentle song, having 24bit is only relevant if we were trying to use some NCIS magic enhancer tools to find out when the truck passed 2 streets below the studio. ^_^  
in practice for most albums you can also remove a few bits for headroom and gain matching of the various tracks mixed into the song. each mic will have recorded the ambient noise of the studio and added it's own noise. when playing back the song in our room, how loud will we play it? depending on the answer, the ambient noise in the room will have by far the loudest noises. 
so already you get a good idea as to why noticing the noise difference in a song is not as easy as audiophiles like to say it is. simply because the stuff we're looking for is drowned in many different and louder noises. when you know all that you can of course manufacture situations where you'll be able to hear the quantization noise of a 16bit track, but such a situation is not similar to listening to your typical recorded album. this is again something the guys from NCIS or some spy in the other room will have issues with. not you and your favorite album. 

that was only about noise and only about an objective assessment of noise levels and how 16bit is so very unlikely to come close to the the highest noises from the track, from your gear, from your room, etc. now consider auditory masking, as soon as you will have sounds close to 0dB on a digital recording at a given frequency, I dare you to notice another quieter signal around that frequency at 70 or 80dB below.  the music itself is effectively covering most noises. in fact it does so at such levels that just the distortions from the music will be signals loud enough to mask most noises down at 16bit or lower if they ever happened to actually be recorded in the first place. our own hearing has a pretty limited dynamic range. we say it's around 120 or 130dB, but that's only when you try in a super quiet room with amazing acoustic treatment and the quietest signals are played alone!!!!! any loud noise in the room, and for a while you won't be able to notice the quiet sounds anymore(just like you need to have some quiet time to start noticing how "loud" your own heartbeat really is). and of course it implies that the loudest sound is around 120 or 125dB SPL where sound becomes actual pain. we've had discussions with people who consider that listening to music with 120dB SPL peaks at home is normal in their opinions. in my country it's about the loudest you can legally play in a club or at a live event. so yes it's possible and yes it happens, but if you listen to music at those levels on a regular basis for significant periods of time, chances are that you soon will have more important things to be concerned about than HiFi. and again, even then, it does not mean you'd get to perceive 120dB of dynamic, because those loud sounds would absolutely mask the quieter ones, no matter how accurately they got recorded. 

so for all those reasons, it's very unlikely that noise from a 16bit track will be audible to us in playback. if you perceive an obvious difference with hires tracks, first it would be a good idea to make sure it's not in your head with a blind test. and maybe even before that, it would be a good idea to check that you're comparing the same mastering, by converting your hires track to 16bit and then doing a blind listening test between those 2. then if you happen to still notice a difference, which is unlikely but happens, you might want to look up if your DAC isn't one of those to roll off the trebles when playing 44.1khz. nothing to do with bit depth but as people usually compare high sample rate and high bit rate against CD resolution, it might be worth checking with a loop and a measurement of the frequency response at various sample rates. 
on the same idea, if using a computer, it could be relevant to look up the resampling settings. on occasion resampling has been known to suck enough that it made an audible difference. again that's sample rate and not bit depth, I really doubt that you would hear anything between 16 and 24bit. once all those things have been controlled, if you still perceive a difference in a blind test, I suggest to sell your listening skills to all the big brands selling hires gears and tracks, and become the guy to show the clearly audible superiority of hires once and for all.


----------



## 71 dB

ALRAINBOW said:


> Regardless of meaning how does the same track sound 24/16 ? For me it's obvious that redbook missed the Boat at using 16 . Any same track at 24 shows a much lower noise floor
> It plays from blacker background.
> Even 24/88.2 is miles better.
> The lower noise floor maybe due to a much higher band width some 256 more in dynamics.
> So asdide from math does anyone hear this improvement ?




*13 bits*
What is needed in _consumer_ audio is 13 bits worth of dynamic range (about 80 dB). CD is kind of overkill already, but only 3 bits so who cares? 24 bit audio is technically 11 bits overkill, but in practice a lot less than that, because of the lowest 8 bits most if not all is noise unless the sound is completely computer generated (so, *Autechre*'s 24 bit files might actually have 24 bit worth of dynamic range, but of course you can't really experience it for many reasons and the same files dithered to 13 bits would sound the same).

If you hear the noise floor of 16 bit audio, the reason is:

(1) The recording contains high levels of background noise. 24 bit version would have the same noise.
(2) Your volume is turned to insanely high levels to hear the noise floor. You would never listen to music at those insane levels.

Dither expands dynamic range below the LSB, in fact there is no limit to that other than the fact that the dither noise masks the signal depending on the type of dither. Noise shaped dither noise mask less and can expand the perceivable dynamic range even 20 dB! Quiet sounds do not granulate (modulate the noise), but they "live" in the noise like signals in analog audio.

How "loud" is dither noise? You can generate TPDF dither in Audacity following these steps:

STEP 1 - Open Audacity and select *Generate* ---> *Silence* ---> 10 seconds
STEP 2 - Select* Audio Track* ---> *Set Sample Format* ---> 24 bit PCM
STEP 3 - Select* Audio Track* ---> *Set Rate* ---> 44100 Hz
STEP 4 - Select *Effect* ---> *Nyquist Prompt* ---> write (mult (sim (noise) (noise)) (recip 65536)) and click ok 

Try to hear the noise. The level meters show there's stuff at level around -90 dB. Turn up volume until you hear the noise. Now, listen to a music file using the same volume setting if you dare! Don't do it, because you probably damage your ears and gear! That's how quiet TPDF dither is. Noise shaped dithers are perceptually even quieter!


*44.1/48 kHz*
No higher samplerate is needed than 44.1 or 48 kHz in _consumer_ audio. Those cover the human hearing range 20-20000 Hz in the childhood (upper limit lowers with age). However, if the music is for dogs (a dog whistle concerto), 88.2 kHz is kind of the lower limit I would use, 96 kHz would be safer as dogs can hear up to 45 kHz I believe. Cats can hear up to almost 80 kHz, so their audio formats would probably use 176.4 kHz or 192 kHz sampling rates. Dolphins would need to use 352.8 kHz!


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## ALRAINBOW

It's very obvious and not a white paper thing. I truly follow you on this and do feel as you do. Big time over kill but to hear it it's the point 
Now why I hear it is another point to be made. 
Take any CD player play red book it's obvious if I go to a sever of same mine is lower even at red book. But go up to 24 bit depth it's a game changer for me. Maybe my setup is this apparent but it is not my ears at 61 no way am I great. 
Try it ok. I am out now when I get back I'll put up two songs at redbook and 24/88.2 you play and advise ok. It's just obvious they its blacker less hash.  Now given why no idea but I can measure the noise with a scope at analog outputs


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## 71 dB

ALRAINBOW said:


> It's very obvious and not a white paper thing. I truly follow you on this and do feel as you do. Big time over kill but to hear it it's the point
> Now why I hear it is another point to be made.
> Take any CD player play red book it's obvious if I go to a sever of same mine is lower even at red book. But go up to 24 bit depth it's a game changer for me. Maybe my setup is this apparent but it is not my ears at 61 no way am I great.
> Try it ok. I am out now when I get back I'll put up two songs at redbook and 24/88.2 you play and advise ok. It's just obvious they its blacker less hash.  Now given why no idea but I can measure the noise with a scope at analog outputs


What your (analog) gear does is another story. Could be _your_ gear does something nasty with redbook and performs better at other formats, but even half-decent DACs of today do very good transparent job with 16/44.1. That wasn't the case in the early 80's when CD format emerged, but digital audio and DACs have since matured to perfection allowing CDs to be a transparent format meaning it's all we need for stereo sound.


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## ALRAINBOW

It's complex for me to say what I use but let's say it's very state of the art being 3 pc s for music process and various dacs all act the same at there own level of quality. 
I'll post soon as I can and let's post Honest of what we hear and I am completly with you on your white paper ideals. It's nuts but I am not either lol.


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## bigshot (Dec 31, 2018)

I think you've got some expectation bias going there. This isn't just theoretical. A lot of us have actually done controlled listening tests and have proven to ourselves that audibly transparent is all you need. The trick is to make sure you're comparing apples to apples. Mastering between formats can be quite different.


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## bfreedma

ALRAINBOW said:


> Just try it forget papers ok I know what you mean and I am not nuts or hear like a bat lol.
> Any same tracks in 24 bit depth blows away 16 even at 44.1 or 88.1 bit res
> Many cd titles were done at 24/88.2 as well a shame it was not the red book standard. Try then post if I have to I'll do a share link for you.
> Formats matter too but this is another discussion



Have to love when people post in Sound Science and the first sentence in the post is an instruction to ignore science...


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## ALRAINBOW

by the comment your not trying  either it's amazing how much can be done on paper with no real-time hearing 
cheers to your ignorance in this. 
So on paper bits are bits right 
Jitter does not exist 
Flabby bass or schilling highs it's the System or recording. Such little actual knowing and so much more on a parrots level and repeat. If I posted it it's real try it or shut up about how it can be. 
I will upload soon as I can to push for anyone who can take the time to hear it.


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## TheSonicTruth

ALRAINBOW said:


> by the comment your not trying  either it's amazing how much can be done on paper with no real-time hearing
> cheers to your ignorance in this.
> So on paper bits are bits right
> Jitter does not exist
> ...



Nirvana fan perchance?

Read up and get back to us:  

https://www.audiostream.com/content/high-resolution-downloads-nevermind


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## bfreedma (Dec 31, 2018)

ALRAINBOW said:


> by the comment your not trying  either it's amazing how much can be done on paper with no real-time hearing
> cheers to your ignorance in this.
> So on paper bits are bits right
> Jitter does not exist
> ...




Insults are no substitute for evidence.  Either produce supporting documentation/test results that anything on your “audiophile list of things you imagine impact reproduction” are audible to humans (assuming your gear isn’t broken/defective) or please take your posting to the appropriate forums on head-FI.  It’s more than a little ironic that you accuse anyone of parroting when that’s exactly what you’re doing - parroting the marketing materials of the vendors looking to sell you solutions in search of problems to solve. 

“If I posted it it’s real try it or shut up”?  Who knew the Sound Science quote of 2018 would come on the last day of the year.


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## ALRAINBOW

You say my stuff is broken lol and quote papers to what end ? You ssy my stuff is broken wjat do you use ?


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## bfreedma

ALRAINBOW said:


> You say my stuff is broken lol and quote papers to what end ? You ssy my stuff is broken wjat do you use ?




I didn’t say “you’re stuff was broken”. I said that what you’re asserting as problems (jitter, altered bass/treble) are not issues on properly working audio reproduction electronics.  If you have evidence supporting an alternate view, feel free to post it.


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## stonesfan129 (Dec 31, 2018)

Show me a single AES paper or some kind of scientific data that shows an audible difference between 16 and 24 bits.  I have never seen one.  As I wrote a few posts back, formats like 24-bit PCM and DSD are beneficial to mastering but make no difference for listening.  The CD-standard (16-bit, 44.1khz, 1411kbps) perfectly recreates everything audible to the best human ears.  This _has_ been backed up by years of research called the Nyquist Theory.  I would even doubt that jitter is affecting the sound.  Everything out there has enough processing power to recreate everything bit perfectly.  This is like the people who say there is an audible difference in WAV and FLAC.


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## ALRAINBOW

Just wait for my files it's obvious


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## bfreedma

ALRAINBOW said:


> Just wait for my files it's obvious



If it’s obvious, then it will be easily measurable.   When you post, please make sure to provide the provenance of the files to make sure we’re dealing with the same master, how it’s been converted, and that no processing occurred during the conversion.

In all likelihood, I’ll take the higher res version and bounce it down to red book myself to ensure this is an apples to apples comparison.

Al, you do realize this is futile, right?  I’ll listen and unless 2019 brings new laws of physics and other impacting science, I won’t hear a difference.  Then you’ll claim I have bad ears or bad gear (neither is true) and we’ll be right back where we started...


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## stonesfan129

I'll keep an open mind and hear the files first.


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## bfreedma

stonesfan129 said:


> I'll keep an open mind and hear the files first.




I’m not saying I won’t, but I also acknowledge that I have expection bias here.  That’s why I’m far more interested in seeing evidence from Al than in performing a subjective listening test on a topic that, for reasons you listed a few posts back, is pretty well defined.


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## 71 dB

ALRAINBOW said:


> Just wait for my files it's obvious


How long do we need to wait? A day? A week? A month? 

It's funny how people with weird claims rarely have the evidence ready when making the claims. Most of the time the evidence never comes. I wonder why…


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## stonesfan129 (Dec 31, 2018)

Can we please hear the files?  Thanks.


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## ALRAINBOW

An open mind is nice it's about hearing it not a paper telling us it's perfect. He tracks has many fake samples I don't buy from them anymore. Yes some are ok but many are not. Of you take the time to analyze it's visually obvious with the noise on top.


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## ALRAINBOW (Dec 31, 2018)

I am out of ny it's New Year's Eve. I'll post it some time tomorrow I promise
My place 
Room built for purpose 20/9/55 ft
speakers cust infinity IRS V , new caps and LPS , magnets etc. 
mark levivson pre no 26 Amps no 33
mogami gold mic interconnects , new soon
digital three cust servers , win ser 2016 , AO 2.20 and cust Linux Kernal for NAS
player roon server , dac PC HQ player USB power cut at main board
network isolators cat 7a shielded , LPS for all pcs and network switches and routers. Audio network ips dedicated nas is music and network switch and router. Dacs lampi various


----------



## stonesfan129 (Dec 31, 2018)

.


----------



## ALRAINBOW

I have no dog in this argument it's real very real. Not some hey did you hear that bull. I know based on data it's ok but it's not true in sound 
Same is true for above 44.1 too but it's not noise floor it's layers so if this is also good enough by data it's also not a true value in life. I am 61 now soon to be 62. 
I will post my rooms f setup if ok.


----------



## 71 dB

ALRAINBOW said:


> An open mind is nice it's about hearing it not a paper telling us it's perfect. He tracks has many fake samples I don't buy from them anymore. Yes some are ok but many are not. Of you take the time to analyze it's visually obvious with the noise on top.



Fake samples? Have we moved from fake news to fake samples?

Visually? You listen to the music with your ears, not with your eyes. One of the most important things to learn about digital audio is that things don't sound the way the look. When you understand digital audio you know in what ways the visual representation is different. For example this _looks_ perfect 2000 Hz square wave sampled at 44100 Hz:




 

But it isn't! It sounds bad, not what 2000 Hz square wave should sound. In fact it is an "illegal signal." No real life signal correctly bandlimited and sampled at 44100 Hz gives this. Instead the correct 2000 Hz square wave sampled at 44100 Hz looks like this:



 

It _looks_ worse, but it _sounds_ better, because it's the correct bandlimited digitized version of analog squarewave signal. This is just one example of visual aspects telling you the wrong thing. To a person who knows the theory and secrets of digital audio the visual side can tell a lot, but it takes trained eyes to interpret things correctly.

CD audio isn't perfect techically (that would take near infinite sampling rate and perhaps 64 bits (?) of dynamic range to record molecules in the air hitting each other and ants farting in China, but it's good enough for human ears. In that sense it's perfect. The science tells it and numerous carefully done listening tests tell it backing up the science.


----------



## 71 dB

ALRAINBOW said:


> Same is true for above 44.1 too but it's not noise floor it's layers so if this is also good enough by data it's also not a true value in life. I am 61 now soon to be 62.



At your age 15 kHz is a real challenge, so in fact 32 kHz sampling rate would be enough! I just turned 48 and listening to stuff above 16-17 kHz is a thing of the past.


----------



## stonesfan129

Please post the files.  Thanks.


----------



## Steve999

My mind is a tabla rasa and I have expectoration bias. A cold will do that to you.


----------



## ALRAINBOW

Did you read my post of my room so I sound like I'm nuts hahaha happy new year


----------



## 71 dB

ALRAINBOW said:


> Did you read my post of my room so I sound like I'm nuts hahaha happy new year



Yes, but the post doesn't prove your claims about high resolution formats being better. It just shows that you have invested a lot of money to your sound gear. The problem is this: What you say indicates expectation bias, lack of deeper understanding of digital audio and lack of careful (double) blind listening tests to back up your claims. If you have believed for years that high resolution formats are superior, it might be difficult to let go such belief, so I understand your situation. You are desperate to prove "us" wrong so you can keep your beliefs, but it's not necessory to do such a thing. Whether you keep your beliefs and continue living in your delusions as you have done so far (not much harm done really) or you learn new things, get wiser and perhaps save a few bucks not paying extra for 24 bit high resolution files.


----------



## ALRAINBOW

Hahaha no man no dog in the fight. It's about proving your claim wrong based on databases papers that all. No disrespect in any way. 
Here is one is dsd 128 better then dsd 64 ? What's your view plesee no bull


----------



## stonesfan129

Please post the files.  If you won't post the files, post an AES paper or some kind of scientific study showing a difference.  Thanks.


----------



## bfreedma

Is it New Years Eve or April Fools day?

Al, you know you’re in the Sound Science subforum, right?  Just checking because you don’t seem to based on your posts.  The notion that AES papers and related research are all wrong and your subjective opinions are some sort of validation is comical.  As is your statement that you “don’t have a dog in this fight” - you clearly do based on your insistence that your investment in high res somehow equals audiblity.  It’s also interesting that you describe your room based on what’s in it down to power supplies and the OS on your computers (seriously? as if the operating system has any impact) but make no mention of room treatments and/or room eq solutions.


----------



## TheSonicTruth

ALRAINBOW said:


> I am out of ny it's New Year's Eve. I'll post it some time tomorrow I promise
> My place
> Room built for purpose 20/9/55 ft
> speakers cust infinity IRS V , new caps and LPS , magnets etc.
> ...



20x9x55'???  55' deep?
The economy class section of a 747?


----------



## castleofargh

ALRAINBOW said:


> by the comment your not trying  either it's amazing how much can be done on paper with no real-time hearing
> cheers to your ignorance in this.
> So on paper bits are bits right
> Jitter does not exist
> ...


he's reacting this way because you wouldn't be the first one to come here with total certainty and no fact. over the years, it takes more and more patience and willpower to receive overconfident people with no proof, because it usually turns out to be a waste of everybody's time and we have all become painfully aware of it.
I see that you plan to try and upload some samples, thank you for that. it at least suggests that you're honestly trying to demonstrate something and that's a nice change for us.
please upload short samples of songs instead of a full song so that you're not doing anything illegal by sharing it.and if you have issues with that, let us know and someone will surely offer to help. 




ALRAINBOW said:


> by the comment your not trying  either it's amazing how much can be done on paper with no real-time hearing


now that's just weird. do you really expect that people staying on audio forums for years and roaming this specific subsection, focused on blind listening and controlled data, would have never tried to compare various formats and resolutions? most people around here have done it many times casually and in blind tests(ABX or other). that's why several of us suggested to check that you're not talking about different mastering of the same album(something that will be answered once you share a sample of both tracks). because it's more likely to get an audible difference from remastered tracks, than it is to get one from a change in resolution between CD and something bigger. so obviously that's was the first suggestion, and the second was that something in your system isn't transparent when playing one of the formats. only when such obvious possibilities have been cleared out, can we start considering that some audible difference is coming from the resolution itself. it's a simple process of eliminating the potential causes.


----------



## stonesfan129

Still waiting for the files to be shared.


----------



## 71 dB

ALRAINBOW said:


> Hahaha no man no dog in the fight. It's about proving your claim wrong based on databases papers that all. No disrespect in any way.
> Here is one is dsd 128 better then dsd 64 ? What's your view plesee no bull



I don't claim deep knowledge of PDM encoding, but I think my knowledge of it makes it safe to say both DSD128 and DSD64 are capable of producing transparent audio for human ears. DSD128 might offer some marginal advantage (reduced distortion?) over DSD64 in the recording process that I am unware of, but I think for human ears it doesn't make a difference. DSD data is the differential of PCM data. Integrating DSD gives PCM. That's why all you need to decode DSD bitstream to analog signal is an analog intergrator, a low pass filter. Anyway, this thread is about PCM format and specifically the significance of bit depth.


----------



## gregorio (Jan 1, 2019)

ALRAINBOW said:


> [1] by the comment your not trying either it's amazing how much can be done on paper with no real-time hearing ... [1a] cheers to your ignorance in this.
> [2] So on paper bits are bits right. [2a] Jitter does not exist.
> [2b] Flabby bass or schilling highs it's the System or recording. [2c] Such little actual knowing and so much more on a parrots level and repeat.



1. Yes it is. Digital audio was invented "on paper", one specific paper to be precise. However, you need to be careful here, there's an old English saying, "those who live in glass houses shouldn't throw stones". How much "real-time" hearing/listening do you have? Almost certainly less than me or some others here!
1a. No, cheers for yours! Again, you need to be careful, you don't know who you're talking to, some of the people here know far more than you and calling others ignorant when you are the one with far less knowledge will just make you look foolishness!

2. On paper AND in a digital audio system.
2a. Where did you get that from? Jitter always exists but today's technology reduces jitter to such low levels that amps/speakers cannot reproduce it and even very cheap modern DACs can do this. If you have a jitter issue that's actually audible, then as already stated, you must have an incredibly poor/faulty system.
2b. Correct, it must be either the system or the recording. If you're getting a flabby bass or shilling highs on all your recordings, then either you ONLY listen to poor examples of certain music genres or you have a poor system (or a good system very poorly setup).
2c. You are the one "parroting", you're parroting audiophile myths and clearly have little actual knowledge. Did you read the OP? If so, what part of it didn't you understand?


ALRAINBOW said:


> Any same track at 24 shows a much lower noise floor ...


Sure, under certain circumstances some tracks will show a lower noise floor with 24bit when viewed on a spectral analysis. However, you should take your own advice and try listening; the noise floor of 16bit is below the noise floor of almost all recordings, below audibility at any reasonable listening level and your system cannot reproduce anywhere near 24bits anyway. Did you not read the OP?


ALRAINBOW said:


> If I posted it it's real try it or shut up about how it can be.


Firstly, you obviously have little idea of what's real and what isn't, if you did then you wouldn't be listening to music recordings in the first place! And secondly, the "glass houses" saying applies again, some of us HAVE tried it, many more times, for many more years and under more stringent conditions than you. So you're making yourself look foolish as in comparison YOU haven't tried it and you should "shut up"!


ALRAINBOW said:


> [1] Did you read my post of my room so I sound like I'm nuts hahaha
> [2] An open mind is nice it's about hearing it not a paper telling us it's perfect.


1. Yes, unfortunately you do sound like you're nuts. Why would you built a room with such poor dimensions for sound reproduction?

2. Firstly, an open mind is nice but only about certain things. Surely it's only "nice" to have an open mind about things that are not already quite certain? For example, is it it "nice" to have an open mind that the Earth is flat or that gravity doesn't exist? Secondly, having an open mind should surely apply to you too, shouldn't it? Shouldn't you have an open mind to well known and demonstrated facts, even if they conflict with your beliefs?

Time and again we have audiophiles come here, try to tell us what's real and what isn't and that they have better hearing and better equipment than us, ALL of which is FALSE! I am used to a system that's almost certainly far better than yours, I have trained listening skills and I listen to recordings far more than you do (and so do ALL my colleagues), so those parts of your typical audiophile argument are false! As to what's "real", there are people here whose job is to make the unreal seem real (to human hearing perception), so who is the person demonstrating ignorance here?

G


----------



## stonesfan129 (Jan 1, 2019)

We are still waiting for the files to be posted.  This is just getting silly.  Please post the files or stop claiming to hear a difference.


----------



## Slaphead

stonesfan129 said:


> We are still waiting for the files to be posted.  This is just getting silly.  Please post the files or stop claiming to hear a difference.



I’ve been reading this thread development with anticipation, however I fear that the relevant files will never appear.

There’s no doubt that Mr Rainbow’s system is impressive - certainly able to reproduce music with a large dynamic range. However assuming the electronics are pretty much the best possible then we’re talking only 19 to 20 bits of dynamic range, not the 24 bits Mr Rainbow is talking about. And then we have the transducers and they’re going to knock the dynamic range back quite a bit.

The other thing to consider is that Mr Rainbow’s system cost an arm and a leg and probably another limb as well, so there is almost certainly a large expectation bias coming into play as well.


----------



## bfreedma

Slaphead said:


> I’ve been reading this thread development with anticipation, however I fear that the relevant files will never appear.
> 
> There’s no doubt that Mr Rainbow’s system is impressive - certainly able to reproduce music with a large dynamic range. However assuming the electronics are pretty much the best possible then we’re talking only 19 to 20 bits of dynamic range, not the 24 bits Mr Rainbow is talking about. And then we have the transducers and they’re going to knock the dynamic range back quite a bit.
> 
> The other thing to consider is that Mr Rainbow’s system cost an arm and a leg and probably another limb as well, so there is almost certainly a large expectation bias coming into play as well.




It's a nice list of gear, but without an in room response plot (and other measurements), who knows what it sounds like.  I've seen plenty of expensive systems in bad rooms and/or poorly configured and EQed which are outperformed by systems costing a fraction of the price.


----------



## Slaphead

bfreedma said:


> It's a nice list of gear, but without an in room response plot (and other measurements), who knows what it sounds like.  *I've seen plenty of expensive systems in bad rooms* and/or poorly configured and EQed which are outperformed by systems costing a fraction of the price.



And that's why I don't buy expensive systems - I simply don't have the house for them . For speaker listening I've always bought pro audio monitors for nearfield listening in the hope that my proximity to the speakers counters the room acoustics to some degree - so far I'm happy.


----------



## sonitus mirus

Slaphead said:


> And that's why I don't buy expensive systems - I simply don't have the house for them . For speaker listening I've always bought pro audio monitors for nearfield listening in the hope that my proximity to the speakers counters the room acoustics to some degree - so far I'm happy.



I got the calibrated measurement mic out earlier today to see if those nasty SBIR cancellations magically disappeared with the start of the new year.  No such luck.  

Sadly, I can only move the dips slightly up or down in frequency with speaker placement, and the frequencies impacted are generally much too low for typical and affordable room treatment to do much of anything to help.


----------



## 71 dB

It's funny how telling high resolution formats do not offer sonic advantage over normal redbook is being ignorant. No, it's the other way around. When you don't know anything you go with the common sense which tells "more is better". I studied digital audio in university and even that level of knowledge and understanding made it a "process" for me to gradually accept the fact that in consumer audio high resolution formats don't have anything to offer (SACD offers multichannel support, but that's a different issue). That's how strong the "more is better" mentality is in us. More is better to a point. 24 bits is clearly better than 8 bits, but in consumer audio it's not better than 16 bits, because the point of "good enough" is around 13 bits.


----------



## bigshot

ALRAINBOW said:


> So on paper bits are bits right
> Jitter does not exist
> Flabby bass or schilling highs it's the System or recording. Such little actual knowing and so much more on a parrots level and repeat. If I posted it it's real try it or shut up about how it can be.
> I will upload soon as I can to push for anyone who can take the time to hear it.



Two files with the same bits should sound the same. If they don't, it is probably an error in testing procedure.
Jitter in the levels it occurs in even the cheapest home audio components is below the threshold of audibility.
Response imbalances are created by transducer error. You correct it by EQ and if that doesn't work, you get better transducers.
I'm happy to do a test to check the accuracy of your findings. Just let me know.


----------



## bigshot

Slaphead said:


> There’s no doubt that Mr Rainbow’s system is impressive - certainly able to reproduce music with a large dynamic range. The other thing to consider is that Mr Rainbow’s system cost an arm and a leg and probably another limb as well.



I don't think there's any reason to assume either of those things. If he actually had a very good system, he might know more about how digital audio works. This seems to have less to do with audio fidelity than it does argumentative thread crapping. Some people believe audiophile sales pitch without any personal knowledge at all. Who knows what kind of system he has? And ultimately, it doesn't matter because I think he's just here to rattle the bars on our cage. People who take positions like this in Sound Science talk through their hats and use bluff and bluster and obfuscation to make their points. I give them a chance and if they don't pony up to prove their point, I dismiss them and don't look back. They just want attention. I don't have time to feed their delusional ego.


----------



## ILikeMusic (Jan 1, 2019)

bigshot said:


> And ultimately, it doesn't matter because I think he's just here to rattle the bars on our cage. People who take positions like this in Sound Science talk through their hats and use bluff and bluster and obfuscation to make their points. I give them a chance and if they don't pony up to prove their point, I dismiss them and don't look back.



This. It's a simple fact of physics that in any practical instance no one is going to hear, casually or otherwise, the difference between a properly-mastered recording of 16 and 24-bit depth, nor is anyone going to produce magical test tracks that demonstrate otherwise. I don't know if the poster is trolling or simply and honestly confused as to the _actual_ cause of what he thinks he's hearing, but it's clear no one is going to be able to convince him of anything so why are we all contributing to crapping on the thread for multiple pages? Maybe let's give it a rest.


----------



## StandsOnFeet

ILikeMusic said:


> it's clear no one is going to be able to convince him of anything


I think you nailed it. We tend to get the burden of proof mixed up. We're going to ABX the tracks the OP sends us, we'll all report that we can't tell them apart, and he won't believe us. He'll claim that his ears or equipment are better than ours, and we'll be no further ahead.

He's the one with the claim that goes against basic physics, so the burden of proof is _his_: he has to demonstrate that he can tell them apart in a _blind _test.


----------



## stonesfan129

Can we hear the files?


----------



## old tech (Jan 2, 2019)

Slaphead said:


> I’ve been reading this thread development with anticipation, however *I fear that the relevant files will never appear*.


Unfortunately they never do.

I have over the years read a few posts like this and the promised files never eventuate while the poster disappears.

I sometimes wonder what makes these people tick, coming into a sound science forum making anti sound science claims and then displaying cowardice, not having the balls to back it up.


----------



## gregorio (Jan 2, 2019)

Slaphead said:


> There’s no doubt that Mr Rainbow’s system is impressive - certainly able to reproduce music with a large dynamic range. However assuming the electronics are pretty much the best possible then we’re talking only 19 to 20 bits of dynamic range, not the 24 bits Mr Rainbow is talking about. And then we have the transducers and they’re going to knock the dynamic range back quite a bit.



And, in addition to that we've got the noise floor of the listening environment! Even with treatment, the chances are that his room has a noise floor of 30dB or more. 120dB dynamic range (CD with noise-shaping) above that would give us a peak output of about 150dB, no speakers at any price can produce 150dB peak levels AND 120dB dynamic range and even if such speakers did exist, you couldn't listen to them at those levels without serious hearing damage.



bfreedma said:


> I've seen plenty of expensive systems in bad rooms and/or poorly configured and EQed which are outperformed by systems costing a fraction of the price.



Me too. Worst I saw was nearly $200k of kit which could have been outperformed with a budget of about $5k. So often I see audiophiles doing the equivalent of spending a fortune trying to fix a paint flaw on the trunk of their car, which can only be seen with a magnifying glass, while ignoring the fact that the front of their car has been flattened by a tank!



Slaphead said:


> For speaker listening I've always bought pro audio monitors for nearfield listening in the hope that my proximity to the speakers counters the room acoustics to some degree - so far I'm happy.



Ideally, you don't want to counter the room acoustics, you still definitely want room acoustics but neutral room acoustics. Pro audio nearfields are therefore not a great solution and although all top class commercial studios have nearfields, they also always have a very high quality mid/far field monitoring system. Consumers are relatively limited though, with both a limited budget and the fact that their listening room typically has to be either a multi-purpose room or a dedicated but very small room, either of which limits the amount/effectiveness of any acoustic treatment. For the serious listener, pro audio nearfields can often represent *the best that can be achieved within the given limitations*. Most likely you've made a wise choice, which is more of a compliment than it sounds because the "serious listener" marketplace effectively does all it can to steer consumers towards unwise choices! Having said this, I've often seen consumers using nearfield monitors completely incorrectly, for example placed very near walls and/or with the LP at double or triple the distance of what constitutes "nearfield".

Apparently though, assuming @ALRAINBOW is being truthful, he is far less limited, both in terms of budget and the room. Why on earth he'd custom build a room with such poor dimensions is a mystery though, the ceiling height relative to the length and width of the room is an acoustic problem that can't be overcome. Double that height would have been appropriate or if that was outside the budget, then a much smaller room would have been preferable, at least half the length for example. While we don't know all the details, it appears obvious that he has NOT done "the best that can be achieved within the [his] given limitations"!

G


----------



## bigshot

old tech said:


> I sometimes wonder what makes these people tick, coming into a sound science forum making anti sound science claims and then displaying cowardice, not having the balls to back it up.



I think they assume that Sound Science is the same as any other audiophile forum. Most places you can claim things that are totally wrong and vehemently defend them. If you get huffy enough and post on it enough times, you "win". That doesn't work as well in Sound Science. Here, that sort of behavior elicits a dog pile of people demanding proof. By the time the poor sap realizes his mistake, he's in too deep so all he can do is go out in a blaze of glory. "You want proof! I'll go get you your proof and then you'll see!" Then they sneak out the back door and never come back. I don't get all worked up over these kinds of people any more. It isn't worth my time.


----------



## ILikeMusic

bigshot said:


> I think they assume that Sound Science is the same as any other audiophile forum. Most places you can claim things that are totally wrong and vehemently defend them. If you get huffy enough and post on it enough times, you "win". That doesn't work as well in Sound Science. Here, that sort of behavior elicits a dog pile of people demanding proof. By the time the poor sap realizes his mistake, he's in too deep so all he can do is go out in a blaze of glory. "You want proof! I'll go get you your proof and then you'll see!" Then they sneak out the back door and never come back. I don't get all worked up over these kinds of people any more. It isn't worth my time.



Agree. And one of the reasons the only subforum I bother to read anymore is Sound Science. In the other sections people are allowed to bloviate endlessly debating utter nonsense, in a 'DBT free' (What?) environment no less. This is the about the only place where facts matter and facts are apparently a very high bar in audio discussions. Pity the soul who comes here without them, and that's the way it should be.


----------



## Slaphead

gregorio said:


> Having said this, I've often seen consumers using nearfield monitors completely incorrectly, for example placed very near walls and/or with the LP at double or triple the distance of what constitutes "nearfield".



LOL, if anything I'm too nearfield for a lot of my listening - the monitors are about a metre apart, and depending on what I'm doing I'm between 0.75 to 1.25 metres distant from them - they're 0.5 metres away from the rear wall, but that's as far away as they're going get. I live in a flat (apartment), which although very sizeable, means I can't really choose my room configuration, and I also have to take into account the fact that my neighbours may not enjoy my particular taste in music in a mid/far field application driven to my preferential SPL - which I may add is not that loud, but I can do it less annoyingly for them in a nearfield configuration and not really feel like I'm losing out.


----------



## gregorio

Slaphead said:


> [1] LOL, if anything I'm too nearfield for a lot of my listening - the monitors are about a metre apart, and depending on what I'm doing I'm between 0.75 to 1.25 metres distant from them
> [2] they're 0.5 metres away from the rear wall, but that's as far away as they're going get.



1. Nope, that's about right, although 1m - 1.25m would be ideal.  
2. More would be better but I've seen a lot worse! They should really be at least 1m from any wall and preferably more, the idea being that the direct sound from the nearfields is many times higher than the reflections, thereby rendering the room acoustics effectively a non-issue. Ideally therefore, they'd be fairly near the centre of the room, although that's highly impractical in most consumer situations of course. I'm sure you're already aware of all this but it might be useful for others.

G


----------



## stonesfan129

So what did this guy just flake?  No files I guess.  Welp I'm sticking to my previous conclusion that CD quality and even 256k AAC is good enough for my needs.


----------



## bigshot

AAC 256 is good enough for anyone with human ears.


----------



## Slaphead

bigshot said:


> AAC 256 is good enough for anyone with human ears.



I have inhuman ears - I get a lot of stuff from my favourite Dub Techno/Drum & Bass DJs in... wait for it... AAC HE encoding which averages around 70Kbps, and I've no complaints, at least not in the environment that I listen to that stuff which is generally on my commute.


----------



## stonesfan129

I'm fine with iTunes downloads.  Many times these are "Mastered For iTunes" which is taken from the exact same 24-bit master as what they sell on sites like HDtracks.  And many times these 256k AAC tracks sound better to me than some of the older CD versions due to being a better mastering.  But it's really on a case-by-case basis.


----------



## Glmoneydawg

Slaphead said:


> LOL, if anything I'm too nearfield for a lot of my listening - the monitors are about a metre apart, and depending on what I'm doing I'm between 0.75 to 1.25 metres distant from them - they're 0.5 metres away from the rear wall, but that's as far away as they're going get. I live in a flat (apartment), which although very sizeable, means I can't really choose my room configuration, and I also have to take into account the fact that my neighbours may not enjoy my particular taste in music in a mid/far field application driven to my preferential SPL - which I may add is not that loud, but I can do it less annoyingly for them in a nearfield configuration and not really feel like I'm losing out.


Yep....my music room is about 1500 sq feet,but my speakers and i form about a 9ft equilateral triangle.I think nearfield takes some of the room acoustics out of the equation and takes away the need for excessive volume.


----------



## castleofargh

when I think about how big my "listening" room is, I remember this :


----------



## Glmoneydawg

castleofargh said:


> when I think about how big my "listening" room is, I remember this :


If thats Columbo....and I'm pretty sure it is...you are MUCH older than i thought....it's ok ..comforting to know i'm being moderated by someone of similar vintage


----------



## stonesfan129

Next thing you know, people will be saying 24 bits isn't enough, we need to have 32-bit music.


----------



## 71 dB

stonesfan129 said:


> Next thing you know, people will be saying 24 bits isn't enough, we need to have 32-bit music.


With shaped dither that's 200 dB of dynamic range! Rocket lauches and heartbeats at natural levels in the same recording! With efficient horn speakers all it takes is a few gigawatts of power to play such recordings and kill humans and animals within 100 yards. Yeah, definitelty 32 bit music is needed!


----------



## stonesfan129

71 dB said:


> With shaped dither that's 200 dB of dynamic range! Rocket lauches and heartbeats at natural levels in the same recording! With efficient horn speakers all it takes is a few gigawatts of power to play such recordings and kill humans and animals within 100 yards. Yeah, definitelty 32 bit music is needed!



It's just silly how people who make these outlandish claims that anything beyond CD quality makes an audible difference can never prove it.


----------



## miksu8 (Jan 14, 2019)

In theory 16 bits is enough, but what about real world? Bad 16bit transfer from 24bit master, due to incompetence or intentionally, to get HD versions sold. Anyone done blind tests or analyses?

Edit: Sorry looks like this is discussed here already.


----------



## castleofargh

miksu8 said:


> In theory 16 bits is enough, but what about real world? Bad 16bit transfer from 24bit master, due to incompetence or intentionally, to get HD versions sold. Anyone done blind tests or analyses?
> 
> Edit: Sorry looks like this is discussed here already.


to go from a 24bit file to a 16bit file you have nothing to do but decide if and which type of dither you wish to use(a choice that under most conditions you will not notice as sounding any different). anytime a hires version is made to sound different from the CD release, of course it is intentional. be it malpractice or simply that they created a different master because somebody asked for it.


----------



## Slaphead

71 dB said:


> *With shaped dither that's 200 dB of dynamic range!* Rocket lauches and heartbeats at natural levels in the same recording! With efficient horn speakers all it takes is a few gigawatts of power to play such recordings and kill humans and animals within 100 yards. Yeah, definitelty 32 bit music is needed!



32 bit probably wouldn't need dither - around 194 dB peak is the maximum an undistorted sound wave can theoretically have in the Earth's atmosphere, and 32 bit already gives you 192 dB - not counting intersample peaks.


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## bigshot (Jan 14, 2019)

You can also puzzle out the numbers and see that 16 is plenty. With dither, a CD can do 90dB of dynamic range. Your listening conditions probably have a noise floor of above 30dB, so to hear the full range of a CD, you would have to boost the level of the quietest sound above that noise floor, bringing the peaks to at least 120dB. Coincidentally, 120dB is the threshold of pain and listening to sound that loud can cause hearing damage. I truth, 12 bit sound is probably enough. For more info see the article in my sig called CD Sound Is All You Need.


----------



## 71 dB

Slaphead said:


> 32 bit probably wouldn't need dither - around 194 dB peak is the maximum an undistorted sound wave can theoretically have in the Earth's atmosphere, and 32 bit already gives you 192 dB - not counting intersample peaks.



Even below 194 dB we start to see the effects on non-linear acoustics at levels of ~160 dB.


----------



## 71 dB

bigshot said:


> 1. You can also puzzle out the numbers and see that 16 is plenty.
> 
> 2. With dither, a CD can do 90dB of dynamic range.
> 
> 3. Your listening conditions probably have a noise floor of above 30dB, so to hear the full range of a CD, you would have to boost the level of the quietest sound above that noise floor, bringing the peaks to at least 120dB. Coincidentally, 120dB is the threshold of pain and listening to sound that loud can cause hearing damage. I truth, 12 bit sound is probably enough.



1. Not plenty enough for those who think more is (always) better. 16 bits is plenty for those who understand digital audio or believe those who know this stuff.

2. TPDF dither gives 95 dB of theoretical dynamic range, 3 dB less than just truncation error without dither (98 dB), but for sacrifying 3 dB of the dynamic range we get rid of distortion in the signal (we have total linearity). Using shaped dither we can have 20 dB (!) more _perceptual_ dynamic range. Signals decay into the noise floor the way they do in analog audio until completely masked by the dither noise (but of course you need CRAZY volume settings to hear quiet things at signal level -110…-120 dBFS).

3. Noise floor at 30 dB and peaks of music at 110 dB or less (sane listening that doesn't make you lose your hearing) means 80 dB or less of dynamic range needed. That translates into 13 bits, but for "DR6 pop" of today 8 bits with shaped dither would be just fine! By comparison, vinyl audio has "10 bits" worth of dynamic range at best.


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## old tech (Feb 5, 2019)

Ian Sheppard has posted a very good video, consistent with the OP.

His demonstration that the only difference between dithered 8bits and 24bits is noise, by reversing polarity, is pure genius.


----------



## gregorio

old tech said:


> His demonstration that the only difference between dithered 8bits and 24bits is noise, by reversing polarity, is pure genius.



To be fair, that reverse polarity test (called a "Null Test") is the first difference test taught to new audio engineering students and is used by almost all professional engineers on an almost daily basis. I've advocated it's use on numerous occasions here on head-fi, it's quick, easy, completely reliable, accurate, entirely objective and doesn't cost anything (using free software). It's hardly ever even mentioned in the audiophile world though and you're free to draw your own conclusions as to why!

G


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## old tech

gregorio said:


> To be fair, that reverse polarity test (called a "Null Test") is the first difference test taught to new audio engineering students and is used by almost all professional engineers on an almost daily basis. I've advocated it's use on numerous occasions here on head-fi, it's quick, easy, completely reliable, accurate, entirely objective and doesn't cost anything (using free software). It's hardly ever even mentioned in the audiophile world though and you're free to draw your own conclusions as to why!
> 
> G


Yes but using the null test, along with music samples, as a simple demonstration that 8bits has identical resolution to 24bits gets that message across very effectively.


----------



## Peter Hyatt

stonesfan129 said:


> It's just silly how people who make these outlandish claims that anything beyond CD quality makes an audible difference can never prove it.




That **** Qobuz marketing got me!


----------



## visanj

I purchased same song from itunes (MFiT) version, 24/96, 24/44 and 16/44 Flac from Qobuz, LAME MP3 from Google Play Music

What I noticed is, the quality differs based on the equipment we use

sound quality:
With Bluetooth (Oneplus Wireless 2 - Aptx HD)
24/44 > 16/44 > MFiT > 24/96 > MP3

With Brainwavz B200 + iBasso DC01 + Comply Audio Pro
24/44 > 24/96 > 16/44 > MFiT > MP3


In all cases, I feel 24/44 sounds better than even 24/96. Don't know why. 24/96 sounds as if it has some noise at higher frequencies (not clear)


----------



## gregorio

visanj said:


> [1] What I noticed is, the quality differs based on the equipment we use
> [1a] sound quality:
> With Bluetooth (Oneplus Wireless 2 - Aptx HD)
> 24/44 > 16/44 > MFiT > 24/96 > MP3
> ...



1. We have to be careful with statements like this. Have you ruled out the other possibilities? For example, are you certain they're all exactly the same master? As MFiT means "Mastered for iTunes", it's very possible they're slightly different masters.
1a. With the Oneplus, you're not really comparing 24/44, 16/44, 24/96 and MP3 you're comparing a lossy, 576kbps codec derived from those original sample rates/bit depths. The  AptxHD codec should be entirely transparent but again, we need to rule out the other possibilities before we can state that quality differences are due to something else (equipment differences).

Having mentioned these possibilities, it's most probable there is an audible difference between the equipment. It's unlikely the two different IEMs have the same frequency response and those differences are almost certain to be within the threshold of audibility. Additionally, it's also likely that your IEMs have different sensitivity and therefore the difference may not be a difference in quality but just a difference in volume.

2. It's possible there is some ultrasonic content (>21kHz) in the 24/96 version that is causing IMD (Inter-Modulation Distortion) in your amp sections or headphones (which obviously doesn't exist in the 24/44 version). It's also possible there is no audible difference and that what "you feel" is just a trick of your perception. There are other possibilities as well though, for example: Slightly different masters again, a slight volume change in the conversion process or even, that a resampling filter has been chosen that starts rolling-off at a relatively low frequency.

G


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## ALRAINBOW

Aside from pages of math,  at what point  Do we use our ears to allow for subjective listening? I can and most people with good ears can pick out 24 over 16 bit depth. I’m not Alone in this. I follow all the math and it’s conjecture with it but what do we hear ? If you don’t hear it there are two reasons 
One being the system used is subpar and can’t show what many can hear.
Second is your not noticing the change. 
to fill up pages of math and all of its own issues is futile.


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## castleofargh

ALRAINBOW said:


> Aside from pages of math, at what point Do we use our ears to allow for subjective listening? I can and most people with good ears can pick out 24 over 16 bit depth. I’m not Alone in this.


indeed you're not alone claiming to be able to. among you guys, those who produce evidence of being able to consistently pick up the 24bit file represent a handful at best. add a criteria where those people have enough control over their rig to ensure that the difference is caused by the bit depth and not some crap going on in the computer, DAC, or the masters being literally different ones, and now we're lucky if we can find 1 guy on the entire forum who can tell anything consistently at normal listening level and a few of his favorite albums.
that intriguing disparity seems to suggest that most people saying they can tell the difference are full of crap and/or too lazy to verify that they actually can do something before bragging about doing it all over the web.


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## ALRAINBOW

A well said reply really. I’m not special except maybe special Ed lol. But I do wonder is it hard. I have had many in my room to show and pretty fast its obvious to them. Now is my setup a causing  it is a good Question. but it’s not thst Simple I’m was a headphone guy and even up til now can use them or even good CIEMS ona good DAP.  But I do wonder what I’m hearing. 
to me it’s a blacker background and a seemingly lower noise in the track. Less air but still has presence. I thank you for not going right into bashing lol.  These forums are my first home in audio. 
thanks all.


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## bigshot (Nov 7, 2019)

I'd suggest getting an independent person who understands the concept to administer the test to you. I'd also check to see if you can tell the difference on different equipment. You may just be hearing some kind of artifacting going on. It might be worthwhile to take a 16 bit recording and bump it up to 24 to see if you're hearing a difference in the file or the recording. When you get unexpected results, that's when you want to test further- tightening controls, checking variables, getting independent verification...


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## 71 dB

ALRAINBOW said:


> I can and most people with good ears can pick out 24 over 16 bit depth.



Should be pretty much impossible in blind tests using reasonable listening levels (so that we are not listening to the 16 bit dither amplified so much you can hear it). If the master is the same, 24 bit and 16 bit should sound the same. Make yourself a properly dithered 16 bit version of a 24 bit file (so the master is the same).Use an audio editor to produce the difference of 16 bit and 24 files. That's the 16 bit dither, because proper dither removes quantization error completely and replaces it with dither noise. (noise that correlates with the signal is replaced with a little louder noise which doesn't correlate with the signal). Can you hear the this dither (difference signal) using reasonable listening levels? Should be pretty hard to hear anything, because the level of 16 bit dither noise without shaping is something like -95 dBFS and even if shaped dither can go up to -70 dBFS it's perceived quieter because of the properties of human hearing, shape of equal loudness levels. Even if you hear something (you should not), this is the ideal situation to hear the difference of 24 bit and 16 bit, because there is nothing else, just the difference. When you listen to a 16 bit file, the signal itself masks the dither making it totally impossible to hear in any listening scenario that makes sense.



ALRAINBOW said:


> I’m not Alone in this. I follow all the math and it’s conjecture with it but what do we hear ? If you don’t hear it there are two reasons One being the system used is subpar and can’t show what many can hear. Second is your not noticing the change. to fill up pages of math and all of its own issues is futile.



The "critical limit" of bit depth in digital audio is about 13 bits, about 20 dB more dynamic range than vinyl at best. That's how much dynamic range you want in consumer audio and what you need to serve all reasonable listening scenarios. 16 bit audio gives you 3 bit (18 dB) worth of safety margin over the critical limit so anyone who understands digital audio can sleep well at night knowing their 16 bit music has enough dynamic range (well, the recording itself may not have, but the media certainly has!). Placebo-effect plays a huge role in audio. You have also the issue of different masters for different bit depths and even your hardware can handle different file formats differently. A 24/44.1 kHz version may sound the best becasue it is the best master, but it sounds just as good when converted properly into 16 bit and played on gear that treats 24 bit and 16 bit the same way and the listener doesn't know about the bits so that placebo effect can't function. Then we are in a situation the math describes.


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## gregorio

ALRAINBOW said:


> [1] Aside from pages of math, at what point Do we use our ears to allow for subjective listening?
> [2] I can and most people with good ears can pick out 24 over 16 bit depth. I’m not Alone in this.
> [3] I follow all the math and it’s conjecture with it but what do we hear ?
> [4] If you don’t hear it there are two reasons: One being the system used is subpar and can’t show what many can hear. Second is your not noticing the change.
> [5] to fill up pages of math and all of its own issues is futile.



1. At the point that the reproduced sound enters your ears.

2. This is the Sound Science sub-forum. If you are going to make factual assertions, they either need to agree with the science or contradict/disagree with it but ONLY if you have reliable evidence to support your assertion. Unfortunately, your assertion both contradicts the demonstrated science and you have presented no reliable supporting evidence! The demonstrated science shows that at normal/reasonable listening levels no one, good ears or not, can "pick out 24 over 16 bit depth" with commercial (competently made) music audio products. Unless, you can provide some reliable evidence to the contrary, the ONLY logical conclusion is that you arrived at this assertion simply because it has been suggested to you (in false marketing for example) or you have performed some seriously flawed test yourself. Furthermore, this conclusion has been demonstrated countless times: Audiophiles often claim they can hear the difference but without exception, on those occasions when they "put their money where their mouth is" and are reliably tested, they can't. Most commonly, it turns out they were actually comparing different masters and falsely attributing the audible difference to 24 vs 16bit.

3. This statement is also false unfortunately. The math was conjecture over 90 years ago but over 70 years ago was proven (by Claude Shannon) and therefore was no longer a conjecture. Furthermore, the entire digital age depends on that proof and it it were wrong, the digital age would not exist.

4. This too is a common falsehood peddled by some/many audiophiles. It is easily disproven by the fact that the very best systems (top commercial studio systems) and highly trained, experienced engineers can't hear the difference (given the conditions above). How then is it possible that untrained amateurs, using audiophile systems that are subpar (compared to top pro studios) are hearing a difference?

5. It would be futile if both: A. The math wasn't proven and demonstrated in practice in every digital device on the planet and B. Someone chooses to believe the marketing hype and audiophile myth and dismiss the demonstrated science. This isn't the "marketing hype and audiophile myth" sub-forum though, it's the sound science sub-forum and therefore it isn't futile, except of course to those who come here by error, thinking it is the "marketing hype and audiophile myth" forum!!


ALRAINBOW said:


> But I do wonder what I’m hearing ... I thank you for not going right into bashing lol.


Don't you think you should find out what you're hearing BEFORE making the assertion (in a sound Science forum) that what you're hearing is a difference between 24bit and 16bit AND (falsely) "bashing" others' equipment and/or listening skills?

G


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## ALRAINBOW

I’m not bashing and my comment on conjecture is based on math and then hearing it. If this thread is purely math then why is it a myth exploded lol. 
even after your comment on it being a fact for 70 years. Geezzz just play and comment then post of math to comment on what’s heard !


----------



## ALRAINBOW

gregorio said:


> 1. At the point that the reproduced sound enters your ears.
> 
> 2. This is the Sound Science sub-forum. If you are going to make factual assertions, they either need to agree with the science or contradict/disagree with it but ONLY if you have reliable evidence to support your assertion. Unfortunately, your assertion both contradicts the demonstrated science and you have presented no reliable supporting evidence! The demonstrated science shows that at normal/reasonable listening levels no one, good ears or not, can "pick out 24 over 16 bit depth" with commercial (competently made) music audio products. Unless, you can provide some reliable evidence to the contrary, the ONLY logical conclusion is that you arrived at this assertion simply because it has been suggested to you (in false marketing for example) or you have performed some seriously flawed test yourself. Furthermore, this conclusion has been demonstrated countless times: Audiophiles often claim they can hear the difference but without exception, on those occasions when they "put their money where their mouth is" and are reliably tested, they can't. Most commonly, it turns out they were actually comparing different masters and falsely attributing the audible difference to 24 vs 16bit.
> 
> ...


I don’t need to know first I can easily discern them Hahahaha. But joking aside it’s easy and can be duplicated like math but math and it’s effects can’t be duplicated always


----------



## gregorio

ALRAINBOW said:


> [1] I’m not bashing and
> [2] my comment on conjecture is based on math and then hearing it.
> [3] If this thread is purely math then why is it a myth exploded lol. [3a] even after your comment on it being a fact for 70 years.
> [4] Geezzz just play and comment then post of math to comment on what’s heard !



1. Clearly you were "bashing", you stated that anyone who can't hear the difference had either subpar equipment or listening skills. You think maybe that stating others have subpar equipment or listening skills is a compliment?
2. No it's NOT, your comment is false, the math is not a conjecture, it's a proven theorem! 
3. Because many audiophiles believe the (false) myth that 24bit is audibly different to 16bit under the conditions mentioned.
3a. Yes, because many audiophiles don't know or don't understand the proven math and instead believe the marketing hype that contradicts it!
4. Why? The "math" dictates what is recorded and reproduced with 24bit vs 16bit, what you think you hear or "wonder what you're hearing" is not covered by the math, as I already effectively stated in my point #1 previously!


ALRAINBOW said:


> [1] I don’t need to know first I can easily discern them Hahahaha.
> [2] But joking aside it’s easy and can be duplicated like math but math and it’s effects can’t be duplicated always


 
1. You stated "_But I do wonder what I’m hearing_" and now you're stating you know exactly what you're hearing, which is it? Hahahaha.
2. Sure, it's trivially easy to play (for example) two different masters and discern a difference, that's the whole point! What would be the point of paying studio time and an engineer to make another different master which sounded exactly the same? Unless you can eliminate this (and other potential audible differences), then you cannot truthfully assert the difference you are hearing/perceiving is due to the difference in bit depth.

G


----------



## bfreedma

ALRAINBOW said:


> I don’t need to know first I can easily discern them Hahahaha. But joking aside it’s easy and can be duplicated like math but math and it’s effects can’t be duplicated always



Short answer:  In this forum, sighted subjective opinions are not an acceptable substitute for valid testing protocols.  As suggested before, try assessing the difference again ensuring the masters are the same and the test is properly proctored.


----------



## bigshot

ALRAINBOW said:


> I don’t need to know first I can easily discern them Hahahaha.



Well, your posts don't exactly inspire confidence in the truth of what you say... I think you don't know what you're talking about, and you're in the wrong forum to try to bluff your way through.


----------



## old tech

Ever wonder why it is necessary for the manufacturer to put a hi-res light or indicator on your playback device if there is a clear difference in sound between 16 and 24 bits?  Perhaps the visual is what you are hearing?


----------



## castleofargh

ALRAINBOW said:


> I don’t need to know first I can easily discern them Hahahaha. But joking aside it’s easy and can be duplicated like math but math and it’s effects can’t be duplicated always


me too, I just read the name of the file I'm going to play and instantly can tell that the 24/96 tag means hires. easy.


----------



## billbishere

Surprised to see this thread still going strong for 10 years .  I was recently trying to get a better grasp on 24bit since qobuzz is cheaper than tidal now


----------



## castleofargh

billbishere said:


> Surprised to see this thread still going strong for 10 years .  I was recently trying to get a better grasp on 24bit since qobuzz is cheaper than tidal now


the short summary goes like this: 
there is nothing wrong with 24bit files, but there are many issues with the beliefs triggered by reading "24bit" somewhere.


----------



## vatch

The real summary is that it 24 bit doesn't really improve on 16 for playback.


----------



## board

billbishere said:


> Surprised to see this thread still going strong for 10 years .  I was recently trying to get a better grasp on 24bit since qobuzz is cheaper than tidal now


Yes, ten years and the argument is still the same:
"I can easily hear a difference"
"No, you can't. You may believe you can't, but if you take a 24 bit file and convert it to 16 bit (instead of taking two different masters) and do a blind test you will see that you can't hear a difference."
"I don't need to do a blind test. I know what I heard. An amazing improvement I tell you!"
":-/"

The argument never stops. I honestly don't know why I even bother trying to help educate people who have no interest in becoming wiser.


----------



## chef8489

board said:


> Yes, ten years and the argument is still the same:
> "I can easily hear a difference"
> "No, you can't. You may believe you can't, but if you take a 24 bit file and convert it to 16 bit (instead of taking two different masters) and do a blind test you will see that you can't hear a difference."
> "I don't need to do a blind test. I know what I heard. An amazing improvement I tell you!"
> ...


That's because people dont want to believe more is not better. Same with the discussion about digital cables changing the sound.


----------



## ALRAINBOW

Last post of course we can if you have a setup to show it. 
I can tell flac from AIFF To but hey I’m crazy I guess. Even if I take a flac fie and convert or to AIFF it’s obvious but I’m an old deaf guy what do I know. Stop using math and listen for once


----------



## castleofargh

ALRAINBOW said:


> Last post of course we can if you have a setup to show it.
> I can tell flac from AIFF To but hey I’m crazy I guess. Even if I take a flac fie and convert or to AIFF it’s obvious but I’m an old deaf guy what do I know. Stop using math and listen for once


"Confidence is a feeling, which reflects the coherence of the information and the cognitive ease of processing it. It is wise to take admissions of uncertainty seriously, but declarations of high confidence mainly tell you that an individual has constructed a coherent story in his mind, not necessarily that the story is true."
Daniel Kahneman. Thinking, Fast and Slow

you clearly could learn a lot about yourself by reading that book. there's an audiobook version but I doubt they have it in hires.


----------



## board (Nov 25, 2019)

ALRAINBOW said:


> Last post of course we can if you have a setup to show it.
> I can tell flac from AIFF To but hey I’m crazy I guess. Even if I take a flac fie and convert or to AIFF it’s obvious but I’m an old deaf guy what do I know. Stop using math and listen for once


Please provide the logs of the ABX tests you've passed of files properly converted from 24 to 16 bits, made from the same master (not two different files from different sources) as well as ABX tests of files converted from FLAC to AIFF. Until you do, your claims haven't been backed up and are only talk.

I'll be a good sport and show you what I mean:
Here's one of my claims: I can tell certain phono preamps apart.
My proof:

Cambridge CP1 vs. Parasound Zphono:

foo_abx 2.0.2 report
foobar2000 v1.3.10
2016-07-03 09:39:16

File A: Fleetwood Mac Cambridge BØJET pickup - oprindelig justering.wav
SHA1: b0c9656b8820fe8ea94af52706316e627e6e2966
File B: Fleetwood Mac Parasound JUSTERET PICKUP.wav
SHA1: b79c8cdf744c1b6f6554af5d57559469f943195d

Output:
DS : Primær lyddriver
Crossfading: NO

09:39:16 : Test started.
09:43:21 : 01/01
09:43:51 : 02/02
09:44:33 : 03/03
09:45:09 : 04/04
09:45:56 : 05/05
09:46:35 : 06/06
09:47:14 : 07/07
09:47:51 : 08/08
09:48:17 : 09/09
09:49:44 : 10/10
09:50:21 : 11/11
09:50:40 : 12/12
09:51:13 : 13/13
09:51:54 : 14/14
09:52:29 : 15/15
09:52:40 : 16/16
09:52:40 : Test finished.

 ----------
Total: 16/16
Probability that you were guessing: 0.0%

 -- signature --
96dd782fe77de0830762d5e78d9a11dda41eb60e




Parasound Zphono vs. Lejonklou Gaia:


foo_abx 2.0 report
foobar2000 v1.3.7
2015-10-06 13:17:47

File A: side 1 UDDRAG.wav
SHA1: 9ccf3ce0d2a31dbb34a2aed114702eb5288d34db
File B: Burzum - Det som DÅRLIG side 1 UDDRAG.wav
SHA1: 4d8c084287e6b3f31d1a5af67e5e9662c701142f

Output:
DS : Primær lyddriver
Crossfading: NO

13:17:47 : Test started.
13:18:34 : 01/01
13:18:51 : 02/02
13:19:07 : 03/03
13:19:35 : 04/04
13:19:45 : 05/05
13:19:57 : 06/06
13:20:09 : 07/07
13:20:16 : 08/08
13:20:26 : 09/09
13:20:37 : 10/10
13:20:55 : 11/11
13:21:05 : 12/12
13:21:14 : 13/13
13:21:37 : 14/14
13:21:50 : 15/15
13:22:29 : 16/16
13:23:09 : 17/17
13:23:50 : 18/18
13:24:03 : 19/19
13:24:14 : 20/20
13:24:14 : Test finished.

 ----------
Total: 20/20
Probability that you were guessing: 0.0%

 -- signature --
f6ffb9de2b47498c7381a271ff9db90bf942d90a




Parasound Zphono vs. NAD PP-4:


foo_abx 2.0.2 report
foobar2000 v1.3.10
2016-10-13 18:12:03

File A: R.E.M. - Out of time - side 1 - EQ (shelf +1 dB ved 5 kHz) - 2.wav
SHA1: 23a0440def43f1c62fb030409bcfa5e9e492efb4
File B: REM - 2 EQ 1 (+1 dB ved 5 kHz) - volumen justeret til NAD + timestretch.wav
SHA1: 0fb5e4694d578ef95d2f3e2370b597586baf713e

Output:
DS : Primær lyddriver
Crossfading: YES

18:12:03 : Test started.
18:13:16 : 01/01
18:13:58 : 02/02
18:14:40 : 03/03
18:15:23 : 04/04
18:15:51 : 05/05
18:18:00 : 06/06
18:18:34 : 07/07
20:26:20 : 08/08
20:26:52 : 09/09
20:28:37 : 10/10
20:28:54 : 11/11
20:29:15 : 12/12
20:29:58 : 13/13
20:30:57 : 14/14
20:32:05 : 15/15
20:32:46 : 16/16
20:32:46 : Test finished.

 ----------
Total: 16/16
Probability that you were guessing: 0.0%

 -- signature --
d178d3bb69b4f0c12805b545e09660821cf40fc6




NAD PP-4 vs. Cambridge CP1:

foo_abx 2.0.2 report
foobar2000 v1.3.10
2017-04-16 12:58:28

File A: Beatles Sgt. Pepper NY CAMBRIDGE JUSTERET PICKUP - Titelnummer.wav
SHA1: 05f44088f182bd182bd0fe0ca2e6369e0c1cf3b3
File B: Beatles - Sgt. Peppers - TITELNUMMER.wav
SHA1: 5832ac7467a8765572df760f156116dd558898c9

Output:
DS : Primær lyddriver
Crossfading: NO

12:58:28 : Test started.
13:00:01 : 01/01
13:03:56 : 02/02
13:04:11 : 02/03
13:04:29 : 03/04
13:04:39 : 04/05
13:04:48 : 05/06
13:05:25 : 05/07
13:05:42 : 06/08
13:05:53 : 06/09
13:06:19 : 07/10
13:07:26 : 08/11
13:07:35 : 09/12
13:08:13 : 10/13
13:09:11 : 11/14
13:09:57 : 12/15
13:10:58 : 13/16
13:10:58 : Test finished.

 ----------
Total: 13/16
Probability that you were guessing: 1.1%

 -- signature --
0214dcadd6acef8e8b803e16b08149fc72cbf29a


As you can see, the last test I "passed" with much less confidence. It was easily the most difficult one.

Here's a completely different, but more relevant, claim:
I can't tell hi-rez (24/96) apart from standard CD resolution.
My proof from the AIX hi-res challenge in August 2018:



foo_abx 2.0 report
foobar2000 v1.3.7
2018-08-21 21:27:30

File A: Tune_1_A.wav
SHA1: a172a18acd31bde18e70254654eb3d6a62a98869
File B: Tune_1_B.wav
SHA1: 23140e6544890298339f2a1de731f972a0285283

Output:
DS : Højttalere (CA USB Audio)
Crossfading: YES

21:27:30 : Test started.
21:38:15 : 00/01
21:39:02 : 00/02
21:40:15 : 00/03
21:41:35 : 01/04
21:45:51 : 02/05
21:47:17 : 02/06
21:48:20 : 03/07
21:49:10 : 03/08
22:00:04 : 03/09
22:01:08 : 04/10
22:02:23 : 04/11
22:04:27 : 05/12
22:06:39 : 06/13
22:07:30 : 06/14
22:08:32 : 06/15
22:09:46 : 07/16
22:09:46 : Test finished.

 ----------
Total: 7/16
Probability that you were guessing: 77.3%

 -- signature --
215bc93fd1e452bd483b437ca134ee648c1c035a




and:


foo_abx 2.0 report
foobar2000 v1.3.7
2018-08-22 20:11:00

File A: Tune_3_A.wav
SHA1: e4db0c5771607dd8cf1a2b629cb7dcbff593ef37
File B: Tune_3_B.wav
SHA1: 990af2551ff3a9fa1a6af0ed5b1773705464866d

Output:
DS : Højttalere (CA USB Audio)
Crossfading: YES

20:11:00 : Test started.
20:15:29 : 00/01
20:17:03 : 00/02
20:18:47 : 00/03
20:20:18 : 00/04
20:22:19 : 01/05
20:23:38 : 02/06
20:24:58 : 03/07
20:26:42 : 04/08
20:28:12 : 05/09
20:29:28 : 06/10
20:31:01 : 06/11
20:32:19 : 06/12
20:33:36 : 06/13
20:35:45 : 06/14
20:36:48 : 06/15
20:39:45 : 07/16
20:39:45 : Test finished.

 ----------
Total: 7/16
Probability that you were guessing: 77.3%

 -- signature --
b38c3f450663f07a9cd2879b9ac225faf53cb198



And I'll double down on the last claim: I can't tell hi-res apart from standard CD resolution, and my claim is that so can NO ONE ELSE, as long as the test was level-matched and properly conducted (meaning, properly converted and files were time-aligned, enough trials (at least 10-12), etc.).
No one who claims to be able to hear the difference has provided proof like that, yet it would be so easy to provide, if the differences were really as "goddamn obvious" as these people claim. And I would be happy to provide many, many more passed ABX logs (as well as failed ones) to show that ABX tests are completely valid. I've even given my aunt and three girlfriends ABX tests, and they passed most of them with flying colours.

I'll also explain why I'm so confident that no one can reliably tell hi-res apart from CD resolution:

* A higher sampling rate ONLY means it contains content above around 22 kHz. In music, the musical content at those frequencies are at an extremely low volume level. Plus, human hearing in anything but teenagers tops out at maximum 20 kHz. I'm 38, and my hearing tops out around 17,5 kHz (I've checked).
* Higher bitrate, as has been demonstrated in this thread, only means an increased dynamic range - higher than 96 dB. People can't hear musical content that soft.


----------



## sander99

board said:


> but if you take a 24 bit file and convert it to 16 bit (instead of taking two different masters) and do a blind test you will see that you can't hear a difference.


One little precision that maybe you forgot: Best would be: after converting down to 16 bit, convert it up to 24 bit again, so that you compare two 24 bit files (in fact: completely identical format) so that there is no way something in the playback chain can handle the two files differently in any way. (One file: the original 24 bit file. The other file: a 24 bit file resulting from converting down, and then converting the 16 bit file up again to 24 bits.)


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## board

sander99 said:


> One little precision that maybe you forgot: Best would be: after converting down to 16 bit, convert it up to 24 bit again, so that you compare two 24 bit files (in fact: completely identical format) so that there is no way something in the playback chain can handle the two files differently in any way. (One file: the original 24 bit file. The other file: a 24 bit file resulting from converting down, and then converting the 16 bit file up again to 24 bits.)


Okay . That would probably be a better way of doing that, I agree .


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## castleofargh

board said:


> Okay . That would probably be a better way of doing that, I agree .


to be fair it just depends on what you test. not going back to both samples at the same resolution also lets you find out if your system does something to the signal it probably shouldn't(that's usually more significant with sample rate in my experience). if you fail to disprove the null hypothesis your way, you can be pretty confident that you won't pass with both samples at the same resolution. so that's taken care of ^_^.


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## board

castleofargh said:


> to be fair it just depends on what you test. not going back to both samples at the same resolution also lets you find out if your system does something to the signal it probably shouldn't(that's usually more significant with sample rate in my experience). if you fail to disprove the null hypothesis your way, you can be pretty confident that you won't pass with both samples at the same resolution. so that's taken care of ^_^.


Fo shizzle!


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## billbishere

i like more.


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## robo24

board said:


> Please provide the logs of the ABX tests you've passed of files properly converted from 24 to 16 bits, made from the same master (not two different files from different sources) as well as ABX tests of files converted from FLAC to AIFF. Until you do, your claims haven't been backed up and are only talk.
> 
> I'll be a good sport and show you what I mean:
> Here's one of my claims: I can tell certain phono preamps apart.
> ...


In general, is there some bit rate at when any and separately when most are able to distinguish lossy versions from either lossless CD or lossless 24/96 files? I feel like some older 128 kbps files I have, especially the mp3 ones just sound terrible, but I don't have other versions with which to compare them with.


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## bigshot

robo24 said:


> In general, is there some bit rate at when any and separately when most are able to distinguish lossy versions from either lossless CD or lossless 24/96 files? I feel like some older 128 kbps files I have, especially the mp3 ones just sound terrible, but I don't have other versions with which to compare them with.



It depends on the codec of the lossy file. In my experience Fraunhofer MP3 *almost* becomes transparent at 320, MP3 LAME is completely transparent at 320, and AAC is completely transparent at 256. But even at lower data rates it can still be transparent with some music, particularly if you use VBR.


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## bigshot

ALRAINBOW said:


> I can tell flac from AIFF To but hey I’m crazy I guess.



Not crazy. You just haven't done a properly controlled listening test to be able to know for sure or not.


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## CoryGillmore

I like FLAC files cause it makes me feel good!
I like hi-res files cause they make me feel good! 
I like fancy headphone cables cause they make me feel good too! 
*in my Waterboy voice*


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## bigshot

I feel good without even superfluous stuff!


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## ILikeMusic

robo24 said:


> In general, is there some bit rate at when any and separately when most are able to distinguish lossy versions from either lossless CD or lossless 24/96 files? I feel like some older 128 kbps files I have, especially the mp3 ones just sound terrible, but I don't have other versions with which to compare them with.


As bigshot noted the codec is important, but in general tests I've seen on Hydrogenaudio and elsewhere seem to indicate that LAME V2 is transparent to most (exceptions are uncommon) and LAME V0 is essentially transparent to all. Thus LAME 320 would be the same but V0 would be a little more space-efficient.


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## gregorio

ALRAINBOW said:


> [1] Last post of course we can if you have a setup to show it.
> [2] I can tell flac from AIFF To but hey I’m crazy I guess. Even if I take a flac fie and convert or to AIFF it’s obvious but I’m an old deaf guy what do I know.
> [3] Stop using math and listen for once



1. Why would I want/have a faulty setup?

2. I can tell a FLAC from an AIFF as well, it's not at all difficult, one has ".flac" at the end of the file name, the other has ".aiff"! And why would being an "old deaf guy" make any difference when reading a file name? An old blind guy, sure! If you're saying you can tell the difference from the sound alone, obviously that's impossible (unless you have a faulty system) because you obviously can't tell the difference between two things that are identical. There's a term for people who can achieve such a feat, it's called "Deluded". So actually, your guess is pretty accurate!

3. This statement raises a couple of OBVIOUS problems:
A. As digital audio IS effectively math, to what are we supposed to "listen for once"? It's like saying; "stop using wheels and drive a car for once" ... "crazy" indeed!
B. As some of us effectively listen for a living (and have done so for decades) and as you are obviously NOT listening yourself but are relying on your eyes, biases and ignorance, then: Condescension + hypocrisy + ignorance is no basis for assertions of fact anywhere, let alone in a sound SCIENCE forum!

By repeating the same ignorant assertions, in a sound science forum of all places, you are demonstrating that either you are trolling, that you want to publicise your ignorance, or both. Either way, your contributions are most unwelcome here!

G


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## ALRAINBOW

How old are you guys here full disclosure please. you all know what I’m saying yet play word games. If I take a simple redbook file and convert it from flac it’s obvious and I don’t have to look. In fact it was found out by accident for me. 
mad for 16 to 24 bit depth this is known it’s just you guys refuse to admit it’s existence.  you can’t take a redbook file and convert to high res the info is not there 
You can take a hi res and down sample. Oddly you Guys are the math don’t listen groupies so is it a trap you asked me this lol. At this point you guys now admit to my point you don’t listen and don’t have a setup to show it. now this is not enemy to be an offensive comment.


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## TheSonicTruth

board said:


> Higher bitrate, as has been demonstrated in this thread,
> only means an increased dynamic range - higher than 96 dB.
> People can't hear musical content that soft.



Bit-_depth_, that is.

That many people still confuse the two terms does remain, and education on the difference needs to be given.


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## board

TheSonicTruth said:


> Bit-_depth_, that is.
> 
> That many people still confuse the two terms does remain, and education on the difference needs to be given.


Sorry, that was a typo. I did, of course, know that it was supposed to be bit depth. Actually, when robo24 asked at what bitrate audio became transparent, I thought "why is he all of a sudden talking about bit-rate instead of bit-depth? We're talking about 16 vs. 24 bits, not mp3. So my mistake .


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## board (Nov 26, 2019)

ALRAINBOW said:


> How old are you guys here full disclosure please. you all know what I’m saying yet play word games. If I take a simple redbook file and convert it from flac it’s obvious and I don’t have to look. In fact it was found out by accident for me.
> mad for 16 to 24 bit depth this is known it’s just you guys refuse to admit it’s existence.  you can’t take a redbook file and convert to high res the info is not there
> You can take a hi res and down sample. Oddly you Guys are the math don’t listen groupies so is it a trap you asked me this lol. At this point you guys now admit to my point you don’t listen and don’t have a setup to show it. now this is not enemy to be an offensive comment.


I'm 38. I've checked my hearing with a frequency sweep that started at 22 kHz, and my hearing tops out at around 17.5 kHz. And yours?

And now, please provide the log of the blind test you've passed between 24 bit and the same file properly converted to 16 bit. This is the second time I've asked, and you still haven't provided the log - you just keep yakking away. So please provide it instead of keeping on talking.
But I actually also have a, seemingly irrelevant, question for you: Are you a native English speaker?


----------



## TheSonicTruth

board said:


> Sorry, that was a typo. I did, of course, know that it was supposed to be bit depth. Actually, when robo24 asked at what bitrate audio became transparent, I thought "why is he all of a sudden talking about bit-rate instead of bit-depth? We're talking about 16 vs. 24 bits, not mp3. So my mistake .



Not at all.  The subject of digital audio itself contains many confusing terms! lol


----------



## gregorio

ALRAINBOW said:


> [1] How old are you guys here full disclosure please. you all know what I’m saying yet play word games.
> [2] If I take a simple redbook file and convert it from flac it’s obvious and I don’t have to look. In fact it was found out by accident for me.
> [3] mad for 16 to 24 bit depth this is known it’s just you guys refuse to admit it’s existence.  you can’t take a redbook file and convert to high res the info is not there. You can take a hi res and down sample. Oddly you Guys are the math don’t listen groupies so is it a trap you asked me this lol.
> [4] At this point you guys now admit to my point you don’t listen and don’t have a setup to show it. now this is not enemy to be an offensive comment.



1. This is the Sound Science forum (how many times?), what has age got to do with anything? Either your assertions agree with the science or contradict it, in which case you're ignorant/deluded unless you have some reliable evidence to support your assertions, regardless of your age!! As you've provided no such reliable evidence then obviously you MUST be ignorant/deluded which you confirm by just repeating the same false/unsupported assertions and throwing in a bunch of insults. YOU are the one playing word games and in response to my accusation of hypocrisy, you reply with even more hypocrisy. Way to go!

2. What's really obvious is that what the DAC chip receives from a 16/44 wav (or aiff) or FLAC of that file is absolutely identical and therefore, by definition, you cannot possibly hear a difference because there isn't any! So either your system is quite seriously faulty or you're imagining a difference (where there is none), there is NO other option here!!

3. The "info" you lose going from 24bit to 16bit is ALL inaudible noise. Did you not read or understand ANY of the OP? How old are you?

4. Please explain how my statement "_some of us effectively listen for a living (and have done so for decades)_" is an admission that we don't listen. Clearly you're now resorting to even more ridiculous falsehoods! And, I freely admit that I do NOT have a setup which is so faulty that it actually creates an audible difference between two identical files!!

Honestly, how foolish do you want to make your self appear, surely there must be limit? 

G


----------



## ALRAINBOW

board said:


> I'm 38. I've checked my hearing with a frequency sweep that started at 22 kHz, and my hearing tops out at around 17.5 kHz. And yours?
> 
> And now, please provide the log of the blind test you've passed between 24 bit and the same file properly converted to 16 bit. This is the second time I've asked, and you still haven't provided the log - you just keep yakking away. So please provide it instead of keeping on talking.
> But I actually also have a, seemingly irrelevant, question for you: Are you a native English speaker?


Lol see it always gets to this level. Yes an America of ITALIAN ancestry. My hearing last tested is on par with you. But this is not about high freq hearing it’s about an observation of noise floor being lower on 24 bit depth. It’s so apparent that I don’t like red book much and for me 24/88.2 is best where bit depth helps noise and 88.2 is still not hi res that seems to soften the sound to me.  Now I’m not saying the noise floor is lower but the Perception of its is there. And I’ll bet most all here can hear it too. Out on some beats and and play same track sourced from a good hi res track. Use dB power amp or j river to downsample then listen. I own about 30 TB of music in many formats it’s how I came to notice it. 
My whole point is on perception not math.  I live in nyc queens maybe one of you can stop by my place to hear what I say.


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## bfreedma (Nov 26, 2019)

gregorio said:


> snip...
> 
> Honestly, how foolish do you want to make your self appear, *surely there must be limit*?
> 
> G




I wouldn't be so sure of that...

We go through this with Al once a year or so.  He's not going to perform a blind test and will just plow ahead based on deep belief in audiophile mythology.


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## ALRAINBOW

Bro I have done this many times. Stop by and video me ok then it’s your group to say it’s true but while at my place I’ll bet you can hear it too once you tune into it. enjoy guys and I do mean stop by my place this hobby is just too alone. I used to enjoy head fi meets from here long ago.


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## bfreedma

ALRAINBOW said:


> Bro I have done this many times. Stop by and video me ok then it’s your group to say it’s true but while at my place I’ll bet you can hear it too once you tune into it. enjoy guys and I do mean stop by my place this hobby is just too alone. I used to enjoy head fi meets from here long ago.



No, you apparently haven't, but it's simple.  There are plenty of instructions on how to create samples for a blind test and use the Foobar plugin to record the results.  Do that and publish them - no need for anyone to come to your location.

I've seen your other claims over the years about hearing differences in data storage topologies, needing to use Windows Server OS rather than desktop versions, cables, DACs, file formats, etc.  Sorry, not buying any of it.   Take them back to the rest of Head-fi where claims don't require actual supporting evidence.


----------



## ALRAINBOW

bfreedma said:


> No, you apparently haven't, but it's simple.  There are plenty of instructions on how to create samples for a blind test and use the Foobar plugin to record the results.  Do that and publish them - no need for anyone to come to your location.
> And it’s all true too stop by and let’s see what can be heard.
> 
> I've seen your other claims over the years about hearing differences in data storage topologies, needing to use Windows Server OS rather than desktop versions, cables, DACs, file formats, etc.  Sorry, not buying any of it.   Take them back to the rest of Head-fi where claims don't require actual supporting evidence.


----------



## gregorio (Nov 26, 2019)

ALRAINBOW said:


> Lol see it always gets to this level. ....
> Bro I have done this many times.



Yep, and clearly you've never stopped to think why "it always gets to this level", you just keep doing "this many times". Congrats, pretty much the definition of a troll!

If you ever do stop and wonder why "it always gets to this level" let me give you a hint, I stated: "_Either your assertions agree with the science or contradict it, in which case you're ignorant/deluded unless you have some reliable evidence to support your assertions, regardless of your age!! As you've provided no such reliable evidence then obviously you MUST be ignorant/deluded which you confirm by just repeating the same false/unsupported assertions_" - Unfortunately, you respond by yet again just repeating the same false/unsupported assertions and therefore, yet again confirming what you've already confirmed. Congrats again Bro!! But it's really not necessary, we're already convinced that you're ignorant/deluded, so you're just flogging a dead horse at this point!! 

@bfreedma - If he does have a limit of how foolish he wants to make himself look, he apparently hasn't reached it yet! 

G


----------



## board

ALRAINBOW said:


> Bro I have done this many times. Stop by and video me ok then it’s your group to say it’s true but while at my place I’ll bet you can hear it too once you tune into it. enjoy guys and I do mean stop by my place this hobby is just too alone. I used to enjoy head fi meets from here long ago.


And again, you just keep talking although I asked you to post the log of your passed blind test. I'm starting to think that if you actually had one, or were wiling to take the ABX test and post the log, you would have done that a looooong time ago.
So last chance: Post a log, or shut up!

But I am of course aware that you will choose a third option: Keep talking and don't post a log :-/.

And for the record: The reason I asked about your nationality is because I found your grammar difficult to understand, so I figured that if you were a non-native speaker it might explain why, since we can't expect non-native speakers to write perfect English. So it wasn't meant as an insult.
But also for the record: I'm actually from Denmark, and I've only spent six months in the US, a year and a half in the UK, two months in Australia and two months in New Zealand, and I would say that I write better English than you. And unfortunately (again, not meant as an insult, just an observation) it's difficult for me to take your claims seriously when you write like this + when you refuse to back up your claims, of course.


----------



## board

bfreedma said:


> I wouldn't be so sure of that...
> 
> We go through this with Al once a year or so.  He's not going to perform a blind test and will just plow ahead based on deep belief in audiophile mythology.


----------



## castleofargh

guys, stick to facts and avoid trolls and insults. you know, rules, Head-Fi, all that.
if like in this case, we're dealing with unsubstantiated claims and you really have no fact to discuss, well then that's it. don't forget the burden of proof and who's carrying it. @ALRAINBOW claims he hears the difference, allegedly with ease(which really puts the last nail in the coffin IMO). listening tests and research disagree pretty consistently, while he's presenting zero supporting evidence. the verdict is pretty straightforward IMO. extraordinary claims require extraordinary evidence that he doesn't have and probably doesn't plan to gather. which calls for my sig "*What is freely asserted is freely dismissed."*
case dismissed!


----------



## bigshot (Nov 26, 2019)

ALRAINBOW said:


> How old are you guys here full disclosure please. you all know what I’m saying yet play word games. If I take a simple redbook file and convert it from flac it’s obvious and I don’t have to look. In fact it was found out by accident for me.
> mad for 16 to 24 bit depth this is known it’s just you guys refuse to admit it’s existence.  you can’t take a redbook file and convert to high res the info is not there
> You can take a hi res and down sample. Oddly you Guys are the math don’t listen groupies so is it a trap you asked me this lol. At this point you guys now admit to my point you don’t listen and don’t have a setup to show it. now this is not enemy to be an offensive comment.



I am 60 years old. I'm not sure what difference that makes, but I will answer you politely and in perfect keeping with the rules of this forum...

If you are basing your discernment on anything but a line level matched, direct A/B switched, double blind test, I can say with almost perfect certainty that you are absolutely wrong.



ALRAINBOW said:


> Bro I have done this many times. Stop by and video me ok then it’s your group to say it’s true but while at my place I’ll bet you can hear it too once you tune into it. enjoy guys and I do mean stop by my place this hobby is just too alone. I used to enjoy head fi meets from here long ago.



You think you can hear a 16 bit noise floor in a hotel ballroom with hundreds of people in it? The noise floor of the room itself would be over 45dB I would guess, more likely in the 50s. Even with isolating cans it would be hard to screen all that room noise out. You set your volume level for the loudest sound you could possibly tolerate... let's say 120dB, the threshold of pain. The room tone bleeds through at around 30dB let's say. You're still not going to hear -96dB. And at that level, you would incur hearing damage if you listened longer than a few minutes. At a normal listening level of 80dB or so, even if you were in an anechoic chamber, you still couldn't hear -96. And I haven't even started talking about the dynamic limitations of human hearing yet... Sorry, but no way. You are wrong. You think you know more than you actually do.



TheSonicTruth said:


> Bit-_depth_, that is. That many people still confuse the two terms does remain, and education on the difference needs to be given.



Actually bit depth is directly related to bit rate. They both relate to the noise floor. The term that might be clearer when referring to overall file size/audio quality is data rate. That is the general term for indicating a higher resolution file- both bit rate and sampling rate.


----------



## TheSonicTruth (Nov 26, 2019)

bigshot said:


> Actually bit depth is directly related to bit rate. They both relate to the noise floor.
> The term that might be clearer when referring to overall file size/audio quality is
> data rate. That is the general term for indicating a higher resolution file- both
> bit rate and sampling rate.



I see it more straight-forward: Bit-depth(Y-Axis) concerns amplitude/dynamic range, and Bitrate(X-Axis) concerns amount of data, measured typically in kilobits-per-second.  This is what I tell members of the public, my clients, etc, when I hear Bit-depth and Bitrate used interchangeably.

Just like the unit of measure - feet: Bits can be used in both the vertical and horizontal axes.


----------



## bigshot

He said a higher bit rate means more dynamic range, which is correct because higher bit rate also means higher bit depth. No matter though.


----------



## TheSonicTruth (Nov 26, 2019)

bigshot said:


> He said a higher bit rate means more dynamic
> range, which is correct because higher bit rate also
> means higher bit depth. No matter though.



I strongly and vehemently disagree:

Bit rate(or data rate) and Bit depth describe _two different axes. _Bit rate is a _temporal_(time-based) attribute, X-axis,  and Bit Depth is an _amplitude-_based attribute, Y-axis.

This is Digital Audio 99, not even 101! smh..


----------



## sander99

TheSonicTruth said:


> I strongly and vehemently disagree:
> 
> Bit rate(or data rate) and Bit depth describe _two different axes. _Bit rate is a _temporal_(time-based) attribute, X-axis,  and Bit Depth is an _amplitude-_based attribute, Y-axis.
> 
> This is Digital Audio 99, not even 101! smh..


I think you are confused, maybe you confuse bit rate with sample rate?
Bit rate means how many bits per second, so that would be (at least) the product of a) bit-depth and b) sample rate and c) the number of channels (without data compression or data reduction, data compression and/or data reduction would bring the bit rate down of course).


----------



## TheSonicTruth (Nov 26, 2019)

sander99 said:


> I think you are confused, maybe you confuse bit rate with sample rate?
> Bit rate means how many bits per second, so that would be (at least)
> the product of a) bit-depth and b) sample rate and c) the number of
> channels (without data compression or data reduction, data compression
> and/or data reduction would bring the bit rate down of course).




Not at all.  'Rate' to me signifies the _frequency_ at which something occurs, again, in _time - X_ axis.  Kilobits-per-second, samples-per-second, hundreds of feet per second(if you're a Nascar driver lol!).  All measures of occurences per unit of TIME.  No confusion there.

'Depth', to me, equates _intensity, _or amplitude, in an audio sense, betweem full scale and the noise floor. Y-axis.


----------



## sander99

TheSonicTruth said:


> Not at all.  'Rate' to me signifies the _frequency_ at which something occurs, again, in _time - X_ axis.  Kilobits-per-second, samples-per-second, hundreds of feet per second(if you're a Nascar driver lol!).  All measures of occurences per unit of TIME.  No confusion there.


The meaning of bit rate is bits per second. So if you have 2 channels of uncompressed PCM audio with 48 kHz sampling frequency and 24 bits bit-depth then every second of audio needs 2 x 48000 x 24 bits, that is 2304000 bits. So that gives a bit rate of 2304 kbits/s. That is excluding extra data that sometimes is added to an audio stream.

Example from Wikipedia:


> *Bit rate*The audio bit rate for a _Red Book_ audio CD is 1,411,200 bits per second or 176,400 bytes per second; 2 channels × 44,100 samples per second per channel × 16 bits per sample. Audio data coming in from a CD is contained in sectors, each sector being 2,352 bytes, and with 75 sectors containing 1 second of audio. For comparison, the bit rate of a "1×" CD-ROM is defined as 2,048 bytes per sector × 75 sectors per second = 153,600 bytes per second. The remaining 304 bytes in a sector are used for additional data error correction.


----------



## ALRAINBOW

Oh one last thought how about you guys upload on a cloud share link some files and I’ll comment honestly on them ok. Just do aiff  and flac and 
16/24 bit depth ok I mean this. You guys are clever enough to make me not know the file info and my dacs don’t show it. Only my player ok. Lets have some fun as a group on this ok. 
I will learn I’m sure 
Good night


----------



## gregorio (Nov 27, 2019)

bigshot said:


> He said a higher bit rate means more dynamic range, which is correct because higher bit rate also means higher bit depth.





TheSonicTruth said:


> I strongly and vehemently disagree:





sander99 said:


> I think you are confused, maybe you confuse bit rate with sample rate?



In this instance, TheSonicTruth is correct and bigshot is wrong. CD for example, has a bit rate of 705,600 bits per sec (per channel), while SACD has a bit rate of 2,800,000 bits per sec (per channel). SACD has a far higher bit rate but a bit depth of only 1bit, while CD has a bit depth of 16bits. So, contrary to bigshot's assertion, higher bit rate does NOT also mean higher bit depth! Same with dynamic range, SACD only has a dynamic range of ~6dB (without noise-shaped dither), while CD has a dynamic range of ~96dB (without noise-shaped dither). So again, SACD has a far higher bit rate than CD but a far lower dynamic range.



ALRAINBOW said:


> [1] Oh one last thought how about you guys upload on a cloud share link some files and I’ll comment honestly on them ok. Just do aiff and flac and 16/24 bit depth ok I mean this. You guys are clever enough to make me not know the file info and my dacs don’t show it. Only my player ok.
> [2] Lets have some fun as a group on this ok.



1. You seem to think this forum (and it's members) exists for you personally. This is the Sound Science forum which as the name suggests, exists to discuss the science of sound, it's NOT the "Alrainbow learning" forum. In other words, if you really want to learn the facts/science that's good but it's your responsibility to learn, it's not our responsibility to teach you. Therefore, if you want various test files, it's up to YOU to make or find them yourself, it's not up to us to make them for you.

2. Why would it be fun for us as a group? Creating various test files for you, in order to prove something to you that science has already proven and that we already know, would not be fun, it would be boring.

If you are polite and ask questions, then the members here are generally very good at answering those questions. However you've been the exact opposite, you've been very impolite and rather than ask questions you've just make-up some false assertions, which is a perversion of this subforum and therefore even more impolite! With regards to your aiff vs flac comparison, you can easily do an equivalent test yourself: On your computer, simply select your aiff file, right click and copy & paste (or duplicate) it. Play both files and if you can hear a difference between them, either you're imagining a difference where there isn't one or you have a faulty system. A 16 vs 24bit comparison is more involved but again, if you're polite and ask, I or another member here will explain how to create the test files.

G


----------



## board

ALRAINBOW said:


> Anyway when I did the math hahaha I think 16 to 24 is about 256 times more level steps in loudness. Am I getting it correctly ? If so can rbis be why I perceive a blacker background and lower noise floor.


This, to me, is another nail in your coffin, because it shows that you didn't read the very first post on page one by Gregorio, which shows exactly why what you're saying is wrong. But please don't take that personal - most unknowing audiophiles believe the same thing as you. The point that Gregorio was making was the entire argument of his post, and you completely missed it. But I won't quote what he said, but instead ask you to go back and read the very first post again, hoping that you will learn something.
Or it's also possible that you did read the original post, but then simply didn't understand the central and very clear message. Either option supports why you're saying the things you're saying - either you didn't read it, or you didn't understand it.

But I do actually have a serious question for you:
What, if anything, can make you change your mind about your claim that you can hear a difference between 16 and 24 bit? 
Or to be more specific, what, if anything, can make you change your mind about your claim that you actually heard audible differences, and can demonstrate it in a blind test, between those particular 16 and 24 bit files that you claim to have listened to?

And in case anyone was wondering (I assume nobody is), then I won't ask you for a log again, 'cause it's very clear that you don't have one, nor will you ever post one.


----------



## board

gregorio said:


> 2. Why would it be fun for us as a group? Creating various test files for you, in order to prove something to you that science has already proven and that we already know, would not be fun, it would be boring.


Also, if we send him files, he will most likely look at what file is which and then claim to have heard enormous differences. Alternatively, he might guess and get it right, since there's a 50 % chance anyway. One thing is for sure: He won't do an ABX test of the things he claims to be able to hear.


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## bigshot (Nov 27, 2019)

I was corrected here in the past for using the term "bit rate" as the combined data rate of both bit depth and sampling rate. I was told that "bit rate" (16, 24, etc) is to "bit depth" (noise floor expressed as -96dB, -144dB, etc.) as "sampling rate" (44.1, 48, 96, etc.) is to "frequency response" (20 to 20kHz, 20 to 48kHz, etc). That makes sense to me, but perhaps in practice the term has different meanings, or perhaps the usage has changed over the years. I'm perfectly comfortable using "bit rate" as a synonym for "data rate". I was just trying to use the most specific definition.

Interesting to see the term "rate" correlated to time. I always have correlated it to the amount... higher rate = bigger number = more. Like with "tax rate" or "rate of speed".


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## Pro-Jules (Nov 27, 2019)

Dr. Strangelove said:


> Quite an excellent write up. I am eager to see the rebuttal.



Decide by using your ears.

My ears prefer 96k 24 bit.

88.2
177
192 

also good. 

I can hear more depth into the production, more sub bass and high frequency extension too.

As a recording engineer I would always opt to capture the final mix @ 24 bit 96k over 44.1k 16bit


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## bigshot

Pro-Jules said:


> I can hear more depth into the production, more sub bass and high frequency extension too.



(whispering) I hear DEAD PEOPLE!


----------



## CoryGillmore

gregorio said:


> In this instance, TheSonicTruth is correct and bigshot is wrong. CD for example, has a bit rate of 705,600 bits per sec (per channel), while SACD has a bit rate of 2,800,000 bits per sec (per channel). SACD has a far higher bit rate but a bit depth of only 1bit, while CD has a bit depth of 16bits. So, contrary to bigshot's assertion, higher bit rate does NOT also mean higher bit depth! Same with dynamic range, SACD only has a dynamic range of ~6dB (without noise-shaped dither), while CD has a dynamic range of ~96dB (without noise-shaped dither). So again, SACD has a far higher bit rate than CD but a far lower dynamic range.
> 
> 
> 
> ...


Can someone point me to where this guy has been impolite? He may be wrong on the topic, but to me everyone else has been a condescending dick to him. I have only seen this guy try to be friendly and laugh off confrontation with light hearted responses only to be met with short snippy replies. Maybe I overlooked some of his replies?


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## bigshot (Nov 27, 2019)

Well for about a year he has been claiming that he can easily tell the difference between 16 bit and 24 bit. Folks have patiently explained to him why that is about as unlikely as the moon being made of green cheese, but he persists in making the claim. When asked for proof, he just goes back to saying he can easily hear the difference again. Around here, we are allowed to ask for proof to verify claims. If someone repeatedly ignores that and just larks around saying things that aren't true, then they are going to be summarily dismissed by the group. That isn't being impolite. This is the internet. You have to expect stuff like that. We just don't have a lot of patience with people who spout nonsense and expect us to nod and smile and let them spread it all over to crap up a thread. We've been down that road too many times.


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## castleofargh

Pro-Jules said:


> Decide by using your ears.
> 
> My ears prefer 96k 24 bit.
> 
> ...


that's what we all do and suggest to others. the problem starts with what people define as "using their ears". to me it is using my ears as opposed to using my eyes and my preconceptions, so I test by literally trying to find out something with my ears(blind test). but for other people, using their ears means simply focusing on sound a little more in their mind, and assuming that by doing so, their impressions of sound will only relate to sound. except that's not how human brains work. so those people often reach mistaken conclusions about sound. it's even more likely when the differences we try to perceive are near or beyond Just Noticeable Differences. the fewer differences we get from sound, the more likely we are to rely on other senses and preconceptions to draw some fake heuristic conclusions about "sound".


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## bigshot

I decide using my feet!


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## ALRAINBOW

board said:


> This, to me, is another nail in your coffin, because it shows that you didn't read the very first post on page one by Gregorio, which shows exactly why what you're saying is wrong. But please don't take that personal - most unknowing audiophiles believe the same thing as you. The point that Gregorio was making was the entire argument of his post, and you completely missed it. But I won't quote what he said, but instead ask you to go back and read the very first post again, hoping that you will learn something.
> Or it's also possible that you did read the original post, but then simply didn't understand the central and very clear message. Either option supports why you're saying the things you're saying - either you didn't read it, or you didn't understand it.
> 
> But I do actually have a serious question for you:
> ...


How do you have a clue where I read the info. ??? And lastly I cannot be wrong if I simply say it’s my observation. you need a weekend here by me to fix ur thoughts please.


----------



## ALRAINBOW

castleofargh said:


> that's what we all do and suggest to others. the problem starts with what people define as "using their ears". to me it is using my ears as opposed to using my eyes and my preconceptions, so I test by literally trying to find out something with my ears(blind test). but for other people, using their ears means simply focusing on sound a little more in their mind, and assuming that by doing so, their impressions of sound will only relate to sound. except that's not how human brains work. so those people often reach mistaken conclusions about sound. it's even more likely when the differences we try to perceive are near or beyond Just Noticeable Differences. the fewer differences we get from sound, the more likely we are to rely on other senses and preconceptions to draw some fake heuristic conclusions about "sound".


Not flaming but are you claiming you don’t HEAR with ears ? lol


----------



## ALRAINBOW

board said:


> This, to me, is another nail in your coffin, because it shows that you didn't read the very first post on page one by Gregorio, which shows exactly why what you're saying is wrong. But please don't take that personal - most unknowing audiophiles believe the same thing as you. The point that Gregorio was making was the entire argument of his post, and you completely missed it. But I won't quote what he said, but instead ask you to go back and read the very first post again, hoping that you will learn something.
> Or it's also possible that you did read the original post, but then simply didn't understand the central and very clear message. Either option supports why you're saying the things you're saying - either you didn't read it, or you didn't understand it.
> 
> But I do actually have a serious question for you:
> ...


Come to my place. I won’t yell or attack you lol. I’m simply saying I can and to add to this plight for you guys I have a few buds who can too. One is a pro who makes music.


----------



## ALRAINBOW

board said:


> Also, if we send him files, he will most likely look at what file is which and then claim to have heard enormous differences. Alternatively, he might guess and get it right, since there's a 50 % chance anyway. One thing is for sure: He won't do an ABX test of the things he claims to be able to hear.


I won’t cheat guys no need to I pass or fail honestly. If I fail I’ll say I did.


----------



## sander99

ALRAINBOW said:


> Not flaming but are you claiming you don’t HEAR with ears ? lol


Not _only_ with the ears no, hearing involves not just your ears and the actual sound but all your senses and the thoughts and ideas and convictions in your mind as well, that's how our brains work.
_Those things can make someone hear "night and day differences" where there are no audible differences at all in the sound itself._
That's why it is possible that so many people believe in so many audio related myths. (And companies of course like to create/enhance/exploit that to increase sales).
And that's why (level matched) blind testing is so important.

Actually this might be the most important question you have asked. If you are not aware of above facts, if you think that what you hear only depends on what sound enters your ears then it must _seem_ like we are all crazy here. Only once you are aware of and understand above facts you can start to understand that your convictions may be wrong.


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## bigshot

ALRAINBOW said:


> I cannot be wrong if I simply say it’s my observation..



Yes, it is your observation, and yes it is wrong.


----------



## gregorio

Pro-Jules said:


> [1] Decide by using your ears. My ears prefer 96k 24 bit. 88.2 177 192 also good. I can hear more depth into the production, more sub bass and high frequency extension too.
> [2] As a recording engineer I would always opt to capture the final mix @ 24 bit 96k over 44.1k 16bit



1. As has been discussed earlier in this thread, there are definite advantages of 24bit over 16bit for recording engineers. Namely, the huge dynamic range of 24bit allows much greater headroom (when recording) and therefore a lower mic pre-amp gain setting, which can result in less mic pre-amp noise/distortion. Additionally, many years ago, some non-linear plugins worked audibly better at 88.2 or 96kHz sample rates. *However:* Firstly, consumers reproducing a completed master obviously don't use mic pre-amps and also obviously don't need any headroom and Secondly, for more than a decade or so, non-linear plugins which can benefit from a higher sample rate, simply upsample internally.

2. Firstly, a recording engineer is not responsible for capturing the final mix, that's the job of the mix engineer, although admittedly, it's common for the recording engineer to also act as the mix engineer. Secondly, it's wise (and standard practice) to reduce the final mix to 24bit, again with significant headroom, for the benefit of the mastering process. Lastly, it's also wise as a music producer, recording, mix or mastering engineer to opt for 24/96 because a client may at some point require a hi-res version (for marketing purposes). Lastly, this thread is not aimed at the music producers and engineers creating music, it's aimed at consumers reproducing completed music masters, where there are no audible benefits of 24bit over 16bit. 

If someone is going to call themselves a Recording Engineer (rather than just a Recording Person), surely it's required to have a decent/good understanding of not only recording but also of the underlying engineering?

G


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## castleofargh (Nov 28, 2019)

ALRAINBOW said:


> Not flaming but are you claiming you don’t HEAR with ears ? lol


I'm saying that our impressions of hearing come from a pudding of senses, ideas, memorized patterns, etc. it's trivial to demonstrate that we can influence what listeners think they hear with non audio stimuli. in fact it is trivial to demonstrate that almost anything can influence impressions, beliefs, and even the way we will think. plenty of examples in the book I suggested to you the other day. and that's exactly why controlled testing is so important when trying to determine what we hear as opposed to what we think we hear. on such a topic the difference tends to be dramatic and be the very root of the debate.


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## bigshot

ALRAINBOW said:


> I won’t cheat guys no need to I pass or fail honestly. If I fail I’ll say I did.



But what if by saying you aren't lying, you are lying? Or maybe you say you are lying... are you, or are you telling the truth? https://en.wikipedia.org/wiki/Liar_paradox

The easiest way to convince people is to do a controlled listening test.


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## gregorio (Nov 28, 2019)

CoryGillmore said:


> [1] Can someone point me to where this guy has been impolite?
> [2] He may be wrong on the topic, but to me everyone else has been a condescending dick to him.
> [3] I have only seen this guy try to be friendly and laugh off confrontation with light hearted responses only to be met with short snippy replies. Maybe I overlooked some of his replies?



1. Sure, here are a few quotes:
"_I can and most people with good ears can pick out 24 over 16 bit depth._" - According to this assertion, all those who cannot "pick out 24 over 16bit" (IE. Everyone!), have poor/bad ears. You think it's polite to tell everyone they have bad ears/poor listening skills, when in fact he has no idea what listening skills others have?
"_If you don’t hear it there are two reasons One being the system used is subpar and can’t show what many can hear._" - As an alternative to us all having poor listening skills/ears, here he's stating that we must have subpar systems. Most (if not all) of us here have put in considerable time, effort and money in order to achieve systems that are above par (good to excellent). You think it's polite to state our systems must be subpar?
"_I follow all the math and it’s conjecture ..._" - This assertion is clearly and demonstrably false and as this is the sound Science forum, false assertions are unacceptable. Of course, we can all suffer from misunderstandings and inadvertently make a false assertion but how is repeating the same unacceptable, false assertion even after it's been refuted, anything other than impolite? Especially as quite easy to avoid making false assertions in the first place, by simply asking as a question, instead of making a factual assertion.

2. And you don't think that stating/implying we all have "ears" (listening skills) and/or systems which are inferior to his is being a "condescending dick" to start with? In all likelihood, some of us have significantly better listening skills than him and it's virtually certain that at least one of us has a system significantly superior to his, so not only was he being a "condescending dick" but a hypocritical one!

3. Then you appear to have an issue with what you have "only seen". As many/most of his posts have included variations of the first two quotes above, you have maybe overlooked many/most of his replies?

G


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## bigshot

Well, I'm not offended. I can spot this kind of thing a mile away now. Hard to get worked up over it any more.


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## gregorio

ALRAINBOW said:


> I cannot be wrong if I simply say it’s my observation.



1. Great, so if I say I've observed ghosts, fairies, unicorns (or whatever), I cannot be wrong and they all must exist?



ALRAINBOW said:


> Not flaming but are you claiming you don’t HEAR with ears ? lol



Of course we don't hear with our ears, we hear (perceive sound) with our brains. Are you claiming you don't have a brain? lol



ALRAINBOW said:


> [1] I’m simply saying I can and to add to this plight for you guys [1a] I have a few buds who can too. [1b] One is a pro who makes music.



1. Which is impolite/insulting because we do not have a "plight", our hearing (and systems) agree with the demonstrated/proven science. You're the one with the "plight" because your hearing (or system) doesn't!
1a. So, you have a few buds with the same "plight" as you.
1b. That's not a good "plight" to have for a pro! Incidentally, I have many buds who cannot tell the difference between identical audio files and most of them are "pros who make music" for a living.

G


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## Pro-Jules (Nov 28, 2019)

gregorio said:


> 1. As has been discussed earlier in this thread, there are definite advantages of 24bit over 16bit for recording engineers. Namely, the huge dynamic range of 24bit allows much greater headroom (when recording) and therefore a lower mic pre-amp gain setting, which can result in less mic pre-amp noise/distortion. Additionally, many years ago, some non-linear plugins worked audibly better at 88.2 or 96kHz sample rates. *However:* Firstly, consumers reproducing a completed master obviously don't use mic pre-amps and also obviously don't need any headroom and Secondly, for more than a decade or so, non-linear plugins which can benefit from a higher sample rate, simply upsample internally.
> 
> 2. Firstly, a recording engineer is not responsible for capturing the final mix, that's the job of the mix engineer, although admittedly, it's common for the recording engineer to also act as the mix engineer. Secondly, it's wise (and standard practice) to reduce the final mix to 24bit, again with significant headroom, for the benefit of the mastering process. Lastly, it's also wise as a music producer, recording, mix or mastering engineer to opt for 24/96 because a client may at some point require a hi-res version (for marketing purposes). Lastly, this thread is not aimed at the music producers and engineers creating music, it's aimed at consumers reproducing completed music masters, where there are no audible benefits of 24bit over 16bit.
> 
> ...



I am not sure if you are trying to insult me with that last paragraph, (laughable if so) but to let you know, I was a 25+ year, career, recording studio audio engineer professional.

Some clarifications

Mixes these days aren't 'reduced' to 24 bit. The individual 'tracks' (instruments, voices) often recorded @ 44.1k or 48k or 88. 2k or 96k *@24 bit -* so a final mix @ 24 bit isn't a reduction. (Its simply a continuation of the recording session's bit rate)

The clients wouldn't want a Hi Res version for "marketing purposes'  (as you cynically state above) - they would want it to offer superior sounding hi-res versions.

And I am out.

Good day gentlemen.


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## ALRAINBOW

Come to my place do the test here witness it. Video it lol. I have a njce place plenty of audio toys to look at. 
do you guys know what stuff I have here ?


----------



## gregorio

Pro-Jules said:


> [1] I am not sure if you are trying to insult me with that last paragraph, (laughable if so)
> [1a] but to let you know, I was a 25+ year, career, recording studio audio engineer professional.
> [2] Mixes these days aren't 'reduced' to 24 bit. The individual 'tracks' (instruments, voices) often recorded @ 44.1k or 48k or 88. 2k or 96k *@24 bit -* so a final mix @ 24 bit isn't a reduction
> [3] The clients wouldn't want a Hi Res version for "marketing purposes'  (as you cynically state above) - they would want it to offer superior sounding hi-res versions.
> ...



1. I wasn't sure either but after your last post, I'm more inclined to think it should have been!! See point #2 below for example.
1a. It's irrelevant but "just to let you know", I've been a recording studio professional for nearly 30 years, in some of the world's top recording studios with some of the top artists, I've also authored and taught a university degree in the subject. I could go on but as it said, it's irrelevant to the actual facts!

2. Oh dear, clearly you do not know/understand some of the basic facts of engineering! Digital mixes these days (and indeed for the last 25+ years) are NEVER made at 24bit. The professional mix environment today and for the last decade or so is 64bit float, prior to that (from the mid/late 1990s) it was 32bit float in digital mixers or 56bit fixed (in the case of the ubiquitous Pro Tools), even my very first personal digital mix environment (in 1992) was 28bit. Therefore, a final mix @ 24 bit MUST be a reduction! As this is taught even in introductory courses to DAWs and digital mixing, how is it possible that a 25+ year professional music/sound engineer doesn't know this?

3. Clients who actually know anything about recording resolution (and admittedly many of them don't) do want a Hi res version for purely maketing purposes because they know there is no audible difference. As a professional engineer, it would be almost unbelievable/unconscionable that you've never done any controlled listening tests?

4. Probably a wise move. Your "appeal to authority", against some basic engineering facts and against a body of reliable scientific evidence, won't work here in the sound Science subforum and is unlikely to end well!
4a. And a good day to you.

G


----------



## gregorio

ALRAINBOW said:


> [1] Come to my place do the test here witness it. Video it lol.
> [2] I have a njce place plenty of audio toys to look at.
> [3] do you guys know what stuff I have here ?



1. Why do I need to come to your place to witness either faulty equipment or a faulty listening test and personal expectation bias, when I've already witnessed those things numerous times?

2. Maybe that's your problem? Instead of looking at your audio toys, maybe you should try only listening to them!

3. I've got no idea what stuff you have but likewise, you've got no idea what stuff I have. So how do you know my stuff is subpar (or that my listening skills are)? Not that it's even slightly relevant but it's very unlikely your place is as "nice" as mine or that you have as many audio "toys" and it's definitely not as nice or has as many audio toys as many of the places I've spent considerable time in!

G


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## ALRAINBOW

Thanks for your post above I never knew there were higher bit depth rates. If I may why do I perceive a lower noise floor at my place ? I’m not saying it is but seems so. Also 16 bit depth does seem to have air like noise at times ? I have tracks at 44.2 16 and same tracks at 88.2 24. 24 does seem to have a perception of blacker back ground. I wish redbook was this rate


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## ALRAINBOW

gregorio said:


> 1. Why do I need to come to your place to witness either faulty equipment or a faulty listening test and personal expectation bias, when I've already witnessed those things numerous times?
> 
> 2. Maybe that's your problem? Instead of looking at your audio toys, maybe you should try only listening to them!
> 
> ...


Bro it’s a kind jester  only bring stuff to measure tell all I’m nuts or damn there is something going on it’s all I mean. Don’t get mad


----------



## chef8489

ALRAINBOW said:


> Thanks for your post above I never knew there were higher bit depth rates. If I may why do I perceive a lower noise floor at my place ? I’m not saying it is but seems so. Also 16 bit depth does seem to have air like noise at times ? I have tracks at 44.2 16 and same tracks at 88.2 24. 24 does seem to have a perception of blacker back ground. I wish redbook was this rate


Because you have it in your mind that what you hear is real. If you did a properly held double blind test it would prove otherwise. They are identical within the hearing spectrum.


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## TheSonicTruth (Nov 28, 2019)

Pro-Jules said:


> a final mix @ 24 bit isn't a reduction. (Its simply a
> continuation of the recording session's bit rate)



Bit DEPTH, bit DEPTH,

smh God I HATE digital audio!


----------



## Pro-Jules

gregorio said:


> 1. I wasn't sure either but after your last post, I'm more inclined to think it should have been!! See point #2 below for example.
> 1a. It's irrelevant but "just to let you know", I've been a recording studio professional for nearly 30 years, in some of the world's top recording studios with some of the top artists, I've also authored and taught a university degree in the subject. I could go on but as it said, it's irrelevant to the actual facts!
> 
> 2. Oh dear, clearly you do not know/understand some of the basic facts of engineering! Digital mixes these days (and indeed for the last 25+ years) are NEVER made at 24bit. The professional mix environment today and for the last decade or so is 64bit float, prior to that (from the mid/late 1990s) it was 32bit float in digital mixers or 56bit fixed (in the case of the ubiquitous Pro Tools), even my very first personal digital mix environment (in 1992) was 28bit. Therefore, a final mix @ 24 bit MUST be a reduction! As this is taught even in introductory courses to DAWs and digital mixing, how is it possible that a 25+ year professional music/sound engineer doesn't know this?
> ...



You are twisting things. 

I was referring to 24 bit sound files. Which are a very standard music recording format. 

Your 64 bit, 32bit, 56 bit Mix +  (I know the person who discovered this and outed Pro Tools on it) float argument is all a warped smokescreen as those numbers relate to the *mix engine architecture* - NOT the format of the recorded sound files. 

A mastering session of analog (Let's say  classical) recordings captured at 24 bit 96k (in whatever 64 or 32 bit floating point mix engine architecture) that produced a 96k 24 bit hi res product - would sound better than its 44.1k 16 bit cd version. It wouldn't be simply be for marketing. 
It would sound different. Very likely better. 

Are you declaring the hi res movement (Qobuz, Tidal, HD Tracks etc) as bogus?


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## bigshot (Nov 28, 2019)

Pro-Jules said:


> I am not sure if you are trying to insult me with that last paragraph, (laughable if so) but to let you know, I was a 25+ year, career, recording studio audio engineer professional.



Then perhaps you have worked with Gregorio or maybe me.

By the way... yes, the HiRez audio movement is bogus. Higher bit rates only contain information that is inaudible in normal home playback. Greater response and a lower noise floor are only needed for mixing. For playback, a CD can contain everything you can possibly hear.

At the end of every mix, I do a bounce down to 16/44.1 and check it against the sound coming off the board. In 35 years, I've never heard any difference. If I did, I would throw up a red flag and we would try to figure out what was wrong with the bounce down.


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## Pro-Jules (Nov 28, 2019)

That may be the limitation of your hearing. You don't speak for me though.


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## Pro-Jules (Nov 28, 2019)

You say



bigshot said:


> The HiRez audio movement is bogus. Higher bit rates only contain information that is inaudible in normal home playback. Greater response and a lower noise floor are only needed for mixing. For playback, a CD can contain everything you can possibly hear.
> .



Only needed?

Can contain?

So a hifi / music enthusiast should only maintain a "normal" playback system?

Not one where higher bit rate information is audible?

Should only people like yourself have access to playback systems for higher bit rates? Or should they only be allowed for professionals mixing?

I think you are trolling us anyway, this sentence is pure gibberish! 





bigshot said:


> Higher bit rates only contain information that is inaudible in normal home playback..



What is ‘normal’ home playback?  My home hifi playback d/a converter handles files up to 192k 24bit. It's in my living room next to the TV. Is that not allowed?
When I play high bit rate audio - its audible!

Definitely trolling. V funny though! Bravo!


----------



## upstateguy (Nov 28, 2019)

bigshot said:


> But what if by saying you aren't lying, you are lying? Or maybe you say you are lying... are you, or are you telling the truth? https://en.wikipedia.org/wiki/Liar_paradox
> 
> The easiest way to convince people is to do a controlled listening test.



It's obvious that the arguer has never been the subject of a double blind test.


----------



## Pro-Jules (Nov 28, 2019)

The subject?

Do you mean a participant?

The subject(s) would be the audio files to be evaluated Participants would be the people evaluating.


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## upstateguy (Nov 28, 2019)

castleofargh said:


> I'm saying that our impressions of hearing come from* a pudding of senses*, ideas, memorized patterns, etc. it's trivial to demonstrate that we can influence what listeners think they hear with non audio stimuli. in fact it is trivial to demonstrate that almost anything can influence impressions, beliefs, and even the way we will think. plenty of examples in the book I suggested to you the other day. and that's exactly why controlled testing is so important when trying to determine what we hear as opposed to what we think we hear. on such a topic the difference tends to be dramatic and be the very root of the debate.



What a wonderful phrase....


----------



## Pro-Jules

Some people in this thread are _pudding_ us on I think.


----------



## upstateguy

Pro-Jules said:


> The subject?
> 
> Do you mean a participant?
> 
> The subject(s) would be the audio files to be evaluated Participants would be the people evaluating.



No I meant subject.  It's either or. Link


----------



## upstateguy

Pro-Jules said:


> Some people in this thread are _pudding_ us on I think.



And some people are trolling.


----------



## bigshot

Pro-Jules said:


> So a hifi / music enthusiast should only maintain a "normal" playback system? Not one where higher bit rate information is audible?!



I’m not trolling. I’m surprised you’ve never heard this before. The lowered noise floor in higher bit depth formats is beyond anything one might be able to perceive in the home. Normal loud listening volume is around 80-85dB. The threshold of pain and the point where you incur hearing damage is 120dB. The natural noise floor in your home is over 30dB. Add 30 dB to the 96dB range of 16 bit audio and you have a dynamic range that exceeds the threshold of pain. Commercially recorded music is mixed to have a dynamic range of 50-55dB at most. What is the purpose of the 144dB of 24 bit when 16 bit is already overkill?

Likewise, CD quality sound covers the entire range of frequencies that human ears can hear... 20Hz to 20kHz. Above that only dogs and bats can hear it. What is the point of the one measly octave of extra frequencies provided by 96k sampling when you can’t even hear it? Also, if you run ultrasonic frequencies through consumer amps and transducers not designed to reproduce them, you end up with harmonic distortion down in the audible range. Therefore, ultrasonic frequencies can’t improve perceived sound quality, they can only degrade it.

It’s good to be able to play formats like SACD or “HD Audio” files for compatibility’s sake, but those formats offer no sonic advantage over CD. The only differences that are audible are differences in mastering or mixing, and that is a crap shoot that is completely unrelated to format.

If you’d like to read about this in more detail, see the link in my sig called CD Sound Is All You Need. As a sound engineer, you should be aware of this.


----------



## bigshot

Pro-Jules said:


> That may be the limitation of your hearing. You don't speak for me though.



I’m assuming you are human. If I’m incorrect, I’d be interested in how you manage to type with paws or floppy bat wings.


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## upstateguy (Nov 28, 2019)

bigshot said:


> I’m not trolling.
> 
> <snip>
> 
> ......



We know you're one of the good guys. 

Were you not instrumental in the inception of the "Objectivist Forum"?

You guys remember the beginning of the Objectivist's Forum don't you?  When we were trying to bring light into the dark?

*"The Objectivist Audio Forum (opening soon) If you feel the uncontrollable urge to use the word "placebo," in here is where you'll post (and only here). Discuss DBT all you want in here."*

Forum: Sound Science
Thread: The Objectivist Audio Forum
"It gives you a place to complain, so that's a good thing. It also keeps you from complaining everywhere else."  <--- this in response to the truth


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## Pro-Jules (Nov 28, 2019)

bigshot said:


> It’s good to be able to play formats like SACD or “HD Audio” files for compatibility’s sake.



So what, in your mind, is good about playing them exactly? (If it's not the superior sound of them!)


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## old tech (Nov 28, 2019)

Pro-Jules said:


> So what, in your mind, is good about playing them exactly? (If it's not the superior sound of them!)


I can't speak for Big Shot but it is fairly well known that many (but certainly not all) SACD and hi res releases are mastered for higher fidelity sound because (particularly with SACDs) the label knows the listener would be more of an audiophile, would have invested in a good home stereo and listen to it in a quiet environment.  That sort of mastering is less suitable for general listening in cars, ear buds etc.  So, SACD and hi res formats in themselves do not offer any sonic advantages within the domain of human hearing (well at least 30 years of controlled testing has not revealed any advantages), any sonic advantages of these formats lies in the mastering to cater for the market of these formats. As a sound engineer I would of thought you'd know this?


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## Pro-Jules (Nov 29, 2019)

You say his-res is for higher fidelity sound, then flip to the garbage mantra I read from some here... “but it's not suitable for general listening".

I put it to you that this community is largely not the general listening population as many own equipment suitable for distinguishing the “higher fidelity” you referred to.

therefore hi-res isn’t a sonic waste of time or bogus.

As a sound engineer I know high sample rates are enjoyed for their extended frequency capabilities, (higher highs and lower lows) more 3D depth into the soundstage and and improved ability to hear reverb decay (or reverb 'tails').

Further, as this is an audio reproduction enthusiast discussion website can we retire the snooty “general listening” arguments please? The visitors to this site are a discerning listening audience and should be credited as such.

Someone else have a go at answering.

You haven't.


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## gregorio

Pro-Jules said:


> [1] You are twisting things.
> [2] I was referring to 24 bit sound files. Which are a very standard music recording format.
> [3] Your 64 bit, 32bit, 56 bit Mix + (I know the person who discovered this and outed Pro Tools on it) float argument is all a warped smokescreen as those numbers relate to the *mix engine architecture* - NOT the format of the recorded sound files.
> [4] A mastering session of analog (Let's say classical) recordings captured at 24 bit 96k (in whatever 64 or 32 bit floating point mix engine architecture) that produced a 96k 24 bit hi res product - would sound better than its 44.1k 16 bit cd version. It wouldn't be simply be for marketing. It would sound different. Very likely better.
> [5] Are you declaring the hi res movement (Qobuz, Tidal, HD Tracks etc) as bogus?



1. No I'm not, I'm just stating the facts.

2. Yes, 24bit is the standard sound file recording format but of course, unless you're recording direct to disk, then those recorded tracks have to be mixed!

3. "_A warped smokescreen_", what are you talking about? You stated "_a final mix @ 24bit_", is this final mix you're talking about in fact not mixed? If it is mixed (in the digital domain), then obviously it's mixed within the "_*mix engine architecture*_" and the bit depth of that mix is whatever the bit depth of the final summing bus is (IE. 64bit float or in older versions of Pro Tools, 56bit fixed). If you want to record that mix as a 24bit sound file, then obviously you have to reduce the bit depth of the mix from 64bit float (or 56bit fixed) to 24bit. Again, this is digital mixing 101, how is it possible that a 25+ year pro wouldn't know this?

4. So are you admitting that as a 25+ year pro, you've never done any controlled listening tests? Effectively, you're claiming that you can hear artefacts that are either around -120dBFS (which are not reproducible in the first place) or above 20kHz. So, you're not only contradicting the well established and accepted science of human hearing thresholds but the demonstrated science of actual DBTs (such as the Meyer & Moran study for example) of hi-res vs CD and what evidence do you present to substantiate your extraordinary claim? That you were a sound/music engineer who's apparently never done any controlled listening tests?

5. You're joking, of course I am. Have you not read the original post of this thread? If you are/were a pro sound engineer then you should know it anyway!



Pro-Jules said:


> [1] So a hifi / music enthusiast should only maintain a "normal" playback system? Not one where higher bit rate information is audible?
> [2] Should only people like yourself have access to playback systems for higher bit rates? Or should they only be allowed for professionals mixing?
> [3] What is ‘normal’ home playback?
> [3a] My home hifi playback d/a converter handles files up to 192k 24bit.
> [4] When I play high bit rate audio - its audible! Definitely trolling. V funny though! Bravo!



1. As higher than 16bit is not audible, what is a non-normal playback system? Some magical system capable of more than 120dB dynamic range and changing a listener's physiology so that it's not painful/damaging?

2. Anyone can have higher bit depth systems if they want, they're just not going to get any audible benefit unless they're using that system as professional mixing engineers do, EG. Generating substantial quantisation error by using numerous digital processors or by significantly deviating from appropriate gain staging, which of course is why professional digital mix environments use a 64bit architecture in the first place!

3. Normal home playback covers a range of scenarios within "reasonable" playback levels (IE. Not painful/damaging).
3a. No it doesn't! Sure, your DAC can handle a 24bit file FORMAT but it cannot "handle" 24 bits of audio data, unless you have a magic DAC that breaks the laws of physics?

4. Again, all the above should be known by a first year recording student, let alone by a 25+ year professional! Plus, just repeating the same assertion that contradicts the science and providing no reliable evidence to support your extraordinary claim/assertion is pretty much the definition of trolling here (or in science in general). So who is "definitely trolling" here? Hypocrisy at it's finest, V. funny though, bravo!



Pro-Jules said:


> [1] So what, in your mind, is good about playing them exactly? (If it's not the superior sound of them!) [And]
> [2] You say his-res is for higher fidelity sound, then flip to the garbage mantra I read from some here... “but it's not suitable for general listening".
> I put it to you that this community is largely not the general listening population as many own equipment suitable for distinguishing the “higher fidelity” you referred to. therefore hi-res isn’t a sonic waste of time or bogus.
> Further, as this is an audio reproduction enthusiast discussion website can we retire the snooty “general listening” arguments please?



And again, basic information already covered in this thread and that any first year music/sound engineering student should already know, actually most first semester students should know! But as you apparently don't know:

1. What "_is good about playing them_" is potentially a different master. A master designed for at least decent equipment in at least a decent listening environment, rather than a more compressed (for example) master better suited to a wider range of equipment and noisier listening environments. HOWEVER, even masters designed ONLY for high quality playback systems/environments rarely have a dynamic range greater than about 60dB, which is about 1,000 times less than the 16bit format allows! Again, how is it possible, as a 25+ year professional, that you don't already know this? Unfortunately though, certain labels/distributors choose to only distribute these masters in hi-res formats (for marketing/pricing reasons), rather than distribute them in 16bit format.

2. All the false assertions you make here are refuted by the point above!

G


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## TheSonicTruth (Nov 29, 2019)

upstateguy said:


> And some people are trolling.





old tech said:


> I can't speak for Big Shot but it is fairly well known that many (but certainly not all) SACD and hi res releases are mastered for higher fidelity sound because (particularly with SACDs) the label knows the listener would be more of an audiophile, would have invested in a good home stereo and listen to it in a quiet environment.  That sort of mastering is less suitable for general listening in cars, ear buds etc.  So, SACD and hi res formats in themselves do not offer any sonic advantages within the domain of human hearing (well at least 30 years of controlled testing has not revealed any advantages), any sonic advantages of these formats lies in the mastering to cater for the market of these formats. As a sound engineer I would of thought you'd know this?




In the cases of popular releases(classic rock, hip-hop, country, etc) all that is done with the SACDs is to master them louder, at the expense of some dynamic range. 

Though there are some on here who might either defend this nefarious practice or deny it is being done(even though anyone can see the evidence in the most basic of DAWs) because they work in the recording industry, I'm not going to argue with them because it's a waste of time.  That said, it might explain some of the differences you are hearing between CD and SACD, and CD vs 'Hi-Res'.


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## old tech (Nov 29, 2019)

TheSonicTruth said:


> In the cases of popular releases(classic rock, hip-hop, country, etc) all that is done with the SACDs is to master them louder, at the expense of some dynamic range.
> 
> Though there are some on here who might either defend this nefarious practice or deny it is being done(even though anyone can see the evidence in the most basic of DAWs) because they work in the recording industry, I'm not going to argue with them because it's a waste of time.  That said, it might explain some of the differences you are hearing between CD and SACD, and CD vs 'Hi-Res'.


Well I did qualify it by saying "certainly not all". In fact, to my ears, many CDs sound exactly the same if not worse as their SACD counterpart and many early1980s CDs sound much better than their current squashed hi res re-issues.


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## castleofargh

Pro-Jules said:


> You say his-res is for higher fidelity sound, then flip to the garbage mantra I read from some here... “but it's not suitable for general listening".
> 
> I put it to you that this community is largely not the general listening population as many own equipment suitable for distinguishing the “higher fidelity” you referred to.
> 
> ...


as an effort to avoid pointless discussions(at least some), nobody is saying that one cannot or should not use hires files if he wants to. we all have some amount of freedom and can use it for such decisions. 
this topic is about bit depth and while in practice sample rate and bit depth go together in a number of ways, bringing sample rate into the conversation is only complicating things. so does comparing typical hires files lower resolution as whatever comes out is not specific to bit depth. we have other thread to go crazy over hires files in general. 
the first post is pretty clear about the intent of this topic, and the conversation should be about bit depth and the benefits(increasing fidelity of the encoding, pushing the quantization noise down, etc). and then wonder when those changes are audible for us or when they're expected to be audible for non mutated humans listening at sensible levels to correctly created audio albums.

now about the subjective benefits, because we're in this section, we do expect statements of audibility to be backed up with supporting evidence. even more so when we happen to have tested this for ourselves under controlled conditions and have consistently failed to pass a blind test between 16 and 24bit with our favorite tracks at normal to loud listening levels. that makes us even more anxious to see evidence from those who say that they do notice a clear difference. maybe it's about hearing abilities, maybe it's about listening skills, maybe it's about the equipment. but maybe it's made up stuff in the mind of a listener who never bothered to test his hearing ability properly. we could really stop wasting so much time and efforts if that last possibility was cleared by the people themselves before coming here to spam their overconfident claims based on garbage testing methods. 

it is a fact that 16bit is more than necessary under most circumstances. the debate only concerns niche cases and those who say otherwise are wrong. that much has been well established by decades of trials and I'm still waiting to see a legitimate research suggesting otherwise. if I take my own listening habits and listening environments, 12bit dithered is all I seem to need, and most people seem to have a hard time hearing differences beyond 13 or 14bit while listening to music at non stupidly high levels. I would not say the same of 8bit or 6bits where I could clearly notice at least the background hiss when testing music at those values. it's clearly a matter of magnitude, and of course if the same track was adjusted to peak at -20dB instead of say 0dB, and I adjusted my listening level by +20dB to get the same typical listening level, that difference would have to be reported on the lowest bit depth I need for transparency.
so the question becomes, how often does that happen? and for me the answer is never! it's not the case for everybody, but it is for me. I do have classical music with quiet passages that are stupidly quiet compared to the rest of the symphony. but the rest of it is way too loud for me to just crank up the volume of the entire piece based on that quietest part. which leaves me with 2 options:
-I leave the volume knob where it usually is and I won't hear crap on the super quiet passage. <= usually what happens and why I stop listening to those particular albums. 
-I become a human compressor and keep increasing the gain on quiet passages, then rush to lower back down when it's loud again. <= I hate that because it means I will have to endure overly loud music for at least a moment. so again in the long run I would stop listening to those musics. 

conclusion my listening habits have no circumstances where I could audibly benefit from more than 16bit. my DAC measures better if I send 24bit signal to it for some reason, so I send 24bit padding to it from 16bit albums and everybody's happy. I pay for the cheaper files, I hear the same, my DAC measures pretty well. I'm objectively and subjectively satisfied. 

a different listener with different habits and priorities, may encounter moment when 12bit isn't transparent on a regular basis despite how it's ultra rare for me. but I would argue that very few people on very rare occasions end up with musical content sounding audibly different because it has more than 16bit. and I would argue that among those, probably more than half get sound differences that have nothing to do with having higher fidelity. instead it's often about the master being different or the playback gear doing some crap when fed with some particular resolution. the legitimate cases remaining, where audibility correlates with the quantization noise going down so bit depth is the relevant factor, I would be surprised if we can find a dozen on the entire forum. and I'm confident that all of them listen to music too loud, or created the circumstances to achieve audible difference(purposefully or by malpractice, like having the digital volume on the computer at -80dB and compensating with the amp or whatever). I'm very confident about that and after all those years hanging around audiophiles, I have yet to see one solid counter example. 
the hundreds or thousands of people who "know what they're hearing" under sighted conditions might contain such counter examples. I can't know that when those never demonstrate their abilities. to me they're no different from guys saying they have seen flying saucers from mars. some could be correct. but in the absence of proper demonstration, we all save time treating the all group as making stuff up. it's just the most pragmatic conclusion. if we consider that this is the "sound science" section, no scientific research would draw conclusions based on knowing a guy who claims he can do it. facts are demonstrated, they're not acts of faith. and as we happen to be on the web, taking random statements as facts without supporting evidence, that's just gullibility. of course we don't want to that, it's internet!


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## chef8489

TheSonicTruth said:


> In the cases of popular releases(classic rock, hip-hop, country, etc) all that is done with the SACDs is to master them louder, at the expense of some dynamic range.
> 
> Though there are some on here who might either defend this nefarious practice or deny it is being done(even though anyone can see the evidence in the most basic of DAWs) because they work in the recording industry, I'm not going to argue with them because it's a waste of time.  That said, it might explain some of the differences you are hearing between CD and SACD, and CD vs 'Hi-Res'.


The loudness wars


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## Pro-Jules (Nov 29, 2019)

I wrote to a mastering engineer friend of mine for some assistance, one career audio professional to another, on the merits of hi-res formats

Here is what he advised:

_*1) For really dynamic material (such as Classical) where brief silences often happen then the extra 8 bits, that you get from 24bits compared to 16bits, helps. The dither noise applied during mastering can become audible far sooner for 16bits (-96dB) than 24bits (-144dB) as we can generally hear to around -120dB we're told. This being said the noise floor of the mic preamps, amp hiss etc. would be audible far sooner than either (maybe -50dB) so can negate this point. 

2) The anti-aliasing filter that is basically the cut off point/slope that is used to determine sample rate affects the audio we hear. So for 44.1kHz the slope is at 22.05kHz and cuts off (quite abruptly) anything over this. No problem cos we're told we only hear to 20kHz right, and the Nyquist Theory suggests we need twice the frequency (which would be 40kHz) to capture what we can humanly hear correctly. Therefore 44.1kHz gives us this plus a couple more frequencies (22.05kHz technically). 

But what you get is reflected back harmonics into the audible range that don't sound right in 44.1kHz, compared to 96kHz where the reflected back harmonics are above our hearing range. So with 44.1kHz audio, a 20kHz harmonic gets reflected back at 10kHz due to the 22.05kHz slope. We can hear 10kHz so this is not favourable. If the sample rate is 96kHz the slope is at 48kHz, so the lowest reflected back harmonic ends up as 24kHz which is above our hearing range - enough for it not the be a problem. So when you compare the two 44.1kHz vs 96kHz it is hard to determine exactly why we prefer it, but you get something 'clearer' or 'truer' in the upper range, bringing transient clarity that excites, to my ears.

Finally digital converters just sound better at higher frequencies despite not hearing the information captured above 20kHz or so. I've conducted the test countless times. I don't care to argue if someone believes I'm wrong - it's obvious with the right system and ears in my opinion.

So there's my view on it Jules. Carry on enjoying your Hi-Res audio mate, it's worth it. *_

He added...

*Now if only the world embraced 32bit float 96kHz audio (negating the need for dither) we truly would have an exact replica of the mastering studio file*.


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## castleofargh

Pro-Jules said:


> I wrote to a mastering engineer friend of mine for some assistance, one career audio professional to another, on the merits of hi-res formats
> 
> Here is what he advised:
> 
> ...


1) is as he said, although -120dB is very optimistic with musical content and a normal adult. but yeah, noises of all kinds can and will negate that point. 
2) is not on topic. 

so, here we are. 
and again, having the desire for higher resolution, or the desire to get the sound exactly like whoever intended it to be, those are fine. misrepresenting why you desire higher bit depth with empty claims and arguments about sample rate is not.


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## sonitus mirus

Pro-Jules said:


> I wrote to a mastering engineer friend of mine for some assistance, one career audio professional to another, on the merits of hi-res formats
> 
> Here is what he advised:
> 
> ...



Now you are simply repeating the same conflated issues that always prop up in this forum with no evidence to support these ideas.  Yes, if you crank up the volume to ear-splitting levels during the silent sections in music, this noise may be measurable at an audible level.  If you are listening to music at such loud volume levels that you might technically be able to notice any dithering noise during the silent parts, your ears/brain more than likely have already adjusted to the louder sounds due to a temporary auditory threshold shift, making it impossible to hear quiet noise from dithering.  Same thing if you put your ear inches from the transducer. 

These are mostly pathological examples which would also depend upon the type of dither being utilized and anti-aliasing filter choice.  Bit depth at 16-bit is not something to worry about with music.  It is more than enough.

This is why audiophiles are their own worst enemy.  If someone is playing a Hi-Res file with content in the ultrasound frequency range over speakers that can play ultrasound frequencies at a high enough dB with an ADC that is using a.45/.55 anti-aliasing filter and playing music at significantly high volume levels due to their hearing loss from age/abuse, and voila, detectable noise.


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## old tech

Pro-Jules said:


> I wrote to a mastering engineer friend of mine for some assistance, one career audio professional to another, on the merits of hi-res formats
> 
> Here is what he advised:
> 
> _*1) For really dynamic material (such as Classical) where brief silences often happen then the extra 8 bits, that you get from 24bits compared to 16bits, helps. The dither noise applied during mastering can become audible far sooner for 16bits (-96dB) than 24bits (-144dB) as we can generally hear to around -120dB we're told. This being said the noise floor of the mic preamps, amp hiss etc. would be audible far sooner than either (maybe -50dB) so can negate this point.*_


 

I have several classical CDs of direct live recordings which are dead silent in quiet passages, even at loud listening levels - some are non-dithered (being original 16 bit recordings and others are dithered but no audible difference.  Some of my early CDs which are known to be flat transfers from analog tape.  The only noise I hear is tape hiss from the master.  I think there could be an issue with incompetent mastering though.

Ian Shepperd, a well known mastering engineer, has compared the sound of competently dithered 8 bits vs 24 bits in the video below.  Note there is no loss of music information, just low level white noise. It would be humanly impossible outside contrived circumstances to hear this white noise with 16 bits.


_*


Pro-Jules said:



			2) The anti-aliasing filter that is basically the cut off point/slope that is used to determine sample rate affects the audio we hear. So for 44.1kHz the slope is at 22.05kHz and cuts off (quite abruptly) anything over this. No problem cos we're told we only hear to 20kHz right, and the Nyquist Theory suggests we need twice the frequency (which would be 40kHz) to capture what we can humanly hear correctly. Therefore 44.1kHz gives us this plus a couple more frequencies (22.05kHz technically).

But what you get is reflected back harmonics into the audible range that don't sound right in 44.1kHz, compared to 96kHz where the reflected back harmonics are above our hearing range. So with 44.1kHz audio, a 20kHz harmonic gets reflected back at 10kHz due to the 22.05kHz slope. We can hear 10kHz so this is not favourable. If the sample rate is 96kHz the slope is at 48kHz, so the lowest reflected back harmonic ends up as 24kHz which is above our hearing range - enough for it not the be a problem. So when you compare the two 44.1kHz vs 96kHz it is hard to determine exactly why we prefer it, but you get something 'clearer' or 'truer' in the upper range, bringing transient clarity that excites, to my ears.
		
Click to expand...

*_
I have heard this claim before, I think it was in 1983 largely solved by 1985 - it it was ever noticeable to most people.  But is this an issue in the 21st century with oversampling?  Why over the past 20 years or so there are no controlled tests supporting this claim?  In fact, this issue is more likely to present itself with higher sample rates which result in ultrasonic noise messing with the playback gear and creating distortion in the audible range.  Monty gives a good explanation here. https://people.xiph.org/~xiphmont/demo/neil-young.html
_*



Pro-Jules said:



			Finally digital converters just sound better at higher frequencies despite not hearing the information captured above 20kHz or so. I've conducted the test countless times. I don't care to argue if someone believes I'm wrong - it's obvious with the right system and ears in my opinion.
		
Click to expand...


So there's my view on it Jules. Carry on enjoying your Hi-Res audio mate, it's worth it. *_

He added...

*Now if only the world embraced 32bit float 96kHz audio (negating the need for dither) we truly would have an exact replica of the mastering studio file*.[/QUOTE]

Yet all the credible controlled test conducted over the past 30 years do not support this conclusion, like the decade old test below. I don't doubt your engineer friend hears a difference - it is either different masterings, volume levels or most likely, placebo.  There are no other credible explanations given what we know about human physiology and psychology, digital audio and its measurements and decades of controlled tests.
http://drewdaniels.com/audible.pdf


----------



## gregorio

Pro-Jules said:


> [A} I wrote to a mastering engineer friend of mine for some assistance, one career audio professional to another, on the merits of hi-res formats
> Here is what he advised:
> _*1) For really dynamic material (such as Classical) where brief silences often happen then the extra 8 bits, that you get from 24bits compared to 16bits, helps. [1a] The dither noise applied during mastering can become audible far sooner for 16bits (-96dB) than 24bits (-144dB) as we can generally hear to around -120dB we're told. [1b] This being said the noise floor of the mic preamps, amp hiss etc. would be audible far sooner than either (maybe -50dB) so can negate this point.
> 2) The anti-aliasing filter that is basically the cut off point/slope that is used to determine sample rate affects the audio we hear. So for 44.1kHz the slope is at 22.05kHz and cuts off (quite abruptly) anything over this. No problem cos we're told we only hear to 20kHz right, and the Nyquist Theory suggests we need twice the frequency (which would be 40kHz) to capture what we can humanly hear correctly. Therefore 44.1kHz gives us this plus a couple more frequencies (22.05kHz technically).
> ...



A. Why did you have to ask a friend for assistance, I thought you were supposed to have 25+ years of professional knowledge yourself? Let's look at the actual responses your friend gave though:

1. Yes, the extra 8 bits "helps" when recording, not on playback though because:
1a. This statement is INCORRECT. Standard triangular dither would technically become audible at about -92dB for 16bit, not -96dB. *However*, for really dynamic material (such as Classical), no competent mastering engineer would use triangular dither! It's been standard practice for nearly 3 decades to apply noise-shaped dither, which indeed provides a dynamic range down to about -120dB with 16bit. And in practice, -120dB is about the limit of any 24bit DAC anyway, due to thermal noise. Furthermore, science does NOT tell us that human hearing has a dynamic range of 120dB, it tells us that we have a dynamic range of about 60dB or less, that's employed in a moving "window" much like human vision with brightness. In other words, given certain conditions (say an anechoic chamber) we can hear down to 0dB and given the condition of a high noise floor environment (say a live gig) we can hear up to 120dB but of course, those two conditions do NOT occur concurrently! So, a dynamic range of 120dB covers all eventualities/conditions but is not appropriate for any single eventuality/condition. Plus, in practice, a 120dB dynamic range above the typical consumer listening environment noise floor (30 - 50dB) would mean a peak level of 150dB - 180dB, which is both impractical and pretty much guaranteed to cause severe hearing damage (or worse). Clearly you HAVE NOT read the OP!
1b. Yes *EXACTLY*, it's negated anyway and his figure of -50dB (about 3,000 times less than -120dB) is even less optimistic than what I stated (about 1,000 times)!! How many commercial music recordings can you name that have a dynamic range greater than about 60dB?

2. As stated this is off-topic but is not true of modern anti-alias filters. And, it is NOT hard to determine why some people prefer 96kHz vs 44.1kHz; numerous controlled tests have determined that it's the result of an expectation bias (placebo effect).

3. Now that one is nonsense! Again, current professional mix environments are 64bit float, so you still have to dither or truncate to get to a 32bit float and therefore you would NOT have an exact replica of the mastering studio mix/master. Although it's completely irrelevant because it's nowhere even vaguely near audible anyway (at any reasonable playback level), either at 32bit float, 24bit fixed or 16bit (with noise-shaped dither)! In fact, his statement above (a recording noise floor of -50dB) would only require about 10 bits (with noise-shaped dither) for the dither noise to be below the recording noise floor and inaudible!!

You stated a few pages ago that "_I am out_", to which I responded; "_Probably a wise move. Your "appeal to authority", against some basic engineering facts and against a body of reliable scientific evidence, won't work here in the sound Science subforum and is unlikely to end well!_" - However you decided to continue, dig yourself an even deeper hole and indeed, it is not ending well for you! 

G


----------



## TheSonicTruth

old tech said:


> I have several classical CDs of direct live recordings which are dead silent in quiet passages, even at loud listening levels - some are non-dithered (being original 16 bit recordings and others are dithered but no audible difference.  Some of my early CDs which are known to be flat transfers from analog tape.  The only noise I hear is tape hiss from the master.  I think there could be an issue with incompetent mastering though.
> 
> Ian Shepperd, a well known mastering engineer, has compared the sound of competently dithered 8 bits vs 24 bits in the video below.  Note there is no loss of music information, just low level white noise. It would be humanly impossible outside contrived circumstances to hear this white noise with 16 bits.
> 
> ...




Yet all the credible controlled test conducted over the past 30 years do not support this conclusion, like the decade old test below. I don't doubt your engineer friend hears a difference - it is either different masterings, volume levels or most likely, placebo.  There are no other credible explanations given what we know about human physiology and psychology, digital audio and its measurements and decades of controlled tests.
http://drewdaniels.com/audible.pdf[/QUOTE]

Ian's technical series is exquisite and educational - but - I find him to be a bit hypocritical in his crusade to both fight the loudness war and promote dynamic range.  He supports amd uses tools, as I do, such as the TT Dynamic Range Meter and Foobar 2000.  You know, the ones that return a figure of "DR6"(squashed) or DR14(dynamic and open).   His threshold for promoting dynamics in the popular genres was, at least a few years ago, DR8 by the scale of those plugins.  I even argued that he should set the bar higher, DR10 maybe, but he insisted, baby steps.

And yes, I do remember exactly what portion of the signal those tools are measuring.  I'm just talking about Shep's tendancy on this subject to talk one way but not carry the stick.


----------



## board

Pro-Jules said:


> You are twisting things.
> 
> I was referring to 24 bit sound files. Which are a very standard music recording format.
> 
> ...


Please just provide the log of the properly conducted ABX test (level-matched down to 0.1 dB, time-aligned, enough trials, etc.) that you convincingly passed between 24 and 16 bit (Foobar's ABX plugin is perfect for this). Then we won't pester you any more. 
By convincingly I mean something like at least 75-80 % correct. I usually use 16 trials for my blind tests, and when I don't pass with 16 out of 16 correct, I usually pass with 15 out of 16 (93.75 % correct).
I've given three girlfriends blind tests as well, and some they did no better than flipping a coin, although I could easily pass the same test (inexperienced vs. experienced listener), even in some cases from my kitchen (seriously), and others they passed with at least 14 out of 16 correct.


----------



## board

Pro-Jules said:


> _*2) The anti-aliasing filter that is basically the cut off point/slope that is used to determine sample rate affects the audio we hear. So for 44.1kHz the slope is at 22.05kHz and cuts off (quite abruptly) anything over this. No problem cos we're told we only hear to 20kHz right, and the Nyquist Theory suggests we need twice the frequency (which would be 40kHz) to capture what we can humanly hear correctly. Therefore 44.1kHz gives us this plus a couple more frequencies (22.05kHz technically).
> 
> But what you get is reflected back harmonics into the audible range that don't sound right in 44.1kHz, compared to 96kHz where the reflected back harmonics are above our hearing range. So with 44.1kHz audio, a 20kHz harmonic gets reflected back at 10kHz due to the 22.05kHz slope. We can hear 10kHz so this is not favourable. If the sample rate is 96kHz the slope is at 48kHz, so the lowest reflected back harmonic ends up as 24kHz which is above our hearing range - enough for it not the be a problem. So when you compare the two 44.1kHz vs 96kHz it is hard to determine exactly why we prefer it, but you get something 'clearer' or 'truer' in the upper range, bringing transient clarity that excites, to my ears.
> 
> Finally digital converters just sound better at higher frequencies despite not hearing the information captured above 20kHz or so. I've conducted the test countless times. I don't care to argue if someone believes I'm wrong - it's obvious with the right system and ears in my opinion.*_



If I understand the point he's trying to make correctly, then what he's trying to say is that there are no anti-aliasing filters in DACs. This is only true if you buy one of those NOS DACs that purposely leave out filters. These DACs/CD players are preferred by some audiophiles, but have by some people subjectively been described as having a "dirty" sound.
It should be noted, however, that some DACs do have anti-aliasing filters but let a lot of ultrasonic content through, but this is simply due to faulty engineering. Often these DACS are the ones costing extraordinary amounts of money. Maybe this faulty technology is what makes audiophiles rave about how it sounds more "analogue" (meaning imperfect).

Your friend's final point about how converters sound better at higher frequencies is again just talk, unless he's willing to submit a log from a passed blind test (or take one, since he probably hasn't taken one).
As one of the other participants in this discussion has said several times: What is asserted without proof can be dismissed without proof.
Submit your logs of your passed blind tests, both of you (you and your friend), please, because no one in this discussion who claims to hear the audible superiority of 24 bit over 16 bit has offered any proof other than stubborn claims to know what they heard.


----------



## board

TheSonicTruth said:


> Yet all the credible controlled test conducted over the past 30 years do not support this conclusion, like the decade old test below. I don't doubt your engineer friend hears a difference - it is either different masterings, volume levels or most likely, placebo.  There are no other credible explanations given what we know about human physiology and psychology, digital audio and its measurements and decades of controlled tests.
> http://drewdaniels.com/audible.pdf
> 
> Ian's technical series is exquisite and educational - but - I find him to be a bit hypocritical in his crusade to both fight the loudness war and promote dynamic range.  He supports amd uses tools, as I do, such as the TT Dynamic Range Meter and Foobar 2000.  You know, the ones that return a figure of "DR6"(squashed) or DR14(dynamic and open).   His threshold for promoting dynamics in the popular genres was, at least a few years ago, DR8 by the scale of those plugins.  I even argued that he should set the bar higher, DR10 maybe, but he insisted, baby steps.
> ...



I once read a mastering engineer say that if he had an artist asking for a squashed and loud mastering he would then make two masterings: One the way he wanted it to sound; and then the same but squashed. Then after compressing it, he would lower the volume of the loud master to match the non-compressed one and burn those two CDs for the artist to compare. He then said that the artist always chose the non-squashed mastering.
I know that procedure would waste time, but it's an attitude that I wish more mastering engineers would have instead of just assuming that the artist want something loud and aggressive.
I do remasters for fun, and if I would play my remasters for the artists, then I think many of them would like them (but this is of course speculation), but I also think that they would realize that they had an idea of a certain aggressive sound that would sound good, but when they hear my mastering (or any good mastering by a professional) they would say "Aaaaah! This sounds much better than before!" and be thrilled. Again, it's of course speculation, but I'm hopeful - we sometimes think something sounds great until we hear something that sounds better .


----------



## board

castleofargh said:


> maybe it's about hearing abilities, maybe it's about listening skills, maybe it's about the equipment. but maybe it's made up stuff in the mind of a listener who never bothered to test his hearing ability properly. we could really stop wasting so much time and efforts if that last possibility was cleared by the people themselves before coming here to spam their overconfident claims based on garbage testing methods.
> 
> it is a fact that 16bit is more than necessary under most circumstances. the debate only concerns niche cases and those who say otherwise are wrong. that much has been well established by decades of trials and I'm still waiting to see a legitimate research suggesting otherwise.


AGREEEEEEEEED!!!!!!!!!!!



castleofargh said:


> I would argue that very few people on very rare occasions end up with musical content sounding audibly different because it has more than 16bit. and I would argue that among those, probably more than half get sound differences that have nothing to do with having higher fidelity. instead it's often about the master being different or the playback gear doing some crap when fed with some particular resolution. the legitimate cases remaining, where audibility correlates with the quantization noise going down so bit depth is the relevant factor, I would be surprised if we can find a dozen on the entire forum. and I'm confident that all of them listen to music too loud, or created the circumstances to achieve audible difference(purposefully or by malpractice, like having the digital volume on the computer at -80dB and compensating with the amp or whatever). I'm very confident about that and after all those years hanging around audiophiles, I have yet to see one solid counter example.
> the hundreds or thousands of people who "know what they're hearing" under sighted conditions might contain such counter examples. I can't know that when those never demonstrate their abilities. to me they're no different from guys saying they have seen flying saucers from mars. some could be correct. but in the absence of proper demonstration, we all save time treating the all group as making stuff up. it's just the most pragmatic conclusion. if we consider that this is the "sound science" section, no scientific research would draw conclusions based on knowing a guy who claims he can do it. facts are demonstrated, they're not acts of faith.


EXACTLY!
Maybe the rest of you have heard about the experiement where they played people the same song two times in a row and a whopping 76 % preferred one over the other.
Every time I have had similar arguments with audiophiles, whether it's about bit depth, sample rate, jitter, mp3 vs. wave, etc. it's the same story EVERY SINGLE TIME: The person did not take a blind test and refuses to do it and prefers to yell and scream at me instead.
You might know the Bertrand Russel quote:
"The whole problem with the world is that fools and fanatics are always so certain of themselves, and wiser people so full of doubts."

I can add to that that wise people who are confident that they're right, or at least on to something, are too polite, permissive, agreeable and open to debate, whereas the fools and fanatics refuse to meet halfway (which in this discussion would be: "Yes, you can make up your mind by listening, as long as it's under controlled conditions, meaning blind"). The wise, and right, people, need to learn to make the fools shut up and then educate them instead of saying "you're welcome to think whatever you want - I'm not stopping you", which is an attitude I've come to despise more and more as time has passed by. I use to hold that position as well, and it's good in many ways, 'cause I don't advocate other countries or anything for being "wrong". But there are limits. If we don't educate people properly and we permit them to believe whatever they want, they will spread this misinformation to other uneducated people, who will spread it to other uneducated people who will spread it to ...


----------



## board

ALRAINBOW said:


> Come to my place. I won’t yell or attack you lol. I’m simply saying I can and to add to this plight for you guys I have a few buds who can too. One is a pro who makes music.


I live in Spain, so I'm not coming.
If you're so intent on doing it in person you can come here (seriously).
But we've all proposed an easier solution for you: Convert a 24 bit file to 16 bit, do an ABX test and post the log here.
And you didn't answer my question. I assume you probably won't this time either, but I'll repeat it for you:

What, if anything, can make you change your mind about your claim that you can hear a difference between 16 and 24 bit?
Or to be more specific, what, if anything, can make you change your mind about your claim that you actually heard audible differences, and can demonstrate it in a blind test, between those particular 16 and 24 bit files that you claim to have listened to?


----------



## ALRAINBOW

Here is my point set up a google share point or any way we can share some files 
Give me the files to comment on 
I promise to do the test as given. I think many here should participate in this. set my rules to follow I would love this really would. I’m Human as  such not perfect period. Please do this. Of you wish I’ll set up the share file location even if we email it. 
Lets do this please


----------



## board

ALRAINBOW said:


> Here is my point set up a google share point or any way we can share some files
> Give me the files to comment on
> I promise to do the test as given. I think many here should participate in this. set my rules to follow I would love this really would. I’m Human as  such not perfect period. Please do this. Of you wish I’ll set up the share file location even if we email it.
> Lets do this please


We will do this on one condition, and only on this one condition: That you use Foobar's ABX plugin to do an ABX test with 16 trials (or more if you wish) and that you post your log and provide other kinds of evidence that we might ask for in case we suspect foul play (although by using Foobar's ABX plugin there most likely won't be anything to ask for).
WE ARE NOT ASKING FOR YOU TO COMMENT ON AAAAAANYTHING! We want you to take an ABX test, not TALK!


----------



## ALRAINBOW

Lol as you wish. Love to.


----------



## board

Then install Foobar and its separate ABX plugin and report back when that is done.


----------



## taffy2207

board said:


> But we've all proposed an easier solution for you: Convert a 24 bit file to 16 bit, do an ABX test and post the log here.
> And you didn't answer my question. I assume you probably won't this time either, but I'll repeat it for you:
> 
> What, if anything, can make you change your mind about your claim that you can hear a difference between 16 and 24 bit?
> Or to be more specific, what, if anything, can make you change your mind about your claim that you actually heard audible differences, and can demonstrate it in a blind test, between those particular 16 and 24 bit files that you claim to have listened to?





board said:


> We will do this on one condition, and only on this one condition: That you use Foobar's ABX plugin to do an ABX test with 16 trials (or more if you wish) and that you post your log and provide other kinds of evidence that we might ask for in case we suspect foul play (although by using Foobar's ABX plugin there most likely won't be anything to ask for).
> WE ARE NOT ASKING FOR YOU TO COMMENT ON AAAAAANYTHING! We want you to take an ABX test, not TALK!





board said:


> Then install Foobar and its separate ABX plugin and report back when that is done.



You may want to consider climbing down from your Soapbox.

He's not answerable to you. You may want to consider working with him in a friendlier manner and dropping the dictatorial tone.

The whole point of an ABX (should be) is to find out what your personal hearing threshold is, he should consider doing it for himself not for the Community or any higher purpose. He has everything to gain by doing it for himself and nothing to lose. If he does an ABX and chooses to publicize his results, good for him. If not, also up to him.

His claims have already been both debunked and beaten down but it's up to him, it's his choice whether he does an ABX or not and it's also up to him if he shares those results IMO.


----------



## ALRAINBOW

Guys I’ll do as asked in fact I look forward to it. I’m no audio god of hearing. I fo commit to the challenge but also want a simple file playback too. Remember I simply say it’s what I observe and while I get kicked here I have many who also lurk  here and laugh at you guys confused conjecture all not based on the one thing we all do play music. One can try to prove what it sounds like with math but since there are many variables from file to the brain some inconsistency is not acknowledged in a pure math conclusion.  But to put this Aside I’ll do as asked. After this what is next ?


----------



## board

taffy2207 said:


> You may want to consider climbing down from your Soapbox.
> 
> He's not answerable to you. You may want to consider working with him in a friendlier manner and dropping the dictatorial tone.


Nope .


----------



## ALRAINBOW

I wonder who’s head will explode first


----------



## bigshot

ALRAINBOW said:


> After this what is next ?



It’s actually very simple to find out if there are differences. You take your favorite HD audio track, convert it to 16/44.1, then back up to HD again. Now you have two files- one is HD and the other is 16/44.1 bumped up to HD. Normalize the two tracks to the exact same volume level. Run the two tracks through Foobar to do a blind comparison and let us know how you do.

We are more than willing to help you do this. But we know how it is going to turn out, because we’ve done this ourselves and we know the difference isn’t audible. If you can consistently hear a difference, the world of digital audio would be interested in studying you and finding out why your ability to hear exceeds the rest of the human race.

Conduct the test fairly and have fun. You will almost certainly be surprised by the results.


----------



## ALRAINBOW

Ok help me it’s installed. Wish me luck lol. Why food bar anyway ? I assume I use foo st to down sample ?


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## ALRAINBOW

Ok question how do you select the output device I’m using a Lampi need amanaro right now it’s using windows it’s horrible help me. It’s been years since foobar.  Help me baby steps remember ima n idiot so need help lol.


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## ALRAINBOW

Ok must have a few bad tracks now playing from internal d drive sounds pretty good 
But why are some tracks louder then others of same album ? It’s albums I dod some format conversions on. ? Wav , aiff and stock pcm cd rip. 
? Any ideas ? the wav seems to have much more body 
Ok since I’m getting crickets I’ll do the high def and down sample too and report does it matter for what format I use it will be the same for both


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## bigshot

Outline the steps you took clearly.

how did you prepare your two files?


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## ALRAINBOW

Need dB power amp guys unless you can share how down sample in foobar. J river only lets me lower the bit depth not rate 
Help help lol. And yes I’ll say exactly how it’s done


----------



## Daiyama

Short question (sorry if this has been answered with the 349 pages):
Why for over a decade now 24 bit vs 16 bit is discussed here, when this is something that is usually not changed independent from the sampling rate?
24/96 is a way more common format than 24/44(48), personally I have seen this with amazon ultra HD.
So why not discussing 16/44 vs 24/96?
Thanks!


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## ALRAINBOW

It’s why I’m sampling now I found a way to use foobar to down convert from 24/96 to 16/44.1 now playing samples now.


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## taffy2207 (Nov 30, 2019)

@ALRAINBOW   if you get stuck, this post by @Brooko  may help :-

https://www.head-fi.org/threads/flac-vs-320-mp3.570621/page-40#post-14953380


----------



## sander99

ALRAINBOW said:


> I found a way to use foobar to down convert from 24/96 to 16/44.1 now playing samples now.


Don't forget to up convert the 16/44.1 to 24/96 again and to normalise to the exact same volume, like bigshot wrote:


bigshot said:


> You take your favorite HD audio track, convert it to 16/44.1, then back up to HD again. Now you have two files- one is HD and the other is 16/44.1 bumped up to HD. Normalize the two tracks to the exact same volume level. Run the two tracks through Foobar to do a blind comparison and let us know how you do.


@ALRAINBOW: Do you understand why?


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## bigshot (Nov 30, 2019)

It's important that you do this step by step. If you let a mistake slip in, it can invalidate your test. It would also help if you wrote in complete sentences so we don't have to puzzle out what you are trying to say.

If you don't know how to prepare your test files, I'm sure someone will volunteer to give you an A and B file to use.


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## ALRAINBOW (Dec 1, 2019)

Thanks I think I got this nailed eating now then I’ll play. I can say I like j river better in interface
But why would I up convert a lower resolution file ? It has Jess info so how can it be correct.
I assumed a hi Res file as one sample and use this file to down sample n compare. ? Yes it no ? I do get shy if your train of thought is to prove no change as you guys say it will be. This seems true by method used. But my pint is use Hd file sample A 
Down sample it as fine B


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## bigshot

I have no idea what you are talking about. Is there a language problem here or are you just posting too fast?


----------



## sander99

ALRAINBOW said:


> But why would I up convert a lower resolution file ? It has Jess info so how can it be correct.


If you directly compare the 96/24 file with the 44.1/16 file there is a possibility that something in your system handles the files differently and that could cause you to hear a difference.
In other words: if you hear a difference this way then it is not certain that it is because there is less info in the 44.1/16 file.

So what should you compare with eachother:
1. The original "real" 96/24 file
2. The "fake" 96/24 file that you create from the original 96/24 by first down converting to 44.1/16 and then up converting again to 96/24.

File 2 would indeed have less info. Only the info that was left in the 44.1/16.


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## Daiyama

Ok, scientifically this is the correct procedure.
(Is normalisation always necessary and if so how do I do that?)

But from a practical view, if the 24/96 file sounds better than the 16/44.1, I do not need to care to much why this is the case (because of the digital format or different handling of the file by my DAC), right?

And if I hear a difference (better?) at 24/96 and both formats are sonically equal, it would make sense to up convert all my 16/44.1 files to 24/96 to take advantage of the different (better?) handling of these files, right?


----------



## old tech

Daiyama said:


> Ok, scientifically this is the correct procedure.
> (Is normalisation always necessary and if so how do I do that?)
> 
> But from a practical view, if the 24/96 file sounds better than the 16/44.1, I do not need to care to much why this is the case (because of the digital format or different handling of the file by my DAC), right?
> ...



If you hear the difference in a controlled test, then I suppose why not. It might be easier upsampling your 16/44 files rather than working out why your DAC does not handle 16/44 properly.

However, if the difference is due to expectation biases, ie you haven't gone to the trouble to test that that there is a real difference with a controlled test, then that is not a good way to go about it because it is better to address your biases through controlled testing (so your brain resets) as it can lead to greater enjoyment of music that is more enduring.


----------



## Daiyama

Yes, you are absolutely right. At the moment I think I hear a very very subtle difference when I compare 16/44.1 to 24/96 openly (mostly in the high end) using my set-up. So I have to confirm this by doing it blind, which is a bit more of a hassle, but I will definitely do that.


----------



## ALRAINBOW

Daiyama said:


> Yes, you are absolutely right. At the moment I think I hear a very very subtle difference when I compare 16/44.1 to 24/96 openly (mostly in the high end) using my set-up. So I have to confirm this by doing it blind, which is a bit more of a hassle, but I will definitely do that.


Thanks for your honesty some files are more obvious and some no change is heard by me. keep in mind the files origin here many HD tracks as an example are indeed. Not true hi def files. 
This can be confirmed with analyzing software. 
One such album I love the sound of is Doug mc cleod it’s a hd download sourced from red book but has been modded 
Below is the album


----------



## ALRAINBOW

This album sounds great but is a fake lol. I’ll do as asked. For the poster who does hear change possible it’s in the high freq but I feel it’s a lower noise floor allowing more details. I’ll be doing a share file post soon to share some files I know show this. 
there are free downloads from Kent moon 
Audio Phil’s Jazz prolog all sourced from high def but lower resolutions are also available there too. . Again it’s there. Also the file type as well. Aiff or wav or wav 64 all have more decay details where flac is smoother. 

my more complete list of audio stuff is
Infinty irs speakers all updated 
Mark Levinson no 33 amps 
Mark Levinson no 26 preamp 
Mark Levinson no 25 phono preamp 
Turn tables 
Thorens td 124 -125 both sme 3009 arms 
New sota nova all options with eminent technologies et 2 arm 
Server is a custom 3 pc setup made by me 
Dacs one SS dac and two LAMPI dacs.


----------



## bigshot

I see red flags. When people talk about opening up files to analyze the wave form, I don’t trust them to conduct their own listening test. I’m bowing out here.


----------



## old tech

ALRAINBOW said:


> Thanks for your honesty some files are more obvious and some no change is heard by me. keep in mind the files origin here many HD tracks as an example are indeed. Not true hi def files.
> This can be confirmed with analyzing software.


The only true hi res recordings would only be a couple decades old, either DSD or 24 bit recordings.  Remasters or copies of analog material is not hi res and can never be high res.  It is the same argument as upsampling 16/44 to 24/96 and calling it high res however rather than software, you'd need to look at what was the format of the original recording and the processes used up to making the final product.


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## bigshot (Dec 1, 2019)

If the way he tells a true HD file is by analyzing the wave form, then he is admitting that he can't hear a difference.
If he can't tell that analogue masters don't contain anything beyond 16/44.1, he is admitting it again.
He says that CD quality sound can be modded to sound as good as 24/96... yet another admission.

I think he has no interest in knowing the truth of the matter. He is just here for idle chatter like the others of his ilk. He'll putter around in circles and make Gregorio repeat himself until he gets frustrated again. I don't feel inclined to waste energy helping people like that. I'd rather help people who care about sound quality and want to understand how to achieve it.


----------



## castleofargh

bigshot said:


> If the way he tells a true HD file is by analyzing the wave form, then he is admitting that he can't hear a difference.
> If he can't tell that analogue masters don't contain anything beyond 16/44.1, he is admitting it again.
> He says that CD quality sound can be modded to sound as good as 24/96... yet another admission.
> 
> I think he has no interest in knowing the truth of the matter. He is just here for idle chatter like the others of his ilk. He'll putter around in circles and make Gregorio repeat himself until he gets frustrated again. I don't feel inclined to waste energy helping people like that. I'd rather help people who care about sound quality and want to understand how to achieve it.


On the other hand it would be meaningless to convert something that's not hires and then abx it. I don't see the harm in being cautious about the file used for a test. Sure it does suggest that the hires sound improvement might not be as obvious to him as he claimed(something we all knew already). And of course if he wants to fake his test, he can. There is nothing we can do about that. But if we already decided to distrust anything he will say or do, then we're the ones wasting his time by asking him to run a blind test.
To me anybody who is willing to bother with a blind test, deserves some extra respect. It's already a big step for an audiophile to pull his fingers out of his own butt and spend time preparing files, app, and then running a few trials. Most will never get that far.


----------



## ILikeMusic

castleofargh said:


> To me anybody who is willing to bother with a blind test, deserves some extra respect. It's already a big step for an audiophile to pull his fingers out of his own butt and spend time preparing files, app, and then running a few trials. Most will never get that far.



In most of these 'Hi-res' disputes I haven't gotten that far because I don't need to. If all available science and theory demonstrates that it's virtually impossible for a human being to detect a difference then... I'm a human being, so not much point in testing.


----------



## ALRAINBOW

Oh geezzz u guys keep beating the same drum  
I have files I know are true hi res I’ll be doing it tomorrow some time and I’ll post Results.


----------



## ALRAINBOW

sander99 said:


> Don't forget to up convert the 16/44.1 to 24/96 again and to normalise to the exact same volume, like bigshot wrote:
> 
> @ALRAINBOW: Do you understand why?


Actually it makes no Sense to do so but I’ll. Do as asked. Can you explain the point to do so. min theory it should have no impact on sound as such conclude the wrong not being able to hear a change. 
my point is to hear a 24/96 and down sample to redbook this is obvious as is red book stock and 24/192 as well. In addition even 16/44.1 to 24.88.2 is easy to hear


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## SoundAndMotion (Dec 2, 2019)

@ALRAINBOW

The reason to upsample to 24/96 is simple. At least some versions (don't know which you have) of the foo_ABX plugin, produce a noticeable pause when you switch from one sample rate to another. So you could keep track of whether the pause is short (same sample rate) or long (different sample rate). If you use a good sample rate converter, there is no harm in upsampling. To keep future skeptical questions to a minimum, just upsample.

You may want to think of the possible outcomes from continuing with the ABX, and be prepared. I can think of 3, each with pros and cons for you. For each outcome, you can report it here or not.

If you decide not to proceed, most people will wonder why, and you'll be the subject of ridicule (being chicken, etc.). But most of those who would ridicule you already are, so it may be a push. That's one outcome.

If you fail to pass (generally considered > 5% chance of guessing), you can repeat the test with another track or other conditions, or just give it another try. But you may also be convinced that in the past you really couldn't "hear" a difference, but rather had "perceived" one due to biases. Then some here will say "I told you so", but many will welcome you into the fold of the converted. Many people here (and on other sites) have had such an experience and may tell you about it. That's the second outcome.

The third outcome is one you really need to consider. What if you pass the test? Most (or all?!) of those pushing you to do the test will _not_ accept this outcome! They will be certain that you_ either cheated or screwed something up_. You have to know that saying "I didn't cheat" won't work. Both people who cheat and people who don't, say that. There may be some effort to figure out how you cheated or how you screwed up, but some simply won't accept a pass. You may want to ask beforehand, something that any honest science-oriented person can answer: Under what circumstances/conditions will you accept a passing score (e.g. < 5% chance of guessing)? Everyone who says "it won't happen", "it can't happen", "it's not possible, so I don't have to answer that", will never be satisfied. Know that ahead of time.


----------



## old tech (Dec 2, 2019)

SoundAndMotion said:


> @ALRAINBOW
> The third outcome is one you really need to consider. What if you pass the test? Most (or all?!) of those pushing you to do the test will _not_ accept this outcome! They will be certain that you_ either cheated or screwed something up_. You have to know that saying "I didn't cheat" won't work. Both people who cheat and people who don't, say that. There may be some effort to figure out how you cheated or how you screwed up, but some simply won't accept a pass. You may want to ask beforehand, something that any honest science-oriented person can answer: Under what circumstances/conditions will you accept a passing score (e.g. < 5% chance of guessing)? Everyone who says "it won't happen", "it can't happen", "it's not possible, so I don't have to answer that", will never be satisfied. Know that ahead of time.



What's with the pessimism?  Regardless what anyone on this forum may think, if the third outcome comes to pass and it can be shown to be valid think of all the positives.

Firstly, Alrainbow might be able to take this test further afield and disprove the year long, multi subject, multi source, multi listening environment 10 years' Meyer and Moran test (and the many controlled tests before it) and now some 30 years after the marketers called studio work files as "hi res" for gullible audiophiles to repurchase their music collection, finally provide some compelling evidence that high res on its own does indeed result in sound differences for consumers.  Perhaps it still won't convince those that could not pass these tests and they will think that alrainbow is a one of kind mutant.  Even so, if Alrainbow can positively hear a difference with files, particularly with ones that originally were analog recordings, that is a double outstanding human achievement - somehow he can hear standard res as hi res too. Nearly all universities have been looking for human freaks so he will be in a very high demand everywhere. 

So I have to disagree with you as I can see nothing but positives for Alrainbow, the advancement of human knowledge and understanding (perhaps even the occult) and research material to keep audio science faculties busy for decades.


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## Daiyama

I sense a little bit of irony here.


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## SoundAndMotion (Dec 2, 2019)

old tech said:


> What's with the pessimism?  Regardless what anyone on this forum may think, if the third outcome comes to pass and it can be shown to be valid think of all the positives.


No pessimism, just experience reading several audio forums. Ignoring the rest of your clown-ish post (you kind of make my point for me), why don't you go ahead and state clearly under which circumstances you'll change your mind? What if @ALRAINBOW gets 8/8 correct? 12/14? 20/20 on three separate occasions, using only the methods offered so far recently in this thread?
None of those would convince me. What about you? What would?
@old tech , why should he do anything with the possibly that a lot of effort will convince no one?


----------



## Hifiearspeakers

SoundAndMotion said:


> No pessimism, just experience reading several audio forums. Ignoring the rest of your clown-ish post (you kind of make my point for me), why don't you go ahead and state clearly under which circumstances you'll change your mind? What if @ALRAINBOW gets 8/8 correct? 12/14? 20/20 on three separate occasions, using only the methods offered so far recently in this thread?
> None of those would convince me. What about you? What would?



I like it when people keep it real and are honest. 

+1 for you sir.


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## ALRAINBOW

You guys are childish really yiu want me to make a comment on a test that must prove your point lol. You can’t hear a change if there is none and if I upsample a fine to hi res it can’t show me one how dumb is this thread. 
uet you jackals atatck me it’s pretty sad


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## SoundAndMotion

ALRAINBOW said:


> You guys are childish really yiu want me to make a comment on a test that must prove your point lol. You can’t hear a change if there is none and if I upsample a fine to hi res it can’t show me one how dumb is this thread.
> uet you jackals atatck me it’s pretty sad


I admit I really struggle to understand your posts... but that begs the question: how well do you understand what others are writing? I suspect: what we've got here is failure to communicate. (Ref. Cool Hand Luke)


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## ALRAINBOW

Hahaha it may be just that.  A great movie and ref. 
Ty.


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## sander99

ALRAINBOW said:


> You can’t hear a change if there is none and if I upsample a fine to hi res it can’t show me one how dumb is this thread.


There definitely is a language problem. What is "a fine" in "if I upsample a fine to hi res"? What is "it" in "it can't show me one"?

I suspect @ALRAINBOW your English is at a level that you don't understand half of what is written in this thread. This is not meant as an insult, there are of course many languages that I don't speak. But please at least try to type carefully, it is hard enough to follow what you write without the typing errors.
Maybe you should find someone that speaks your language very well and speaks English very well to carefully translate some of the posts for you. For example the posts (at least three) that explain what the upsampling of the 44.1/16 is good for.


----------



## taffy2207

File


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## SoundAndMotion

@ALRAINBOW , @sander99  è giusto. Hai davvero bisogno di aiuto con il problema della lingua. Se questa traduzione ha senso in italiano, prova Google Translate. Sarebbe davvero uno spreco di tempo (e altri) continuare senza una migliore comunicazione.

Translated from: @ALRAINBOW , @sander99  is right. You really need help with the language issue. If this traslation makes any sense in Italian, try Google Translate. It would really be a waste of your time (and others) to continue without better communication.


----------



## chef8489

Maybe he is typing on his phone at times. I know its a pain the arse sometimes when doing so and phone takes over.


----------



## miksu8

So in theory 16 and 24 bits sound the same, but how is it in practice? What is the likelyhood that the 24bit version of some record sounds better because of better production?


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## bigshot (Dec 2, 2019)

miksu8 said:


> So in theory 16 and 24 bits sound the same, but how is it in practice? What is the likelyhood that the 24bit version of some record sounds better because of better production?



That is totally possible, and it is just as possible that the 16 bit sounds better than the 24 too. I have albums that sound best on LP and others sound best on CD and others sound best on SACD. And I have albums that are the exact opposite. The quality of the mix and mastering is what matters, not format.



castleofargh said:


> of course if he wants to fake his test, he can. There is nothing we can do about that. But if we already decided to distrust anything he will say or do, then we're the ones wasting his time by asking him to run a blind test.
> To me anybody who is willing to bother with a blind test, deserves some extra respect. It's already a big step for an audiophile to pull his fingers out of his own butt and spend time preparing files, app, and then running a few trials. Most will never get that far.



Exactly, and I'll go the extra mile and help someone do their first controlled test if they are interested in finding out the truth for themselves. I've done that before. I gave this guy a chance because he said he wanted to do a test, but he's proven himself to be not worth my time. Now that I've dismissed him, he is blathering on with the troll speak... jackals. I'm done with people like this. I've wasted way too much time on them, and they scare away people who have honest questions. They should be given a chance, and if they continue to cause trouble like this, they should be weeded out.



SoundAndMotion said:


> What if you pass the test? Most (or all?!) of those pushing you to do the test will _not_ accept this outcome!



I'm already convinced he is going to cheat. He has already told us how he is going to do it too. At this point, I won't pay any attention to any "results" he gets. No, 10 out of 10 will not convince me.

If he wants to do this, he should either just do it for himself and know that he cheated to achieve the results he wanted (a total waste of his time, not mine), or he should look for someone impartial to administer the test to him, so he isn't suspect.

It isn't my fault that he is discrediting himself. I'm not being unfair if I don't acknowledge "results" when I am doggone sure that they didn't come from an honest place.

I will say this... This thread and the Testing Myths thread are the ones that always seem to be the target of trolls. There is a reason for that. The top posts in these threads are smoking guns showing that cherished audiophile concepts are not based in reality. There are people who would love to see these threads get all mixed up in the comments and eventually locked. Those people don't belong in the Sound Science forum. There's a whole world of hoodoo outside of here for them to play their smoke and mirrors games.


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## castleofargh

ALRAINBOW said:


> You guys are childish really yiu want me to make a comment on a test that must prove your point lol. You can’t hear a change if there is none and if I upsample a fine to hi res it can’t show me one how dumb is this thread.
> uet you jackals atatck me it’s pretty sad


 Is this some reverse psychology trick where you hurt yourself so hard that others will be tempted to leave you alone?
In any case, careful with the insults.


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## old tech (Dec 2, 2019)

SoundAndMotion said:


> No pessimism, just experience reading several audio forums. Ignoring the rest of your clown-ish post (you kind of make my point for me), why don't you go ahead and state clearly under which circumstances you'll change your mind? What if @ALRAINBOW gets 8/8 correct? 12/14? 20/20 on three separate occasions, using only the methods offered so far recently in this thread?
> None of those would convince me. What about you? What would?
> @old tech , why should he do anything with the possibly that a lot of effort will convince no one?


Reads just like a retort from an anti-vaxxer... It might be clownish to you but I, and I suspect most others that understand the science of audio and human physiology in this context would be excited if there was someone that could defy science and logic. And I don't get your point that it would convince no-one.  If he does hear a difference with all the variables other than bit depth and sample rate controlled (including upsampling the 16/44 file to 24/96 which he doesn't seem to comprehend why this is important to control another variable) then who knows where it would lead.  Surely others would be interested in replicating the findings and possibly change the paradigm?


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## bigshot (Dec 2, 2019)

Here is the thing about us suggesting that people do tests around here... We aren't looking for some scientific be-all and end-all intended for publication in a peer-reviewed journal. No one needs to convince anyone. We're recommending it so people can find out the truth for themselves. First, we let them know that just about every controlled test that has ever been done comparing 16 and 24 has ended up with no audible difference, then we offer our own experience doing tests for ourselves, then we explain the science behind audibility and show that there is no reason to believe anything above 16/44.1 is audible, then we patiently explain how bias, auditory memory and level imbalances can affect the results. If after all that, the person still clings to their belief that they can hear a "night and day difference" between 16 and 24, we offer to help them set up a controlled test to find out for themselves.

No one is required to prove anything. They should have the integrity and honesty to want to put their opinions to the test for their own sake. If they have no interest in learning, or if they don't have the integrity or honesty necessary to conduct a fair test, then I have no time for them. If they are unconvinceable, then fine. They can go through life ignorant of the truth. No skin off my nose.

However, if they choose to utter a falsehood, I will call them on it and tell them that they are wrong and remind them that they don't know what they're talking about. Just because they have an opinion and I have an opinion, it doesn't mean our opinions are equal. You find out whose opinion is right by putting them to the test. I'll give people a chance to either learn from us or teach themselves, but that is as far as I go. You can lead a horse to water, but you can't make him think.

"Being ignorant is not so much a shame as being unwilling to learn." -Benjamin Franklin


----------



## castleofargh

old tech said:


> Reads just like a retort from an anti-vaxxer... It might be clownish to you but I, and I suspect most others that understand the science of audio and human physiology in this context would be excited if there was someone that could defy science and logic. And I don't get your point that it would convince no-one.  If he does hear a difference with all the variables other than bit depth and sample rate controlled (including upsampling the 16/44 file to 24/96 which he doesn't seem to comprehend why this is important to control another variable) then who knows where it would lead.  Surely others would be interested in replicating the findings and possibly change the paradigm?


You guys are looking at the same thing from different angles. Obviously anything but a positive abx result would fail to shake the status quo. That's when things become interesting. When someone disproves the null hypothesis, it triggers the need for more questions, more tests to find out what specifically caused the audible difference. Even if it turns out that some device is defective or that a file was poorly manipulated(like we've seen a few times), at least something is happening and we can play detectives. In that respect you're correct IMO. A surprising pass would have a bunch of people curious to find out why and maybe discover something we didn't know.

On the other hand, to determine how conclusive an experiment will be, we need to be able to judge the degree of confidence we can have in it. And that would never amount to much with an online amateur ABX. In that respect I certainly agree with @SoundAndMotion. I will not change my belief about bit depth no matter the result of his test. I don't think it's a pessimistic or dishonest position. Just the logical outcome based on having too little control/documentation over the experiment and the experimenter to put our full confidence in the results.
Just one success can disprove the null hypothesis. Disproving an idea that way is what science does best. But that's assuming we can overwhelmingly trust the data. Which should never be the case with an online anecdote. The point of having @ALRAINBOW run this test isn't to change what the world understands about human hearing. It's to give him an opportunity to actually test himself. He's the one who can really gain from doing this. Us, not so much.


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## ALRAINBOW

How about few groupies come to my Secret lair in queens hahaha I have many toys to play with. bring your files and blind folds and I’ll feed you guys 
I have both digital and analog too. vinyl and tape too.  I’ll bet you guys will hear what I’m saying too.  
mad for what’s asked I don’t hear any change on my end and I did this with three files 
But I can’t get why I would hear a change is the hi res file is down sampled and back up again. 
The 24/96 file is not the same as 16/44.1 but it’s not as much as downloaded files in two formats 
Also do any of you guys have tidal ? They also have music at varying sample rates too. These also sound different as well.  
Please consider stopping by


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## TheSonicTruth

ALRAINBOW said:


> They also have music at varying sample rates too. These also sound
> different as well. Please consider stopping by



Different masterings, different sound.  Get a hold of the same exact mastering, both in 16/44.2 and 24/96 or higher format WAV, abx-test them, and THEN tell us you can hear a difference.


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## bigshot

ALRAINBOW said:


> How about few groupies come to my Secret lair in queens hahaha I have many toys to play with. bring your files and blind folds and I’ll feed you guys



Could this guy sound any more creepy? I think there is something wrong with him. He definitely doesn't know how he looks to other people.


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## ALRAINBOW (Dec 2, 2019)

bigshot said:


> Could this guy sound any more creepy? I think there is something wrong with him. He definitely doesn't know how he looks to other people.


Ok ur not invited now. Bro so when  one guy makes a reference to a movie it’s cool and when I do I’m weird bro grow up get over what ever major malfunction ur on lol. I’m truly being serous in the invite. To me I have much to learn on how we hear and why it sounds like it does. You seem to know all there is. Well at my place there is both. Don’t get creepy bro I’m too old to care how you feel about me but also old enough to cut introvert slackers some Space too. Take a chill pill or the blue pill I don’t care but chill.


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## bigshot (Dec 2, 2019)

Go away. It would be better for all of us. I have your best interests at heart here.


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## Daiyama (Dec 3, 2019)

TheSonicTruth said:


> Different masterings, different sound.  Get a hold of the same exact mastering, both in 16/44.2 and 24/96 or higher format WAV, abx-test them, and THEN tell us you can hear a difference.



Just to make sure.
The files linked in post #36 of this thread are fine for this purpose?

Edit: I refer to this source for the files: http://www.soundkeeperrecordings.com/format.htm

Thx.


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## gregorio (Dec 3, 2019)

miksu8 said:


> [1] So in theory 16 and 24 bits sound the same, but how is it in practice?
> [2] What is the likelyhood that the 24bit version of some record sounds better because of better production?



1. The practice is the same as the theory.

2. You seem to have answered your own question; If it "_sounds better because of better production_" then the reason it sounds better is because of the better production (not because it's 24bit). Let's take a hamburger analogy; let's say you've got a Big Mac (the master) and you serve this Big Mac in a 4" box (the bit depth). Here's 3 simple questions:
A. Will this Big Mac taste different if you serve it in a 6" box?
B. How about if you've got two different Big Macs, say one has got extra pickles and you serve them both in a 4" box, will they taste different now?
C. What about if you serve these two different hamburgers in different sized boxes, would it be nonsense/false to state that they taste different because they're in different sized boxes?



ALRAINBOW said:


> [1] mad for what’s asked I don’t hear any change on my end and I did this with three files
> [2] But I can’t get why I would hear a change is the hi res file is down sampled and back up again.
> [3] The 24/96 file is not the same as 16/44.1
> [3a] but it’s not as much as downloaded files in two formats
> ...



1. Exactly, you've DISPROVEN your repeated assertion that you CAN hear a change/difference!

2. Neither does anyone else, which is why we've been arguing with your repeated assertion to start with!

3. Correct, they are different. The one you've down and up sampled will be missing the extra 8bits (replaced with zero's) and also missing audio frequencies above about 20kHz but as both of these things are inaudible, you're not going to hear any change/difference, as you've demonstrated with your "three files"!
3a. Which indicates that the two files you've downloaded are different versions/masters and of course you should be able to hear a difference between different masters, in fact that's the whole point of making different masters in the first place, as already explained!

4. But you've stated there IS a change/difference, that "_the 24/96 is not the same as the 16/44.1_" and I've explained (in the previous point) what that difference is! Once you down sample to 16/44 those extra 8bits and frequencies above about 20kHz are gone forever, upsampling back to 24/96 doesn't magically regenerate them. You can check the data in the upsampled version yourself, it's just 8 added zero's and still no material above 22kHz.
4a. Very dumb indeed! 4b. Obviously we've attacked you, because you're the one who's made this thread "dumb" and what's "pretty sad" is you trying to make a thread in the sound science forum "dumb"!!!

5. I don't have tidal but if the music sounds different at different sample rates, how do you know that it's because of the different sample rate OR because they're different masters? Your experiment so far (with you "three files"), along with all the reliable science, indicates it's the latter! Therefore ....


ALRAINBOW said:


> [1] In addition even 16/44.1 to 24.88.2 is easy to hear ....
> [2] I’ll bet you guys will hear what I’m saying too.
> [3] For the poster who does hear change possible it’s in the high freq but I feel it’s a lower noise floor allowing more details.


1. Why don't you try your test with 16/44 and 24/88 (instead of 24/96)? If you want to save yourself the effort, the result will be the same, IE. It will NOT be "easy to hear", in fact you won't hear any change/difference!
2. True, we won't hear any difference either, unless of course we're comparing different masters!
3. But you've stated you couldn't hear a difference, so you must be describing the difference between two different masters!


ALRAINBOW said:


> To me I have much to learn on how we hear and why it sounds like it does.


Which is at the root of all your problems here! Why would you come to a Sound Science forum and contradict the science if you have "much to learn"? Shouldn't you learn BEFORE you contradict the facts/science and isn't it "dumb"/"pretty sad" to do otherwise?

G


----------



## ALRAINBOW

My only comment from the first post is simply this 
It’s obvious that a file at 24:96 sounds different then a file at 16/44.1. It’s my only point here but you guys ice skate around it. Why the title of 16/24 bit depth exploded. There is no myth to disprove you even just said they are not the same and yes I can hear it. It’s also why is used one file and down sampled it. So to dispel any mastering change. this is my only claim here so am I correct yet. This is a little boys club and yes I’m tying to learn but it seems you guys just pick at null points of mine only. So can anyone posting here claim to hear a 24/96 over a 16/ 44.1 file change.  If yes open your mouth and say it. If not then shut up and I’m correct.  And what is the myth that is exploded here please explain to me. I’m serous now.


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## ALRAINBOW

In reading the title post I see why you guys attack me. And I agree in theory it should not matter. but having said this the playback does seem to have a lower blacker noise floor. I don’t claim to know why nor accept is my perception correct. But I do ask here if anyone can post an answer to my comments by simply playing files. 
since this is where math meets practical let’s reply. Also GREG your a good poster.


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## SoundAndMotion (Dec 3, 2019)

@ALRAINBOW , when I first looked at your avatar closely, I laughed and thought it was clever. Now that I realize it's a real picture, I feel bad and want to send you a sympathy card...
LOL, just kidding...
Seriously though, you make the same comments again and again and again and...
You ask the same questions AGAIN and AGAIN and AGAIN and ...
You get the same answers, sometimes worded differently, AGAIN AND AGAIN AND AGAIN AND ....

Do you really think asking one more time will change anything? What do you really want?

Good Luck!!


----------



## ALRAINBOW

Lol I want an answer this will admit to me science meets factual observations. the pic is not me but I do own it. In audio life there are those who claim to know and those who do know. Like baseball either can be correct someone is also wrong. 
I agree in theory it should not matter but yet it does in practicality. The major malfunction all or most here have is to admit what they hear and accept it. So while you guys feel I’m wrong you skip past reality into pure science.  lastly one should never goof on someone’s possible mental acuity mishaps without knowing the person first hand.  
My avatar is exactly why I use it to fool the real fools.


----------



## TheSonicTruth

ALRAINBOW said:


> Lol I want an answer this will admit to me science meets factual observations. the pic is not me but I do own it. In audio life there are those who claim to know and those who do know. Like baseball either can be correct someone is also wrong.
> I agree in theory it should not matter but yet it does in practicality. The major malfunction all or most here have is to admit what they hear and accept it. So while you guys feel I’m wrong you skip past reality into pure science.  lastly one should never goof on someone’s possible mental acuity mishaps without knowing the person first hand.
> My avatar is exactly why I use it to fool the real fools.



I hope by "fools"(your words not mine) you don't include this gentleman: https://ethanwiner.com/

He will straighten you - and quite a few others on here - out, about a few many things...!


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## sander99 (Dec 3, 2019)

@ALRAINBOW: There definitely is a big communication problem. From what you write I do not understand what exactly did you do and what exactly was the result. Could you please answer the questions below.

Now, do I understand correctly that:
1. You toke a "real" high quality 96/24 file, let's call this file A?
2. You created from this file, by downconverting, a 44.1/16 file, let's call this file C?
3. You created from file C, by upconverting, a new 96/24 file, let's call this file B?
4. You compared all these 3 files with each other with blind ABX testing?
And what was the result?
5. Did you hear a difference between file A and file C?
6. Did you hear a difference between file A and file B?
7. Did you hear a difference between file B and file C?

[Edit: oh, I forgot: did you make sure the levels where exactly the same in all 3 files?]


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## gregorio (Dec 3, 2019)

So, apparently you're moving on from just making false assertions which contradict the science to even contradicting yourself now! For example:

"_It’s obvious that a file at 24:96 sounds different then a file at 16/44.1._" and "_mad for what’s asked I don’t hear any change on my end and I did this with three file ... But I can’t get why I would hear a change is the hi res file is down sampled and back up again._" - So which is it? Does a 24/96 file obviously/easily sound different to a 16/44 *OR* were you in fact NOT able to hear a difference "with three files" you have tested and don't understand why you would???


ALRAINBOW said:


> [1] In reading the title post I see why you guys attack me.
> [2] And I agree in theory it should not matter. but having said this the playback does seem to have a lower blacker noise floor.
> [3] I don’t claim to know why nor accept is my perception correct.
> But I do ask here if anyone can post an answer to my comments by simply playing files.


1. You're joking right? You've responded how many times to this thread and you've only just read the title post? That's nuts!

2. And again! You said you could NOT hear any change/difference and now you're saying you can hear a difference; "a lower blacker noise floor"! So which is it?

3. That's absolute nonsense,  you definitely accepted your perception is correct and you've even insulted the rest of us for not having the "ears" or "setup" to hear "your perception"!!


ALRAINBOW said:


> [1] I want an answer this will admit to me science meets factual observations.
> [2] I agree in theory it should not matter but yet it does in practicality.
> [3] The major malfunction all or most here have is to admit what they hear and accept it.
> [4] So while you guys feel I’m wrong you skip past reality into pure science.


1. Huh? The science "meets" the test you've done (where you couldn't hear any difference) and it "meets" countless other controlled listening tests, the Meyer & Moran published study for example (which has been mentioned previously and let me guess, you haven't bothered to read that either)? The science already meets factual observation, if it didn't, then it wouldn't be the science! How can you not know this, do you know what science is?

2. How does it matter in practice, you said you couldn't hear any change?

3. You mean YOUR major malfunction! You said you couldn't hear any change, do you accept what you yourself said or not?

4. What hypocrisy is this? You're "skipping past" both reality and science and even past what you yourself have said!!!

As is so often the case, making false assertions here in the science forum and then defending them ad infinitum from a position of ignorance, without any reliable supporting evidence ALWAYS results in self-contradictions and/or making ever more ridiculous or hypocritical claims. The only thing this all demonstrates is their own ignorance, foolishness and inability to think critically. Surely YOU don't want to demonstrate you're ignorant, a fool and can't think logically/critically, do you?

G


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## SoundAndMotion (Dec 3, 2019)

@gregorio  , take a chill pill!! I don't think you've understood @ALRAINBOW  , and therefore you're just arguing with yourself. Do you really think berating him will get clearer responses. Or is your goal to simply frustrate him beyond his limit, so he'll just go away?

My interpretation is that he says he hears a difference between hi-res and RBCD with files "as provided by ...?". When he downsamples and then upsamples, the difference disappears, which he thinks makes sense because he thinks _since it has been upsampled it is the same as the hi-res_, i.e. not different anymore. @ALRAINBOW : This is not true!!

@ALRAINBOW  If you have 2 files: Album5Track4_2496.wav and Album5Track4_1644.wav and you are certain you hear a difference (e.g. the first one clearly has a lower blacker noise floor), then I can only think of 3 possibilities:

1- You really don't hear a difference, but believe you do. It's a type of illusion, just like an optical illusion.

2- The 2 tracks are supposed to be exactly the same version, other than the first is high res., and the second is RBCD, but(!!) in fact the 2 are different (different masters, or someone did a flawed job of sample rate conversion, or different in some other way).

3- The 2 tracks really are exactly the same except for the sample rate/bit depth, and you have a special gift of amazing hearing. This would be unusual, but very exciting.

So, how to distinguish the 3 cases... (did I miss a possibility?)
A good idea that has already been suggested (and I thought you tried) is use foobar2000 with the foo_ABX plugin. If you need help, check back a few posts, or just ask again.

If you do the ABX, say 16 times, and the difference is not clearly detected anymore, then it seems to be possibility 1 above. Do you see it differently? If so, why?

If you clearly keep hearing the difference with the ABX method, then the question is how to tell the difference between 2 and 3 above. One idea, that would require knowing how and taking the time, would be to analyze the 2 files to see that they really are the same versions.

The simpler method is to do what sander99 wrote above (and others have mentioned in the last few pages): make your own 16/44 version to compare. This will eliminate option 2 (if you do it correctly). Do you know how? Do you have concerns about then upconverting the 16/44 file to a 24/96 file (sander99 's file B above)?


----------



## SoundAndMotion

@ALRAINBOW 
If you imagine that 24/96 is kind of like writing a number_* 98.765432*_, and writing the "same" number in 16/44 is *98.765 *(i.e. downsampling)
Then upsampling back to 24/96 is like _*98.765000 *_. We do this in case your software/hardware treats 8 digits a little differently from 5 digits.
Do you see that the upsampled is the same as the downsampled, except for the extra zeros? And they both have less resolution/precision than the original 24/96.


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## bfreedma

Whether intentional or somehow unintentional, there is clearly trolling going on.  

When posting history is reviewed, it becomes clear that this pattern has repeated itself over time.  Doubtful that anything is going to change based on responding.


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## SoundAndMotion

bfreedma said:


> Whether intentional or somehow unintentional, there is clearly trolling going on.
> 
> When posting history is reviewed, it becomes clear that this pattern has repeated itself over time.  Doubtful that anything is going to change based on responding.


That was an interesting skim... his English used to be better! Good tip.


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## board

bfreedma said:


> Whether intentional or somehow unintentional, there is clearly trolling going on.
> 
> When posting history is reviewed, it becomes clear that this pattern has repeated itself over time.  Doubtful that anything is going to change based on responding.


I've also used some strong language in this debate, which someone asked me to tone down.
I can include @gregorio here, I love your initial post and have linked to it several times, and I love all the knowledge you have on the subject, but what I'm slowly coming to realize is that for these types of discussions there are two groups of people (with somewhat fluid borders): 

1: People who are completely sure that what they believe in is the truth, and nothing or almost nothing can make them change their mind. This includes audiophiles who believe 24 bits sound better than 16 bits, and it also includes religous people, people who believe in dowsing, zodiacs, etc. and often build their entire lifestyle around this belief (going to church every Sunday/buying more products for their stereo systems, etc.).

2: People who believe in science, proof, and experiments, and who are happy to read scientific papers, listen to scientists explain certain issues, and they are happy to participate in experiments, such as blind tests, and throughout life they change their mind about quite a bit of topics. They also often, foolishly, think that you can reach people from group 1 with rational arguments and debate.

So the problem is when people from group 2 think they can convince people from group 1. They can't. We can't. We are from group 2, and Alrainbow is from group 1.
We've been arguing with him for a while now. Before him it was Michael Fremer we argued with. Before that it was someone else.
Unfortunately, some of us become a bit upset when the people from group 1 don't agree to do a simple experiment (blind test). Someone like Michael Fremer also refuses to do any kind of ABX tests ever again, because on that fateful day in 1991 Stanley Lipshitz cheated him out of his victory, and now Fremer is bitter (not that he wasn't before, but that's a different topic).
So the discussion is going nowhere, and people from group 1 will never, ever change their mind about their beliefs, unless they were already close to the border to group 2 to begin with. This is something about personality, not that we haven't presented the right argument or method, etc.
It should also be said that people who believe in science can actually belong to group 1.They have the attitude that we already know everything there is to know.


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## bigshot (Dec 3, 2019)

bfreedma said:


> Whether intentional or somehow unintentional, there is clearly trolling going on.



Scientific proof of the Dunning-Kruger effect.



SoundAndMotion said:


> Do you see that the upsampled is the same as the downsampled, except for the extra zeros?



I'm interested to see how low the game of limbo needs to go to get down to the right level.


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## Daiyama

I would be very great full if someone could answer my question in post #5277.
https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-352#post-15341001
Thank you
Kind regards


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## bigshot

Download their 24/96 file and knock it down to 16/44.1 yourself. Then bounce it back up to 24/96 and level match.

I wouldn't trust an audiophile record label to prepare the lower resolution file properly.


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## old tech

board said:


> Someone like Michael Fremer also refuses to do any kind of ABX tests ever again, because on that fateful day in 1991 Stanley Lipshitz cheated him out of his victory, and now Fremer is bitter (not that he wasn't before, but that's a different topic).


Tell me more...  I recall he piked out on James Randi challenge on cables but haven't heard of this one.


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## miksu8

gregorio said:


> 1. The practice is the same as the theory.
> 
> 2. You seem to have answered your own question; If it "_sounds better because of better production_" then the reason it sounds better is because of the better production (not because it's 24bit). Let's take a hamburger analogy; let's say you've got a Big Mac (the master) and you serve this Big Mac in a 4" box (the bit depth). Here's 3 simple questions:
> A. Will this Big Mac taste different if you serve it in a 6" box?
> ...



If you quote someone please don't change the original message. I just asked how often 24bit music sounds better because it's produced better. It's reasonable question, since if someone has a better quality version of a track, it's probably sold as 24 bit version for marketing reasons.

Your answer didn't make any sense to me, sorry.


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## gregorio (Dec 4, 2019)

SoundAndMotion said:


> [1] I don't think you've understood @ALRAINBOW , and therefore you're just arguing with yourself.
> [1a] Do you really think berating him will get clearer responses. Or is your goal to simply frustrate him beyond his limit, so he'll just go away?
> [2] My interpretation is that he says he hears a difference between hi-res and RBCD with files "as provided by ...?".
> [2a] When he downsamples and then upsamples, the difference disappears, which he thinks makes sense because he thinks _since it has been upsampled it is the same as the hi-res_, i.e. not different anymore.



1. I think I have understood him, as my interpretation is similar to yours (see point #2) but even if I haven't, it doesn't make a substantial difference because ...
1a. No, I don't think berating him will get clearer responses, we're past the point of clearer responses. Even if his responses become a little clearer as far as his use of language is concerned, they'll become less clear as far as a factual basis for his arguments and rational thought/critical thinking are concerned. And, neither is my primary goal to frustrate him so he'll just go away. My aim was to highlight the self-contradiction, hypocrisy and sheer ridiculousness of his responses. This might get him to evaluate his future posts more carefully (my primary goal) but probably not. He'll probably just ignore it completely, brush it off as me twisting his words or invent some other false excuse and just carry on regardless. There's also the possibility that others reading this thread might think that underneath the poor use of language, Alrainbow could have a rational/valid point. Highlighting Alrainbow's self-contradiction, irrationality (etc.) might make it easier for them to realise that he doesn't (my secondary goal). And lastly, as he descends ever deeper into his logical black-hole, he'll sooner or later reach a point where even he realises how ridiculous he's being or at least, that his arguments are futile. At which point he'll either leave the thread due to embarrassment or frustration or, there's a very small chance he might re-evaluate. Whichever it is though, his (deliberate or inadvertent) trolling/thread cr@pping will cease and the (third) "goal" of my post was to reach that point sooner rather than later!

2. That's my interpretation too but that just brings us back to my (and others') very first response to him, that he's falsely conflating different masters with different container formats. So, we're now going round in a circle of his making!
2a. No, he doesn't think they're the same, he specifically stated: "_The 24/96 file is *not the same* as 16/44.1 but it’s not as much as downloaded files in two formats_" [emphasis mine] - Clearly he's talking about the file he has converted rather than different masters and clearly he has somehow analysed the difference. Also clearly: If he thinks that upsampling the 16/44 will restore everything that downsampling removed, then firstly that's already been refuted and secondly, he can easily find out for himself by making exactly the same comparative analysis between his downsampled 16/44 and it's upsampled version as he did between his 16/44 version and the "downloaded files". Whichever way you look at his assertions, they are self-contradictory/nonsensical!



board said:


> 1: People who are completely sure that what they believe in is the truth, and nothing or almost nothing can make them change their mind. This includes audiophiles who believe 24 bits sound better than 16 bits, and it also includes religous people, people who believe in dowsing, zodiacs, etc. and often build their entire lifestyle around this belief (going to church every Sunday/buying more products for their stereo systems, etc.).
> 2: People who believe in science, proof, and experiments, and who are happy to read scientific papers, listen to scientists explain certain issues, and they are happy to participate in experiments, such as blind tests, and throughout life they change their mind about quite a bit of topics. They also often, foolishly, think that you can reach people from group 1 with rational arguments and debate.
> 
> So the problem is when people from group 2 think they can convince people from group 1. They can't. We can't. We are from group 2, and Alrainbow is from group 1.



I'm not sure that is the problem. I'm obviously a group 2 person and I DON'T think I can convince people from group 1 or rather, I think there is only a tiny chance I can convince them. The problem is sort of the reverse, group 1 people come here, to a group 2 (science/fact based) forum and try to convince us with IRRATIONAL arguments and debate. That approach can never work here, so why do they try it? It indicates an inability to differentiate the rational from the irrational, ignorance not only of the science specific to audio but of what science is and why it exists, arrogance and rudeness in trying such an approach in an actual science/fact based forum, hypocrisy in that they'll happily employ science when it suits them and, when applied to areas beyond/outside of audiophilia, increasingly represents an existential threat to our entire species (and many others)!

G


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## gregorio (Dec 4, 2019)

miksu8 said:


> [1] If you quote someone please don't change the original message.
> [2] I just asked how often 24bit music sounds better because it's produced better.
> [2a] It's reasonable question, since if someone has a better quality version of a track, it's probably sold as 24 bit version for marketing reasons.
> [3] Your answer didn't make any sense to me, sorry.



1. What exactly did I change?

2. What reason, apart from a better production (or better master), could there be for it sounding better, baring in mind that the whole point of this thread is a simple explanation of the science/facts for why 24bit does not sound better than 16bit?
2a. It would be a reasonable question, if there were a reasonable alternative (to being due to a better production/master). The only reasonable alternative is that two tracks (at say 24/96 and 16/44) are in fact the same but one is perceived by some people as better because of some expectation/cognitive bias but then, there can be no reasonable answer to your question, because you'd have to DBT a significant number of people with most of the available 24bit and 16bit versions (or all of them if you want a precise answer) and it's not practical/reasonable to do such a test/study.

3. I can't help that, sorry. Maybe if your question were more reasonable or you reworded it so it made more sense, then I could provide you with an answer that would make more sense to you?

G


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## castleofargh (Dec 4, 2019)

Daiyama said:


> Just to make sure.
> The files linked in post #36 of this thread are fine for this purpose?
> 
> Edit: I refer to this source for the files: http://www.soundkeeperrecordings.com/format.htm
> ...


While I understand the attraction for a ready made test file, I can only suggest to convert files yourself.
I see a few reasons for that: 
- you get to pick songs you know very well, and that could probably improve your chances to notice a change.
- You get to test your own conversion method. So if you find no audible difference, you gain confidence that your way of doing it is at the very least good for your ears. I personally find that reassuring given how many albums I've converted to 16/44 or lossy formats to put on my DAP.
- You make sure that nobody messed with the files to push their own agenda.

If whatever tool you use offers to apply dither in some way when converting from 24 to 16bit, you probably should do it. Chances are that the file will still be audibly transparent without any dither, but why take a chance?


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## taffy2207 (Dec 4, 2019)

gregorio said:


> 1. I think I have understood him, as my interpretation is similar to yours (see point #2) but even if I haven't, it doesn't make a substantial difference because ...
> 1a. No, I don't think berating him will get clearer responses, we're past the point of clearer responses. Even if his responses become a little clearer as far as his use of language is concerned, they'll become less clear as far as a factual basis for his arguments and rational thought/critical thinking are concerned. And, neither is my primary goal to frustrate him so he'll just go away. My aim was to highlight the self-contradiction, hypocrisy and sheer ridiculousness of his responses. This might get him to evaluate his future posts more carefully (my primary goal) but probably not. He'll probably just ignore it completely, brush it off as me twisting his words or invent some other false excuse and just carry on regardless. There's also the possibility that others reading this thread might think that underneath the poor use of language, Alrainbow could have a rational/valid point. Highlighting Alrainbow's self-contradiction, irrationality (etc.) might make it easier for them to realise that he doesn't (my secondary goal). And lastly, as he descends ever deeper into his logical black-hole, he'll sooner or later reach a point where even he realises how ridiculous he's being or at least, that his arguments are futile. At which point he'll either leave the thread due to embarrassment or frustration or, there's a very small chance he might re-evaluate. Whichever it is though, his (deliberate or inadvertent) trolling/thread cr@pping will cease and the (third) "goal" of my post was to reach that point sooner rather than later!
> 
> 2. That's my interpretation too but that just brings us back to my (and others') very first response to him, that he's falsely conflating different masters with different container formats. So, we're now going round in a circle of his making!
> ...



Let's not forget the monetary aspect as well. A lot of Audiophiles have invested heavily in Hi-Res Music. It's a bitter bill to swallow for them to admit that they could be wrong about it and thus potentially wasted their money.

It doesn't just apply to Hi-Res Music, unfortunately. It applies to gear as well, particularly people who bang on about 'endgame' set ups.


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## board (Dec 4, 2019)

Daiyama said:


> I would be very great full if someone could answer my question in post #5277.
> https://www.head-fi.org/threads/24bit-vs-16bit-the-myth-exploded.415361/page-352#post-15341001
> Thank you
> Kind regards


@gregorio might be able to answer your question if he sees your post, as he and *nick_charles* talked about music samples on the same page (page 3) of this thread. However, nick_charles erased his post, so I couldn't figure out if they were talking about the samples provided by Soundkeeper Recordings.
So, I downloaded the samples from the first songs Soundkeeper provided (Minis Azaka) and looked at them briefly.
I did the following:
* Using SoX resampler in Foobar (quality: best; passband: 95 %; phase response: 50 % (linear)). I downsampled the hi-res files to 16/44.1 and then I also upsampled the 16/44.1 to 24/96.
* Then using the dynamic range meter I analyzed the three original files, which all showed identical numbers:











Then I used Voxengo's plugin CurveEQ in Audacity to compare two files, which had to be converted to the same sample rate, as they would otherwise show quite big differences (that's apparently how CurveEQ works), and then it became "interesting". CurveEQ shows the difference in EQ between two files (volume level doesn't matter). The orange line represents the difference between the two files, so if the orange line is completely flat, the EQ is identical (although volume levels can be different).

24/96 downsampled to 16/44.1 vs. 24/192 downsampled to 16/44.1:






24/96 vs. vs. 24/192 downsampled to 24/96 looks exactly the same.

HOWEVER, when compared the original 16/44.1 to the other two files downsampled to 16/44.1 it looked like this:






It's the original 16/44.1 file that has this incread high frequency energy.
So, I'm wondering if the 16/44.1 file has been created with a bad sample-rate converter, which has a lot of ringing around 20 kHz. Or could it be something else?
Also, it's not for me to say if this was done intentionally by the label, or they simply used bad conversion without knowing it and then concluded (if this is audible) that 16/44.1 is crap.


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## board (Dec 4, 2019)

gregorio said:


> 1. I think I have understood him, as my interpretation is similar to yours (see point #2) but even if I haven't, it doesn't make a substantial difference because ...
> 1a. No, I don't think berating him will get clearer responses, we're past the point of clearer responses. Even if his responses become a little clearer as far as his use of language is concerned, they'll become less clear as far as a factual basis for his arguments and rational thought/critical thinking are concerned. And, neither is my primary goal to frustrate him so he'll just go away. My aim was to highlight the self-contradiction, hypocrisy and sheer ridiculousness of his responses. This might get him to evaluate his future posts more carefully (my primary goal) but probably not. He'll probably just ignore it completely, brush it off as me twisting his words or invent some other false excuse and just carry on regardless. There's also the possibility that others reading this thread might think that underneath the poor use of language, Alrainbow could have a rational/valid point. Highlighting Alrainbow's self-contradiction, irrationality (etc.) might make it easier for them to realise that he doesn't (my secondary goal). And lastly, as he descends ever deeper into his logical black-hole, he'll sooner or later reach a point where even he realises how ridiculous he's being or at least, that his arguments are futile. At which point he'll either leave the thread due to embarrassment or frustration or, there's a very small chance he might re-evaluate. Whichever it is though, his (deliberate or inadvertent) trolling/thread cr@pping will cease and the (third) "goal" of my post was to reach that point sooner rather than later!
> 
> 2. That's my interpretation too but that just brings us back to my (and others') very first response to him, that he's falsely conflating different masters with different container formats. So, we're now going round in a circle of his making!
> ...


I see your point, and I suppose I agree, but if you and I (and others) don't want to convince group 1, why are we trying to argue with them?
Why are we not just saying "please back up your statements with an ABX test and until you do we won't respond to you or engage in any kind of argument about this" and then simply stop responding to these people?
Certain scientists refuse to debate with creationists because they believe that engaging in a discussion gives some credibility to the creationists, meaning that by participating the scientists show that they take the creationists seriously.
Bill Nye did debate Ken Ham on the issue, and although I've only watched very short clips, then I think it ended up being a fun debate for the spectators, so ...


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## Daiyama

Thank you all and especially board for his extensive research.
I will take your advice and will look through my music library and see which 24/96 tracks I have (surely less than 1%) and will downsample this by myself.
I have foobar2000 (no special converting plugin), is this sufficient? (and audacity to check).
And also I do not want to be the hundredth person to whom you explain all that.
Is there a link to a page where most of this stuff is easily explained?


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## castleofargh

miksu8 said:


> If you quote someone please don't change the original message. I just asked how often 24bit music sounds better because it's produced better. It's reasonable question, since if someone has a better quality version of a track, it's probably sold as 24 bit version for marketing reasons.
> 
> Your answer didn't make any sense to me, sorry.


Intuitively, we would expect that. But in practice it's a lot messier. The best argument in favor of getting the hires file is often the worst reason of all. That a sound engineer was asked to make a different version for the different formats. While that was completely justified for vinyls vs digital because of the physical limitations of vinyl, it makes no sense for various digital resolutions. And yet it's not uncommon. In fact I'd argue that pretending to have audibly improved resolution was such a fail that they have no other choice but to manufacture the audible difference.
I've tried to get some understanding of this and watched/read as many interviews of sound engineers on this topic as I could. A few argue that the hires version will be purchased by an elite with better more neutral gear and a taste for more dynamic more blablablah elite audiophile flattery. While the rest of the pleb will listen to the album in their car or some cheap stereo system without extended frequency response. So the argument was to make CD versions that will still work fine with those poor uneducated people's circumstances. And this is ironically about the most reasonable explanation I've seen beside the guys who simply admit that they were asked to make different versions so they did their job. Most other explanations felt to me like the talk of crazy people or at least the talk very unscrupulous salespeople. My favorite was a guy who argued that the 16/44 conversion sounded nothing like the glorious hires master he had made, so he needed to remaster the CD version to bring back some subjective life into it. I'd love to see such a person try a blind test, just for the lolz.

And that's just the tip of the iceberg where we assume that all resolution deliveries of an album will have been made at the same time by the same guy. Just think of the MQA ubermasters of death (or whatever they're called when they can be unpacked to ??/192*). We rarely know what master they took or why, we have no clue what they did with it. So does it sound different? Is it a different master? Maybe. If so, is it better or worst? I'm going to guess that the correct answer is a solid "it depends on the track". And that's been the feedback of most people over the years about the many attempts to sell us the same things in a different box for a boosted price. Some masters are certainly enjoyed by many in those hires formats. But just as well, you can find people who prefer their old CD version for some albums. Is it habit? Is it just a vague subjective matter of personal taste? Did someone murder that latest master that just happens to be released in hires? Probably a little bit of everything happens to all formats. And I suspect those who claim that a resolution or a format always sounds better, to be people who live high on placebo.

*MQA is a good or very bad example as their super revolutionary products might have lower effective bit depth than even a CD. Maybe we should ask them to explain audibility and bit depth? ^_^


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## board

Daiyama said:


> Thank you all and especially board for his extensive research.
> I will take your advice and will look through my music library and see which 24/96 tracks I have (surely less than 1%) and will downsample this by myself.
> I have foobar2000 (no special converting plugin), is this sufficient? (and audacity to check).
> And also I do not want to be the hundredth person to whom you explain all that.
> Is there a link to a page where most of this stuff is easily explained?


You can at least install the separate SoX resampler plugin in Foobar to convert sample rate.
Otherwise, if you only want to change bit-depth, you can also use Foobar:
Right-click on the track, choose "convert", choose "wave" and then in the drop-down menu there's the option of bit-depth.
It should also be said that if you want to change both sample-rate and bit-depth you would have to go through both processes describred, if I'm not mistaken.

Maybe some of you have already seen this, but Mark Waldrep from AIX records and Real HD Audio is doing round II of his "hi-res challenge", where he grants access to select songs of his AIX catalogue of 24/96 recordings, which he has then converted to 16/44.1. I trust him and his equipment, but the way he is doing his challenge, it is possible to cheat. But you can, however, get true hi-res files from him. As he has repeatedly pointed out, many files being sold as hi-res are actually not real hi-res, as they were recorded on analogue tape in days of yore, so they don't contain much ultrasonic content to begin with. All his AIX recordings are true 24/96 recordings.


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## board

taffy2207 said:


> Let's not forget the monetary aspect as well. A lot of Audiophiles have invested heavily in Hi-Res Music. It's a bitter bill to swallow for them to admit that they could be wrong about it and thus potentially wasted their money.
> 
> It doesn't just apply to Hi-Res Music, unfortunately. It applies to gear as well, particularly people who bang on about 'endgame' set ups.


I completely agree. As Ethan Winer once said, people don't like to be told that their hyper expensive gear has been a waste of money. In extension of my post above about the two groups of people, the people who spend extraordinary amounts of money on their stereo system tend to build their entire lifestyle around this, so telling tham their equipment has been a waste of money is in their eyes equivalent to saying that they're wrong and that there's something wrong with them and that their entire adult life has been a joke. Just look at how almost all audiophiles react when they're being told these things about their beliefs. It's usually "the test is wrong", "some stupid test equipment can't measure better than what I heard", etc., and usually their falacious beliefs end becoming reinforced.


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## bigshot

miksu8 said:


> If you quote someone please don't change the original message. I just asked how often 24bit music sounds better because it's produced better.



I'm sorry. I misread your question too. I didn't see the "how often". I have lots of SACDs and lots of CDs. I haven't found that SACDs sound better more often than CDs do. It's pretty much a crap shoot. The best way to find out what the best sounding release is, is to speak with knowledgeable collectors who have heard all the available masterings and can point you to the best one. That is just as likely to be a CD as it is an SACD or blu-ray audio disc. In many cases, they all use the same master, so it doesn't matter which one you buy.


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## board

bigshot said:


> I'm sorry. I misread your question too. I didn't see the "how often". I have lots of SACDs and lots of CDs. I haven't found that SACDs sound better more often than CDs do. It's pretty much a crap shoot. The best way to find out what the best sounding release is, is to speak with knowledgeable collectors who have heard all the available masterings and can point you to the best one. That is just as likely to be a CD as it is an SACD or blu-ray audio disc. In many cases, they all use the same master, so it doesn't matter which one you buy.


Whenever possible I compare the different versions first and then I buy the version I prefer. I've sometimes seen people rave about certain releases, and when I listen for myself I find that I prefer the one that those people dispise, although there is often consensus between me and others.

As for "24 bit versions sound better because they're better produced", if it wasn't clear by everything that has been written by various people here, then it's not the bit depth that makes it sound better, since it will sound the same when converted to 16 bit - it's simply that it's better produced to begin with, no matter the bit depth.
I don't have so much experience with SACDs, but I agree that it's a craps shoot.


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## castleofargh

board said:


> Whenever possible I compare the different versions first and then I buy the version I prefer. I've sometimes seen people rave about certain releases, and when I listen for myself I find that I prefer the one that those people dispise, although there is often consensus between me and others.
> 
> As for "24 bit versions sound better because they're better produced", if it wasn't clear by everything that has been written by various people here, then it's not the bit depth that makes it sound better, since it will sound the same when converted to 16 bit - it's simply that it's better produced to begin with, no matter the bit depth.
> I don't have so much experience with SACDs, but I agree that it's a craps shoot.


I work in a very simple way. If I've heard one master a lot(usually discovered the album with it), anything else will feel wrong to me. No matter how clean or nice another version is, for me habit and nostalgia triumph over almost anything else. To the point that I've been looking to acquire all the old classical music from my youth(quite the battle to try and find a symphony from vague memories of what the cover looked like, and google image ).So far I've very much enjoyed the ones I could find, and I honestly like them more than other better known and acclaimed versions. Even though I admit that several of those acclaimed version deserve every bit of praise.

 In the end I like my familiar stuff better. And while it's obvious with classical music where mastering is but a small piece of the puzzle, I also tend to feel that way for other genres. I discovered Iron Maiden with the "A real live one" CD, I now have pretty much all the albums(so long as Dickinson is the singer!!!!!!), and a bunch of DVDs of their concerts. so again I'm talking more than subtle remastering. But even after all these years, if a song was on "A real live one", that's the version I'll like the most. Pretty much all remasters of my favorite music tend to disappoint me.
So I have to say, it's been a massive money saving bias. I usually hunt biases like I'm Highlander, but I'm starting to feel grateful for that one. I just hope someone will shoot me the day I drift from that to saying: "it was better before".


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## bigshot

Nowadays, I buy music I'm not familiar with, so I'm not stuck with the duckling patterning.


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## old tech

board said:


> You can at least install the separate SoX resampler plugin in Foobar to convert sample rate.
> But you can, however, get true hi-res files from him. As he has repeatedly pointed out, many files being sold as hi-res are actually not real hi-res, as they were recorded on analogue tape in days of yore, so they don't contain much ultrasonic content to begin with. All his AIX recordings are true 24/96 recordings.


I understood that it is not the lack of ultrasonic content which means that analog recordings can never be hi res, but rather the resolution ie signal to noise and the associated dynamic range through the relevant (to young humans) 20 to 20khz frequency range - hence the 'res' in the high res...  From this perspective first generation analog tape on studio equipment with noise reduction is at best equivalent to 14 bits digital. The ultrasonic aspect is irrelevant.


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## board

old tech said:


> I understood that it is not the lack of ultrasonic content which means that analog recordings can never be hi res, but rather the resolution ie signal to noise and the associated dynamic range through the relevant (to young humans) 20 to 20khz frequency range - hence the 'res' in the high res...  From this perspective first generation analog tape on studio equipment with noise reduction is at best equivalent to 14 bits digital. The ultrasonic aspect is irrelevant.


That would be the 24-part of the 24/96 equation (the bit depth). The ultrasonic part would be the 96 part.


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## board

castleofargh said:


> I work in a very simple way. If I've heard one master a lot(usually discovered the album with it), anything else will feel wrong to me. No matter how clean or nice another version is, for me habit and nostalgia triumph over almost anything else. To the point that I've been looking to acquire all the old classical music from my youth(quite the battle to try and find a symphony from vague memories of what the cover looked like, and google image ).So far I've very much enjoyed the ones I could find, and I honestly like them more than other better known and acclaimed versions. Even though I admit that several of those acclaimed version deserve every bit of praise.
> 
> In the end I like my familiar stuff better. And while it's obvious with classical music where mastering is but a small piece of the puzzle, I also tend to feel that way for other genres. I discovered Iron Maiden with the "A real live one" CD, I now have pretty much all the albums(so long as Dickinson is the singer!!!!!!), and a bunch of DVDs of their concerts. so again I'm talking more than subtle remastering. But even after all these years, if a song was on "A real live one", that's the version I'll like the most. Pretty much all remasters of my favorite music tend to disappoint me.
> So I have to say, it's been a massive money saving bias. I usually hunt biases like I'm Highlander, but I'm starting to feel grateful for that one. I just hope someone will shoot me the day I drift from that to saying: "it was better before".


Fair enough . I'm a bit the opposite. I remaster music myself fairly often - I even remastered a song from my favourite album of all time, which is an album I've lived with since 1996 when it was released, and I was very happy with the result. In a similar vein, when "The White Album" by The Beatles was remixed I was instantly impressed (although I did prefer two songs from the 2009 remaster and original mix).


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## old tech

board said:


> That would be the 24-part of the 24/96 equation (the bit depth). The ultrasonic part would be the 96 part.


Yes but the 24 bit part relates to resolution, ie SNR.


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## board

old tech said:


> Yes but the 24 bit part relates to resolution, ie SNR.


Agreed, although instead of "resolution" I would simply call it signal to noise ration, as you also say, as people often understand "resolution" to mean something else (like Gregorio pointed out in his original post). But then still, hi-res is defined by being recordings of higher specs than standard CD specs, which, if I'm mistaken, would have to include both the bit depth and the sample rate.
Just to avoid any confusion: I don't see any need for playback at higher than 16/44.1. I've also recorded quite many vinyl LPs onto my computer, and I do all of that at 16/44.1 (or 16/48 due to computer issues). I've done ABX tests of CD spec vs. hi-res and heard no difference.


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## gregorio

board said:


> It's the original 16/44.1 file that has this incread high frequency energy.
> So, I'm wondering if the 16/44.1 file has been created with a bad sample-rate converter, which has a lot of ringing around 20 kHz. Or could it be something else?



It's very unlikely that the original used a bad sample rate converter. There were some dodgy sample rate converters around the turn of the millennium but that situation only lasted for a few years and you probably wouldn't expect to see that amount of ringing unless you used a test signal specifically designed to highlight it. Far more likely is that it's the result of noise-shaped dither, where quantisation error is effectively converted to white noise and "shaped", IE. The dither noise is reduced in the frequency band/area where our hearing is most sensitive and moved to areas where we're least sensitive, typically from around 16kHz - 22kHz.



Daiyama said:


> [1] I have foobar2000 (no special converting plugin), is this sufficient? (and audacity to check).
> [2] Is there a link to a page where most of this stuff is easily explained?



1. Foobar should be fine. Using a dedicated converting plugin with noise-shaped dither would be more representative of how commercial music releases are handled and is theoretically better. Although as castleofargh stated, standard (triangular) dither or no dither at all (truncation) is very unlikely to audible except in rare cases and at very high (uncomfortable) listening levels. If you're going to do it manually without a dedicated plugin, reduce the sample rate first and then reduce the bit depth, not the other way around.

2. No idea, I never really use foobar but quite a few here do and can point you in the direction of a simplified page, if there is one.



castleofargh said:


> Intuitively, we would expect that. But in practice it's a lot messier. The best argument in favor of getting the hires file is often the worst reason of all. That a sound engineer was asked to make a different version for the different formats. While that was completely justified for vinyls vs digital because of the physical limitations of vinyl, it makes no sense for various digital resolutions. And yet it's not uncommon. In fact I'd argue that pretending to have audibly improved resolution was such a fail that they have no other choice but to manufacture the audible difference.
> I've tried to get some understanding of this and watched/read as many interviews of sound engineers on this topic as I could. A few argue that the hires version will be purchased by an elite with better more neutral gear and a taste for more dynamic more blablablah elite audiophile flattery. While the rest of the pleb will listen to the album in their car or some cheap stereo system without extended frequency response. So the argument was to make CD versions that will still work fine with those poor uneducated people's circumstances. And this is ironically about the most reasonable explanation I've seen beside the guys who simply admit that they were asked to make different versions so they did their job.



In practice, it's even messier than you've explained. The membership here on head-fi are generally very serious about sound quality, which is largely why they're members in the first place. Engineers are very serious about SQ too, though typically not in the same way that the membership here is. To us engineers, SQ is dependant on the consumers' receiving media/equipment but most here on head-fi have a rather simplistic, narrow/exclusive notion of SQ.

There is, potentially, a justification for quite a number of different versions/masters. For example, a version for analogue radio playback, which accounts for the time restrictions and the severe multi-band limiters typically employed on music broadcast stations. So going back to SQ, the goal with a radio edit/master is to achieve the best SQ for radio listeners. However, if you were to play this audio file from disk (rather than receiving it as a radio transmission) on a good sound system (rather than a typical radio system), most/all the membership here would consider this version to be significantly "worse", while us engineers consider it to be "better" (for the intended purpose). In fact, when radio play was the single greatest driver of record sales, those engineers with the specialist knowledge, experience and skill to create a master that wasn't degraded by the radio broadcast chain (and compared favourably to other broadcast tracks), were highly sought after (and rewarded). From the narrow audiophile point of view, these skilled engineers were rewarded for effectively making "worse" masters! Likewise, a Youtube version/master is quite commonly desired, with the requirement that the track/s at least work to some degree when played on the internal speakers of laptops and other mobile devices, which is difficult to achieve if you want that master to still compare decently when played back on more reasonable equipment. There may also be another version/master required for TV broadcast and another for film, etc.

Another consideration, unfortunately, is that the worth of most major record labels is largely dependant on their back catalogue. All the corporations that own the labels have back catalogues valued in the billions of dollars but of course, the actual worth of a back catalogue is dependant on the revenue it can generate and one of the ways that can be achieved is by issuing new versions, re-mixes and re-masters. This can be difficult from the engineers' point of view, especially if a particularly good (or particularly loved) mix/master has already been released previously.

So, we've got a messy set of scenarios, circumstances and considerations, only one of which is primarily aimed at audiophiles. The set of digital "resolutions" is therefore (potentially) just an extension of an already messy set that started in the 1950's, which record labels will exploit if they feel it's worth the cost/effort, baring in mind that it is specifically aimed at the small (and misinformed) audiophile market.

G


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## Sterling2

gregorio said:


> It's very unlikely that the original used a bad sample rate converter. There were some dodgy sample rate converters around the turn of the millennium but that situation only lasted for a few years and you probably wouldn't expect to see that amount of ringing unless you used a test signal specifically designed to highlight it. Far more likely is that it's the result of noise-shaped dither, where quantisation error is effectively converted to white noise and "shaped", IE. The dither noise is reduced in the frequency band/area where our hearing is most sensitive and moved to areas where we're least sensitive, typically from around 16kHz - 22kHz.
> 
> 
> 
> ...


Sidebar: I download hi-res multi-channel music from Acoustic Sounds in FLAC, which I send to  Foobar 2000 Library from Music Folder. To play these multi-channel files I use an HDMI connection from Laptop to my OPPO's multi-channel DAC. The result is gapless but not glitch free. So far, I do not know if the dropouts are related to HDMI connection, Foobar 2000, OPPO-205, OPPO Driver,  synchronization of devices, Windows 10, or other. The only thing I know for sure is I can only get glitch free gapless via usb drive ports which allow me to play directly from the Music Folder.


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## TheSonicTruth (Dec 5, 2019)

gregorio said:


> Another consideration, unfortunately, is that the worth of most major record labels is
> largely dependent on their back catalogue. All the corporations that own the labels
> have back catalogues valued in the billions of dollars but of course, the actual worth
> of a back catalogue is dependant on the revenue it can generate and one of the
> ...



And the above, in particular, is precisely where you and I have sparred, lol!

I'm a firm believer in the 'leave the past alone!' mentality, but I sort of understand how changing the sound of an older album and reissuing it as "Digitally Remastered"(or words to such effect) on the packaging can work marketing wonders with the (uniformed) masses at large.

My question then, is:

Does my preference for, IE: my 1986 original CD issue of a particular artist/album over a recent 'remaster' of it make me (A.) an audiophile, (B.) a melophile(just loves the music), or (C.) something else? (be gentle here, Gregorio... remember, bedside -or, board-side - manner is your 2020 resolution!  )


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## gregorio

TheSonicTruth said:


> [1] I'm a firm believer in the 'leave the past alone!' mentality, but I sort of understand how changing the sound of an older album and reissuing it as "Digitally Remastered" on the packaging can work marketing wonders with the (uniformed) masses at large.
> [2] My question then, is: Does my preference for, IE: my 1986 original CD issue of a particular artist/album over a recent 'remaster' of it make me (A.) an audiophile, (B.) a melophile(just loves the music), or (C.) something else?



1. But you can't have that mentality .... or rather, you can't only have that mentality, because it comes with consequences. The valuation of the major record labels is largely based on the valuation of their back catalogue, if they can't/don't monetize those back catalogues to the max (with reissues, remasters, remixes, licensing the copyrights, etc.) the value of those back catalogues are significantly lower or effectively zero if they aren't monetized at all and therefore the value of the record label is significantly lower. It's share price will reduce proportionately and most likely the label wouldn't survive, baring in mind the pressure they're already under from the falling revenue of their new artists/releases (due to streaming media). Your "mentality" is therefore "leave the past alone" and not have major record labels or their investment in new artists and new recordings. That mentality is potentially extremely damaging to pretty much the entirety of the commercial recording industry but most particularly to the high quality end. Isn't that the opposite of what audiophiles/melophiles would want?

2. It could be that a particular/original release has some personal significance at that point in time and a different version doesn't evoke the memories or sound quite right, so it could just make you (C) An ordinary member of the public. The correct answer IMHO is (D) Any of the above.

G


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## castleofargh

old tech said:


> I understood that it is not the lack of ultrasonic content which means that analog recordings can never be hi res, but rather the resolution ie signal to noise and the associated dynamic range through the relevant (to young humans) 20 to 20khz frequency range - hence the 'res' in the high res...  From this perspective first generation analog tape on studio equipment with noise reduction is at best equivalent to 14 bits digital. The ultrasonic aspect is irrelevant.


to me random audiophile, the hires standard is what was pushed by Sony with their golden logo thingy. and the definition is fairly simple, it refers to anything above 16/48. but I don't know what level of traceability they have for the files themselves. it's clear that the rules are nonsensical when it comes to gears. if a speaker can burps some vibration above -30dB at 40kHz, it's a hires speaker. doesn't matter if the thing has a stupid FR and 10%THD, it's hires. so I'm expecting similar jokes for audio files and I doubt that they have much consideration for noise floor if any at all. so at least under that particular standard, old noisy and distorted analog stuff could still get a nice golden logo.


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## TheSonicTruth

gregorio said:


> 1. But you can't have that mentality .... or rather, you can't only have that mentality, because it comes with consequences. The valuation of the major record labels is largely based on the valuation of their back catalogue, if they can't/don't monetize those back catalogues to the max (with reissues, remasters, remixes, licensing the copyrights, etc.) the value of those back catalogues are significantly lower or effectively zero if they aren't monetized at all and therefore the value of the record label is significantly lower. It's share price will reduce proportionately and most likely the label wouldn't survive, baring in mind the pressure they're already under from the falling revenue of their new artists/releases (due to streaming media). Your "mentality" is therefore "leave the past alone" and not have major record labels or their investment in new artists and new recordings. That mentality is potentially extremely damaging to pretty much the entirety of the commercial recording industry but most particularly to the high quality end. Isn't that the opposite of what audiophiles/melophiles would want?
> 
> 2. It could be that a particular/original release has some personal significance at that point in time and a different version doesn't evoke the memories or sound quite right, so it could just make you (C) An ordinary member of the public. The correct answer IMHO is (D) Any of the above.
> 
> G



1. Higher quality - But is  a remaster of higher quality, even if created from a better source than available 20-30 years ago, if the end product is just processed to be louder and potentially less dynamic than the original?

2. Sound quite right - For me, it's a matter of the fact that the remaster sounds different than what I'm used to, or different than the original release something.  That difference, to me, removes such remaster from canon.


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## gregorio

TheSonicTruth said:


> 1. Higher quality - But is a remaster of higher quality, even if created from a better source than available 20-30 years ago, if the end product is just louder and less dynamic than the original?



1. My point #1 previously does not address the quality of remasters, it addresses the quality of future (new) recordings if remasters/remixes/etc are not used to monetize the back catalogues!

G


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## old tech

castleofargh said:


> to me random audiophile, the hires standard is what was pushed by Sony with their golden logo thingy. and the definition is fairly simple, it refers to anything above 16/48. but I don't know what level of traceability they have for the files themselves. it's clear that the rules are nonsensical when it comes to gears. if a speaker can burps some vibration above -30dB at 40kHz, it's a hires speaker. doesn't matter if the thing has a stupid FR and 10%THD, it's hires. so I'm expecting similar jokes for audio files and I doubt that they have much consideration for noise floor if any at all. so at least under that particular standard, old noisy and distorted analog stuff could still get a nice golden logo.


That logo, and the JAS standard developed for it, relates to hardware only (although it doesn't stop the labels and distributors from using it!).  Mark Waldrep discusses this issue in the link below.  Interestingly, a separate standard developed by the Digital Entertainment Group an others for a hi res definition for musical content.  They settled on anything that comes from a source better than CD quality - a subjective term rather than "better than CD specifications" - an objective term, because that would have excluded all analog recordings.  Waldrep asked the committee that developed this definition whether that means that a 1920 recording transferred to a 24/96 file makes that recording hi res, and they said yes!  

The whole hi res thing as a consumer format is snake oil designed to keep dollars flowing into the industry.

https://www.realhd-audio.com/?p=6405


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## board

old tech said:


> That logo, and the JAS standard developed for it, relates to hardware only (although it doesn't stop the labels and distributors from using it!).  Mark Waldrep discusses this issue in the link below.  Interestingly, a separate standard developed by the Digital Entertainment Group an others for a hi res definition for musical content.  They settled on anything that comes from a source better than CD quality - a subjective term rather than "better than CD specifications" - an objective term, because that would have excluded all analog recordings.  Waldrep asked the committee that developed this definition whether that means that a 1920 recording transferred to a 24/96 file makes that recording hi res, and they said yes!
> 
> The whole hi res thing as a consumer format is snake oil designed to keep dollars flowing into the industry.
> 
> https://www.realhd-audio.com/?p=6405


I find Mark Waldrep to be the only person in the hi-res business worth listening to. He also admits that he can't distinguish hi-res from CD specs, and he also said, after his failed hi-res challenge last year, that he doesn't believe anyone else can either. No one else in the hi-res industry has ever said that as far as I know!

On another note: Have you noticed how Alrainbow went away? If we're "lucky" so to speak maybe it's not only temporarily .


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## board

gregorio said:


> It's very unlikely that the original used a bad sample rate converter. There were some dodgy sample rate converters around the turn of the millennium but that situation only lasted for a few years and you probably wouldn't expect to see that amount of ringing unless you used a test signal specifically designed to highlight it. Far more likely is that it's the result of noise-shaped dither, where quantisation error is effectively converted to white noise and "shaped", IE. The dither noise is reduced in the frequency band/area where our hearing is most sensitive and moved to areas where we're least sensitive, typically from around 16kHz - 22kHz.
> 
> G


Is it honestly very likely that dithering could produce such a spike?


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## bigshot

I can think of an example where a remix/remaster completely changed an album for the better... Fleetwood Mac's Tusk. I remember when the album came out. Fleetwood Mac were riding high off of two huge albums in a row. Tusk landed with a resounding thud because it sounded horrible... an undifferentiated mass of noise and distortion. The CD wasn't a whole lot better than the original LP. Recently, it has been remixed and remastered and there is absolutely no reason to ever go back to the original version. The mix makes total sense now. Buckingham was experimenting with different kinds of sounds, but whoever was in charge of the mix was totally unsympathetic to it, and wallpapered over the contrasts with a muddy mix. It sounds like it might have even been deliberately sabotaged. Another example like this is Miles Davis's B itches Brew. They were doing cut and paste experiments that the technology didn't quite support yet There were so many layers of sound, it started to go opaque. The producer recently came out of retirement to remix it for multichannel and now all of the overlapping sound is clear and balanced, with no mist of noise obscuring the details.

The quality of remixing and remastering is much better now than it was a decade ago. You just have to look at the right titles.


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## TheSonicTruth (Dec 5, 2019)

bigshot said:


> I can think of an example where a remix/remaster completely changed
> an album for the better... Fleetwood Mac's Tusk. I remember when the album came out.
> Fleetwood Mac were riding high off of two huge albums in a row. Tusk landed with a resounding
> thud because it sounded horrible... an undifferentiated mass of noise and distortion. The CD
> ...




Point well taken.  But for every "Tusk" remaster scenario there are ten 'compress-crank-up & repeat' remasters out there of albums from the popular(Pop, Hip-Hip, Country, & RnB) genres. That is why I have largely sworn off anything with "remastered" on it, in my collection.

"_The quality of remixing and remastering is much better now than it was a decade ago. You just 
have to look at the right titles_"

You sound like such a salesman with that.  No thanks!


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## bigshot (Dec 5, 2019)

That is mostly true of ephemeral pop music. It isn't true at all for classical, jazz, classic country or classic pre-1980s reissues. All of those genres have experienced huge improvements in the past ten years. Sony in particular has worked wonders on back catalog RCA, Columbia, and Epic titles. Likewise, many 70s rock bands are going through their albums and remixing for multichannel. Just about anything remixed by Steven Wilson sounds better than any previous release. Digital restoration techniques have made a big impact particularly on pre-hifi stuff, and they are working from the original metal parts, not dubs. 78rpm recordings generally are mastered much better than they ever were in the LP or early CD era. There's some wonderful work being done, but it's on classic legacy recordings intended for adults with good systems, not throwaway pop stuff for kids with portable earphones.

I'd say less than 20% of what I buy is not an improvement over previous releases of the same material. I just got a expanded version of Prince's 1999 that I'm looking forward to. His stuff is crying out for first class reissues.


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## gregorio

board said:


> [1]I find Mark Waldrep to be the only person in the hi-res business worth listening to. He also admits that he can't distinguish hi-res from CD specs, and he also said, after his failed hi-res challenge last year, that he doesn't believe anyone else can either.
> [2] No one else in the hi-res industry has ever said that as far as I know!



1. Yes, I always considered him to not only be very well informed but he also backed it up by producing high quality recordings. My only criticism of him was his adherence to the hi-res bandwagon. I didn't know he'd recently actually done a hi-res vs CD test and admitted he couldn't distinguish them. My respect for him has increased for publicly admitting this and of course, my only criticism of him is no longer valid!

2. I've been part of the hi-res industry and I've been saying that for over a decade, as this thread evidences. Many others have too but I agree, as an owner of a hi-res label, rather than only as an engineer, I don't recall ever hearing a hi-res label ever admitting this.


board said:


> Is it honestly very likely that dithering could produce such a spike?


Yes. I'm not sure exactly what your CurveEQ is displaying but it appears there's little material above about 16kHz. So the difference (spike) indicated, an extra 13dB or so between 17kHz - 22kHz, would be about right for many noise-shaped dither algorithms. 

Noise-shaped dither usually employs about 1.5 bits of dither, which is about 9dB of white noise throughout the spectrum (up to the Nyquist limit) but if we shape that noise, move some of it from the critical hearing band and concentrate it in the least sensitive hearing band (17kHz and higher), then you'll end-up with roughly +13dB or so of noise in that high freq band. Remember that noise shaping doesn't reduce digital noise, that's not mathematically possible, you get exactly the same total amount of noise with 1.5 bits of noise-shaped dither as you would with 1.5 bits of triangular (standard) dither but it's concentrated in a smaller (high) freq band and therefore must have proportionately higher amplitude in that band. While +13dB or so of noise above about 17kHz might appear a bit drastic, it's not, it's inaudible. 

Human hearing sensitivity has a sharp fall-off starting around 12kHz - 14kHz, You can see this starting point even in the 1933 Fletcher-Munson loudness contours, which indicates that at around 14kHz -15kHz hearing sensitivity is about 20dB below peak sensitivity. By 17kHz sensitivity is, at the very least, -30dB below peak sensitivity but that's a best case scenario (teenagers with very good hearing), the average twenty something would be at least -40dB and probably lower. So although +13dB of noise might appear to be quite a lot, compared to the hearing sensitivity roll-off, it's insignificant (and inaudible), which of course is the whole point of noise-shaped dither in the first place.

G


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## Isloo

@gregorio - Great thread! I only ever dabbled in HiRes a few years ago. However, I couldn't tell the difference between HiRes and CD quality, so gave up on it. To be honest, I can't tell the difference between 320 Mp3 or 256 AAC and red book, so I save the space. It's really interesting to learn some of the theory behind all of this and to be a little less uninformed. A big Thank you to you and all of the other contributors to the thread who have been kind enough to share their knowledge.


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## TheSonicTruth (Dec 6, 2019)

bigshot said:


> That is mostly true of ephemeral pop music. It isn't true at all for classical, jazz, classic country or classic pre-1980s reissues. All of those genres have experienced huge improvements in the past ten years. Sony in particular has worked wonders on back catalog RCA, Columbia, and Epic titles. Likewise, many 70s rock bands are going through their albums and remixing for multichannel. Just about anything remixed by Steven Wilson sounds better than any previous release. Digital restoration techniques have made a big impact particularly on pre-hifi stuff, and they are working from the original metal parts, not dubs. 78rpm recordings generally are mastered much better than they ever were in the LP or early CD era. There's some wonderful work being done, but it's on classic legacy recordings intended for adults with good systems, not throwaway pop stuff for kids with portable earphones.
> 
> I'd say less than 20% of what I buy is not an improvement over previous releases of the same material. I just got a expanded version of Prince's 1999 that I'm looking forward to. His stuff is crying out for first class reissues.



Again, great pitch, bigshot!  The late Chick Lambert and you have a lot in common, and he'd be proud of the above 

I may have alluded to this some while ago, but my issue with remasters is not always one of improvement(that is for surr a subjective measure), but that with many remasters a _change_ has been made to the sound.

I'd prefer the slightly rolled-off-high-end original version of something, as opposed to something that has had V-shaped EQ and who knows what else applied to it, in a fancy new jewel case.

Same with remixes: I'll take certain original stereo Beatle albums, where John or Paul is way off in one channel, opposite the meloldy on the other side, with slight background hiss, VS a remixed reissue where those vocals are perfectly centered down to one-tenth of one dB, and elements of the melody are panned modestly to both sides, and dead sterile silence in the quiet passages. That slightly hissy original stereo version is canon, compared to the cleaned, and sometimes loudened up, remix, which makes a sixties legend sound sickeningly modern - a la One Direction, etc.

Succinctly, bigshot, and Gregorio: in most cases, I'm not always after a collection that sounds 'better'(again, subjective opinion!), but rather, what WAS - and contemporary to the era it came out.


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## bigshot (Dec 6, 2019)

A lot of recordings were far from perfect on original release, particularly mid to late 70s LPs during the oil crisis. Mixing and mastering was done to suit the limitations of the LP format, and when the same masters are used for CDs, the warts can be revealed. Whether or not a release is the "best" or not is both a subjective determination and one that varies from case to case. I don't think it's possible to say that all remasters sound bad or all remasters sound better. The only thing you can say is remasters in general sound different to varying degrees and you have to hear it and decide for yourself.

The things that can be generalized about is that Sony has done great work improving the sound of the RCA and Columbia library. RCA's dynaflex LPs (prime era David Bowie) deliberately introduced distortion in the mastering to make it sound better on crappy stereos. That is fine if you have a crappy stereo, but if you have a good one, the remastered CDs sound a LOT better. That is also what the problem is with modern ephemeral pop music. It's *designed* from the ground up to sound good on crappy stereos. I played a recent funk CD on my system and the bass nearly blew me out the door. It had been mastered to compensate for lousy bass response in cheap ear buds.

The trick is to look for the masterings that are designed to be played on good systems. That is easy to do if you are interested in classical and jazz, because just about everything in those genres is designed to be played on first class stereo systems. If you want to listen to current pop music, you are probably out of luck because the audience for that doesn't have fancy stereos... they listen on their phone using cheap buds. In that case, your best bet is to just focus on the music if you like it, and try not to think about the sound quality. It's not apt to ever be released sounding perfect. And if it was, it sure wouldn't sound the same.

Which Beatles remix are you referring to? Rubber Soul? The stereo version of that was remixed back on first CD release because Martin wasn't happy with the stereo mix. Other than that (and the recent multichannel mixes of Sgt Pepper, White Album and Abbey Road- and Let It Be Naked) all of the Beatles CDs have always sounded basically the same as the masters. The difference between the first release and the recent CD remaster is negligible. I have both and I've compared.

If you want truly authentic sound, not better sound, you should get a Marantz receiver, a Technics turntable and buy up used vinyl. You can't get more medium-fi goodness than that.


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## TheSonicTruth

bigshot said:


> If you want truly authentic sound, not better sound, you should get
> a Marantz receiver, a Technics turntable and buy up used vinyl. You
> can't get more medium-fi goodness than that.



No need for sarcasm.  My original CD issues are fine for my purposes, as horrid as they must sound to you, lol!


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## TheSonicTruth

bigshot said:


> and when the same masters are used for CDs, the warts can be revealed. W



Such as?  I have the first three Van Halen(late 70s Roth era) CDs, quite likely transfers of LP master tapes, and they sound great to me on whatever I listen to them on - my full-sized home stereo, my car system, ripped as MP3 on my phone, whatever.

What "warts" should I be looking out for?  LF roll-off below 50Hz?


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## bigshot (Dec 6, 2019)

That isn't sarcasm. If you are listening to 70s music and you want it to sound exactly as you remember it- not changed nor improved- a turntable is the way to get that. I have phonographs to play my 78s. It's a specific sound signature that recordings in that era were designed to work with. When these recordings get released on CD, they remaster them for CDs and the way the expected audience listens to them. Same goes for streaming. Music is mastered for the medium.


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## TheSonicTruth (Dec 6, 2019)

bigshot said:


> That isn't sarcasm. If you are listening to 70s music
> and you want it to sound exactly as you remember
> it- not changed nor improved- a turntable is the way
> to get that. I have phonographs to play my 78s. It's
> ...



When I listen to albums of which I own both the LP and first issue CD, I hear very little difference between them, aside from the absence, on the CD, of background noise and crackle associated with the vinyl. The CD volume *is* a tad higher, but that probably has more to do with the CD deck having a slightly higher output than the turntable.

Timbre/EQ wise, their signatures are similar, save perhaps for a tad more bottom on the CD.  Perhaps *some* processing was done to the master destined for CD, I don't know. But it doesn't seem like enough to make a huge difference, compared to the sound of the LP.

Now, compare that 40 year old but gently used LP to the recently remastered CD reissue, the biggest difference I hear is how much LOUDER the remaster  CD is than either the original CD or the LP.

Do I consider that remastered CD to be an improved version?  No! Just a louder version.  Not an important enough criterion for me to want to keep it, except to make these points to fellow music fans and friends.


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## bigshot

I've got over 10,000 records and 10,000 CDs, and I can't make any generalization like that. I think you make stats like that up to support your agenda.

If your LPs don't sound as loud, you can simply turn up the volume.


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## TheSonicTruth

bigshot said:


> I've got over 10,000 records and 10,000 CDs, and I can't make any generalization like that. I think you make stats like that up to support your agenda.
> 
> If your LPs don't sound as loud, you can simply turn up the volume.



No agenda.  The remastered CDs of many popular artists(The Who, Billy Joel, etc) ARE louder than the 30 year old original CDs and the LPs.

According to what?  My EARS. Sorry if my ears aren't a "scientific" enough tool for judging differences in how loud different sources are.


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## Chris Kaoss

bigshot said:


> I've got over 10,000 records and 10,000 CDs, and I can't make any generalization like that. I think you make stats like that up to support your agenda.
> 
> If your LPs don't sound as loud, you can simply turn up the volume.



It isn't a matter of volume level at all.
I did comparison between the first queen releases and the 2011 remastered ones.
To me, the former sounds better couz they are more dynamic and natural then the latter.
It doesn't belong to all remastered old pieces, but with the first releases of cd's of queen or that decade, you would hear it the best.


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## TheSonicTruth

Chris Kaoss said:


> It isn't a matter of volume level at all.



Ok, try to wrap your head around the following(because nobody else here seems to be able to!):

I have two CD issues, of the same 1970s era album that I like: 'A' = a 1985 original CD release, and 'B' = a remastered reissue from within the last ten years.

I put 'A' in my system and play it, and adjust the volume for comfortably loud.

Next, I remove 'A' and put 'B' in and play it, without touching my volume.

The music is now UNCOMFORTABLY LOUD, and I must lower my volume setting to approximate the listening level I set previously, while playing 'A'.

Why must I set my volume lower for CD B than I set it for A?  Because version B of that album IS LOUDER than version A!!

It's not my 'perception', it's FACT!  Just as I am watching the sun rise here in CT as I am typing this!


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## Chris Kaoss

TheSonicTruth said:


> Ok, try to wrap your head around the following(because nobody else here seems to be able to!):
> 
> I have two CD issues, of the same 1970s era album that I like: 'A' = a 1985 original CD release, and 'B' = a remastered reissue from within the last ten years.
> 
> ...



This is exactly what i wanted to express. 
Plus, the music isn't as dynamic, quieter and louder parts, as in the former original records.


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## castleofargh

Chris Kaoss said:


> This is exactly what i wanted to express.
> Plus, the music isn't as dynamic, quieter and louder parts, as in the former original records.


As you might be able to guess from his profile pic even if you're not yet familiar enough with him, @TheSonicTruth is always grateful for the opportunity to turn any given topic into one about the Loudness War.


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## Chris Kaoss

I know. 
Did read the whole thread. Btw, thank you all for this great one.
I've learned so much with it.
I just wanted to pointing out my findings on this.


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## bigshot (Dec 7, 2019)

TheSonicTruth said:


> I have two CD issues, of the same 1970s era album that I like: 'A' = a 1985 original CD release, and 'B' = a remastered reissue from within the last ten years. I put 'A' in my system and play it, and adjust the volume for comfortably loud. Next, I remove 'A' and put 'B' in and play it, without touching my volume. The music is now UNCOMFORTABLY LOUD, and I must lower my volume setting to approximate the listening level I set previously, while playing 'A'.



How do you know the two discs were normalized to the same volume level? Maybe one is just mastered at a lower level than the other one.


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## TheSonicTruth

bigshot said:


> How do you know the two discs were normalized to the same volume level?



I never said or implied they were.  Stop twisting things!   (like you're so good at)

"Maybe one is just mastered at a lower level than the other one."

NOW you're on to something.


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## gregorio (Dec 8, 2019)

@TheSonicTruth, your recurring diatribe about mastering is off topic and therefore effectively thread cr@pping. However I'll address your points, for those who aren't already aware and because some falsely conflate bit depth/"resolution" with different masters.


TheSonicTruth said:


> But is a remaster of higher quality, even if created from a better source than available 20-30 years ago, if the end product is just processed to be louder and potentially less dynamic than the original?


Yes, it can be, depending on: It's intended purpose (range of targeted consumer playback scenarios), how dynamic the original master was and how well it's been processed to be louder and less dynamic.


TheSonicTruth said:


> [1] I may have alluded to this some while ago, but my issue with remasters is not always one of improvement(that is for surr a subjective measure), but that with many remasters a _change_ has been made to the sound.
> [2] I'd prefer the slightly rolled-off-high-end original version of something, as opposed to something that has had V-shaped EQ and who knows what else applied to it, in a fancy new jewel case.
> [2a] Same with remixes: I'll take certain original stereo Beatle albums, where John or Paul is way off in one channel, opposite the meloldy on the other side, with slight background hiss, VS a remixed reissue where those vocals are perfectly centered down to one-tenth of one dB, and elements of the melody are panned modestly to both sides, and dead sterile silence in the quiet passages. That slightly hissy original stereo version is canon, compared to the cleaned, and sometimes loudened up, remix, which makes a sixties legend sound sickeningly modern - a la One Direction, etc.



1. Again, that's the whole point of a remaster! There's obviously no financial sense in paying for a new master to be created that sounds identical to a previous master.
2. Firstly, it's unusual to apply V-shaped EQ to a master these days (or for many years), although it does happen sometimes depending on the "shape" of the previously released master. Also, even if the aim of a remaster is just to be louder/less dynamic, that can't be achieved without affecting the EQ to some degree, due to how compressors/limiters work. Secondly, a reoccurring theme is that you are talking about what YOU'd prefer or what YOU'RE after. Record labels do not make masters specifically/only for you personally! For example:
2a. You've answered your own point. Such a remix is obviously intended for consumers who listen to "One Direction, etc." (IE. Modern masters), not for those accustomed to (and in love with) 50+ year old masters. And, it should be obvious why! ...


TheSonicTruth said:


> [1] Succinctly, bigshot, and Gregorio: in most cases, I'm not always after a collection that sounds 'better'(again, subjective opinion!), but rather, what WAS - and contemporary to the era it came out.
> [2] Now, compare that 40 year old but gently used LP to the recently remastered CD reissue, the biggest difference I hear is how much LOUDER the remaster CD is than either the original CD or the LP.
> [2a] Do I consider that remastered CD to be an improved version? No!



1. Again, you're talking about what you're personally after. The vast majority of popular music consumers (more than 99% at a guess) are not after an accurate historical document but simply music they enjoy listening to, with their listening scenarios/environments. There are very few situations (with popular music) where it's both possible and profitable to make a remaster intended for a sub-group of consumers and playback scenarios which represents a tiny fraction of 1% of consumers.
2. How many portable LP or CD players were there 40 years ago? What then, was the likely consumer playback scenario (and therefore intended use) of the original CD/LP?
2a. That's your problem. Louder/Less dynamic clearly IS an improvement in scenarios with a high noise floor or with equipment incapable of a wide dynamic range, either or both of which are massively more common popular music consumption scenarios than a consumer in a low noise floor environment + equipment capable of a high dynamic range!


TheSonicTruth said:


> I have two CD issues, of the same 1970s era album that I like: 'A' = a 1985 original CD release, and 'B' = a remastered reissue from within the last ten years.
> I put 'A' in my system and play it, and adjust the volume for comfortably loud.
> Next, I remove 'A' and put 'B' in and play it, without touching my volume.
> The music is now UNCOMFORTABLY LOUD, and I must lower my volume setting to approximate the listening level I set previously, while playing 'A'.
> Why must I set my volume lower for CD B than I set it for A? Because version B of that album IS LOUDER than version A!!



And again, how often does the scenario you describe occur: Where a consumer is listening to a bunch of 1985 original CD releases and then plays a modern remaster?
Compare that to how often this scenario occurs: A consumer is listening to a bunch of tracks released in the last 25 years or so and then plays an original 1985 release? The music is now UNCOMFORTABLY QUIET and why must this consumer set their volume higher for the 1985 CD than for the rest of the music they generally listen to, especially at it's likely they won't, they'll simply "skip" it instead?

Your posts demonstrate exactly what my Post #5314 was largely written to avoid! You're judging SQ of masters/remasters by a "narrow/exclusive notion" that typically is NOT applicable (to popular music genres) and even when it is applicable (classical, jazz and a tiny number of popular genre releases) can be fully realised with 16bits anyway.

G


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## TheSonicTruth

gregorio said:


> must this consumer set their volume higher



Which is the way it should be:  I run my source inputs low, and keep the final volume high - not maxxed, mind you, but high enough so the amplifier is doing most of the work, and most available headroom: What I call a "pull" scenario.  Unless the final amp in question is an unusually hissy older one, then I experiment for  a good balance between input level and amp output level.

This is in contrast to what is going on today, what I call a "push" scenario:  Higher source/input levels and lower amp output settings to speakers.  I hooked a multimeter to the outputs of a CD player, and took note of the average output voltage of the original CD ('A' from above) and that of the recent remastered reissue('B'), and I suppose you'd guess which had the higher average output voltage.

I find that my pull scenario sounds better, more open, when running sound, vs the push scenario.


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## bigshot (Dec 8, 2019)

Whether a CD is normalized up to 99% or 85% has absolutely no impact on perceived sound quality. There's more than enough dynamic range to normalize sound down to be noticeably quieter. Just balance the volume levels and they might be basically the same. Quieter or louder isn't necessarily better. Sometimes it's just quieter or louder.

Last night I put on the new remaster of Prince's 1999 album. It sounds fantastic. Much better than any previous release. Pricey, but worth it. Included in the package is a live performance on DVD. It has decent cinematics since it's a multicamera shoot, but the video quality is well below broadcast quality- fuzzy, low contrast, dim. The sound appears to be recorded from a microphone at the mixing board in the middle of the auditorium. There's lots of hall ambience. Lyrics aren't always clear because of the reverberation. Not at all what you would call high fidelity sound. But that DVD is much better than the album because it has every one of the elements necessary for a good recording of music but the least important ones.

I can listen to a Caruso record that was recorded with a horn and played back with a steel needle and still get shivers down my spine. I can listen to the Stones original single of Street Fighting Man and feel the excitement, even though it is compressed like a pancake and slathered in distortion. I can listen to early recordings of Eddie Arnold in mono on a 78 and feel like I've gone back in time to 1948 and I'm sitting in the room with the band. None of those recordings come anywhere come close to CD quality, yet they are alive and vital sounding. Why?

Musicality- The musical creativity is of primary importance. Is the artist expressing himself through his music in a way that is totally original and new? (Think The Beatles)
Musicianship- Does the artist have the chops to make his instrument an extension of his creativity? Is there no limit to his ability to play what he hears in his head? (Like Jimi Hendrix)
Charisma- Does the performer have an aura that carries through the music so you know exactly who it is from the first note? (Like David Bowie)
Control of the Audience- Does the performer work the audience and take them through an emotional roller coaster though the power of the music? (Like James Brown)
Balance- Is the music balanced in such a way that you aren't straining to hear elements in the music, and leads aren't so loud or harsh that they block other elements? (Like Stokowski was able to take a score and make it his own through balance)

On all of those criteria, the Prince DVD scored 10 out of 10. It was better than the album itself. It wasn't just music, it was an experience that grabbed you and took you along for the ride.

Way, way down the list of priorities for a good musical recording is the elements of sound fidelity. Yes, it's nice to have good fidelity. Things like dynamics and extended frequency response and low distortion are very good to have. But if you feel that you have to reject music that scores high on all of those primary criteria because you would prefer to listen to mediocre music recorded in high fidelity, you are listening to all the wrong things.

Music is what matters. In the past I've read posts in audiophile forums where people go on and on about half speed mastered LPs and hi-res audio tracks and how important sound quality is, and I ask them what their favorite albums are. They mention Mannheim Steamroller or The Podunk Philharmonic performing Tchaikovsky's 6th on the "Absolutely Pristine" label recorded in astronomical data rates or some obscure prog rock album on half speed mastered virgin vinyl that was originally recorded in a back room of an apartment in a weekend using equipment that didn't even cost as much as a junker car. It makes me wonder what they are listening to. They sure aren't listening to the music. If I get to the point where I listen to music and all I hear are technical nit picks, and if I can't enjoy a record because my OCD tells me there might be another release of it that is better, then that is the time to shoot me in the head.

My advice to everyone regardless of where they are on their musical journey is to focus on upgrading your music, not your equipment. I've read posts in classical music forums where reviewers go into great detail about a recording, pointing out insightful details that really helped me understand the work better... and then I find out that they are playing the record on a turquoise blue Califone schoolhouse phonograph, or a $100 CD player with little bookshelf speakers. Not to say that sound fidelity doesn't matter, because it does. It's just that there are things that are much more important. Ideally, a sound system should get out of the way so you don't have to think about it. That frees you up to focus on the music. But it's possible to focus on the music without absolutely perfect sound quality.

Thankfully, we live in an age of technology that has reduced problems with fidelity down to the lowest level it's ever been. We have discs we can buy for $6 that hold 70 minutes of music in perfect sound. We can stream music in perfect sound through thin air. We no longer have to contend with wow and flutter, audible distortion, frequency response imbalances or hiss, crackle and pops. Yet some people still listen to the minuscule amounts of noise instead of the music. I don't get it.


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## TheSonicTruth (Dec 8, 2019)

bigshot said:


> Whether a CD is normalized up to 99% or 85% has absolutely no impact on perceived sound quality. There's more than enough dynamic range to normalize sound down to be noticeably quieter. Just balance the volume levels and they might be basically the same. Quieter or louder isn't necessarily better. Sometimes it's just quieter or louder.
> 
> Last night I put on the new remaster of Prince's 1999 album. It sounds fantastic. Much better than any previous release. Pricey, but worth it. Included in the package is a live performance on DVD. It has decent cinematics since it's a multicamera shoot, but the video quality is well below broadcast quality- fuzzy, low contrast, dim. The sound appears to be recorded from a microphone at the mixing board in the middle of the auditorium. There's lots of hall ambience. Lyrics aren't always clear because of the reverberation. Not at all what you would call high fidelity sound. But that DVD is much better than the album because it has every one of the elements necessary for a good recording of music but the least important ones.
> 
> ...




This, succinctly, IS me musically, in a nutshell:


It represents most of what I listen to - 1960s-80s music for the masses.  Infinitely better sounding and certifiably more crankable than anything in the popular genres released in the last fifteen years.

Not the most dynamic, not the highest fidelity, but man, that drum and rhythm section, the space between the notes, takes me somewhere that nothing recent can!  The source on this upload could be a remaster, or not, I don't know.  But certainly, it sounds great even if left as it was in the late '60s.  And it sounds even better on the 25-year old Steppenwolf compilation CD I bought two weeks ago at a library benefit.

I believe stuff like this could can be composed, performed, recorded, mixed, and mastered, again, the way that Steppenwolf piece was fifty years ago, if people put their hearts and minds to it again.


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## TheSonicTruth (Dec 8, 2019)

And another - a bar-room jukebox gem! ...


Again, no Respighi in the dynamics dept, but 3 minutes of just well composed and performed late '70s POP.  Space between the notes in the rhythm dept, drums that sound like, well... DRUMS!  The crack of the snare throughout, and the crash cymbals in the female back-up refrains sufficiently well preserved in both this Walter Egan one hit wonder as well as the Steppenwolf I included one post prior.  Either of these can be appreciated at any volume level, on a wide range of listener platforms, from a ubiquitous iHome dock up to a 200W/channel set of components and speakers.

When I play it over systems at my job, all the kiddy cashiers crick their necks to look where this song is coming from!  Like, "Whoa, what is that, who is that artist?"

Remasterers: Keep your PAWS off my classic jams!


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## gregorio

TheSonicTruth said:


> Which is the way it should be: I run my source inputs low, and keep the final volume high - not maxxed, mind you, but high enough so the amplifier is doing most of the work



What you're describing is called "gain-staging" and the way you've described how you implement it is DEFINITELY NOT how it "should be"!! The DAC output should be very near or at max and your amp lower. If it's the other way around, with your output lower and your amp higher, you will be achieving a worse SNR. The noise floor of your DAC is effectively fixed, so if for example the noise floor of your DAC is say -110dB then with a full scale signal you've got 110dB of signal to noise ratio. If you reduce your source by say 40dB, your DAC still has a noise floor of -110dB but peak level is now -40dB and therefore your SNR would be 70dB. Additionally, depending on exactly how your source is handling the reduction of output volume, quantisation noise could become audible and finally, any noise/interference picked-up between the output of your DAC and input of your amp would also be amplified by an additional 40dB.



TheSonicTruth said:


> I believe stuff like this could can be composed, performed, recorded, mixed, and mastered, again, the way that Steppenwolf piece was fifty years ago, if people put their hearts and minds to it again.



Of course stuff like that could be done but there's two BIG problems which should be OBVIOUS to anyone who understands the facts and thinks about them rationally! 

Firstly, it is OBVIOUSLY NOT just about people putting their hearts and minds to it. Hearts and minds do not pay for the professional composers, arrangers, musicians, producers, engineers and the considerable amount of high quality studio time required, money does. Where do you think that money comes from and why would it be invested in such a project? The commercial music industry is commercial and it's an industry, it's not a charity or other non-profit organisation aimed at creating what you personally (and maybe a few hundred/thousand others) nostalgically desire!

Secondly, as you say, that's how it was done 50 years ago, IE. It's been done. All art but especially popular art (by definition) evolves and moves forward, it doesn't de-evolve and move backwards. I'm sure many painters, if they put their hearts and minds to it, could paint the same as say Turner or Van Gogh did but why bother? It's already been done, the world has moved on and therefore there's little/no financial reward to be had.



TheSonicTruth said:


> [1] This, succinctly, IS me musically, in a nutshell: It represents most of what I listen to - 1960s-80s music for the masses.
> [2] Infinitely better sounding and certifiably more crankable than anything in the popular genres released in the last fifteen years.



1. Exactly, it's YOUR preference of what YOU listen to but neither the industry nor everyone else is defined by your personal preferences. Also, it was OBVIOUSLY music for the 1960s-1980s masses, not for today's masses!

2. If it were "better sounding", then 1960s-1980s music would still be for today's masses (but obviously it isn't) AND, it's demonstrably/"certifiably" LESS "crankable"! Today's popular music is specifically designed from the ground-up to be "crankable" and is indeed cranked far more than music of the 1960s-1980s, as you yourself keep complaining!



TheSonicTruth said:


> [1] Space between the notes in the rhythm dept, drums that sound like, well... DRUMS! The crack of the snare throughout, and the crash cymbals in the female back-up refrains ...
> [2] Either of these can be appreciated at any volume level, on a wide range of listener platforms, from a ubiquitous iHome dock up to a 200W/channel set of components and speakers.



1. You're joking right, have you ever heard what a real drumkit sounds like? Obviously not! The snare drum, which you specifically comment on, could only sound somewhat like that in an anechoic chamber (but has been achieved with extremely close mic'ing and quite heavy compression, to reduce the initial transient!) but the cymbals have considerable acoustic information (been mic'ed from some distance to capture the room acoustics). What you're actually stating is your personal preference for a particular type of artificially manufactured drumkit sound, an old fashioned drumkit sound that defines an old fashioned style/genre of music, certainly NOT what drumkits actually sound like or even what they should sound like (to anyone but you)!

2. But that's not a wide range of listener platforms! I just listened to that track on my laptop and there is literally no kick drum whatsoever and very little bass of any sort, and it would be even worse on many tablets/mobiles. Maybe you can appreciate that but I and most others can't and also, iHome docks are clearly NOT ubiquitous, you just made that up!

As in previous threads, you demonstrate an ignorance of: What "popular music" actually is/means, what the music industry is, how recording/production/mixing/mastering techniques ALL define the different popular genres and that your personal (nostalgic, out of fashion/un-popular) preferences do NOT define what is even practical/possible in today's music industry, let alone what artists should create and what other consumers should listen to!! You can of course remain ignorant (lack understanding of the facts, or simply ignore them) and believe whatever you want but you cannot just keep repeating the same inaccurate (or outright false) assertions here! No one is forcing you to buy new releases or remixed/remastered old releases but paradoxically, if you really were interested in high SQ (rather than just your own personal nostalgia) then your assertions of what should (or rather shouldn't) be done are counter-productive!

ALL this has been explained to you before but you just ignore it and blunder on regardless, effectively thread cr@pping. For that reason and because it's very definitely well off-topic now, if you do just carry on, I'll ask castleofargh to delete it!!

G


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## TheSonicTruth (Dec 9, 2019)

gregorio said:


> Today's popular music is specifically designed from the
> ground-up to be "crankable" and is indeed cranked far
> more than music of the 1960s-1980s, as y




You're joking, right? 

I cannot set my volume control up nearly as high for a modern pop piece(Swift, Bieber, Drake) as I could set it for one of the vintage pop examples above.  It's much louder, crowded, and fatiguing.

As far as the proliferation of those iHome thingies, all one has to do is go down to their Salvation Army or other second hand charity, and find up to a dozen of them on the shelves, next to the dozen computer printers because nobody prints at home anymore! Mostly ones with older iPhone connectors, because their donors have upgraded either to iDocks with the newer connector, or gone Blue Tooth.  Either way, more people listen to music on something in that _size category_ nowadays, than on a full-sized system, meaning music must be squa- ahem - 'engineered' to sound 'good' on them.

Those stores are a good indicator of lifestyle changes!  Thirty years ago, those shelves would have been flooded wih eight-track cartridges, indicating a dying segment of that time.


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## gregorio

TheSonicTruth said:


> [1] You're joking, right? I cannot set my volume control up nearly as high for a modern pop piece(Swift, Bieber, Drake) as I could set it for one of the vintage pop examples above. It's much louder, crowded, and fatiguing.
> [2] As far as the proliferation of those iHome thingies, all one has to do is go down to their Salvation Army or other second hand charity, and find up to a dozen of them on the shelves ...
> [2a] Those stores are a good indicator of lifestyle changes!



1. Exactly, it's much louder because it's already been cranked-up, more so than you can crank-up your examples and therefore your examples are NOT "more crankable" they're LESS "crankable", contrary to your (FALSE) assertion! And, how is it relevant to 16 vs 24bit? So if you're not joking, what ARE you doing??

2. Neither the definition of "ubiquitous" nor the target scenarios of the popular music industry are defined by what's on the shelves of your local second hand shop!
2a. No they're not. They're nowhere even vaguely near as "good an indicator" of popular music trends/evolution as the actual sales and consumption of popular music! Again, if you're not joking, what ARE you doing??

G


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## TheSonicTruth

gregorio said:


> 1. Exactly, it's much louder because it's already been cranked-up, more so than you can crank-up your examples and therefore your examples are NOT "more crankable" they're LESS "crankable", contrary to your (FALSE) assertion! And, how is it relevant to 16 vs 24bit? So if you're not joking, what ARE you doing??
> 
> 2. Neither the definition of "ubiquitous" nor the target scenarios of the popular music industry are defined by what's on the shelves of your local second hand shop!
> 2a. No they're not. They're nowhere even vaguely near as "good an indicator" of popular music trends/evolution as the actual sales and consumption of popular music! Again, if you're not joking, what ARE you doing??
> ...



Do you even know what crankable means - in a consumer context?

It means turning the volume knob up really high because of the feeling you get from a certain song or other piece of audio.

In relatively few instances have I been able to do such, with examples of modern popular genres.


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## gregorio

TheSonicTruth said:


> [1] Do you even know what crankable means - in a consumer context?
> [1a] It means turning the volume knob up really high because of the feeling you get from a certain song or other piece of audio. In relatively few instances have I been able to do such, with examples of modern popular genres.



1. Yes, it means how much louder a piece of music designed for consumer consumption can be made, before distortion becomes excessive. But you apparently don't know what it means because ...
1a. You are describing what YOU are "able to do" with YOUR equipment/volume knob, not what all consumers can do with their equipment.

Thanks for proving my point but enough is enough and it's OFF-TOPIC!!!!

G


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## bigshot (Dec 9, 2019)

TheSonicTruth said:


> I believe stuff like this could can be composed, performed, recorded, mixed, and mastered, again, the way that Steppenwolf piece was fifty years ago, if people put their hearts and minds to it again.



There are lots of cover bands dressing up like The Beatles, Jimi Page and Elton John and doing shows of sound-alikes. You'll find them at casinos in Las Vegas that cater to the over 50 crowd. Other than that, the world of music continues to turn and move on. You can grow and change with it, or you can get left behind to live in your memories. Either way is fine. I choose "all of the above".


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## bigshot

Chris Kaoss said:


> I did comparison between the first queen releases and the 2011 remastered ones.
> To me, the former sounds better couz they are more dynamic and natural then the latter.
> It doesn't belong to all remastered old pieces, but with the first releases of cd's of queen or that decade, you would hear it the best.



The best sounding Queen releases I've heard are the Night At The Opera SACD and the Queen Video Hits 1 & 2 DVDs. Have you heard them? I'm told there is a Japanese 15 album SACD series that is better than the CDs on some albums. I don't have that myself though. The Video Hits DVDs are a revelation. Fantastic multichannel mix. Couldn't sound better.


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## TheSonicTruth

gregorio said:


> 1. Yes, it means how much louder a piece of music designed for
> consumer consumption can be made, before distortion becomes excessive.
> But you apparently don't know what it means because ...
> 1a. You are describing what YOU are "able to do" with YOUR equipment/volume
> ...



You just love to twist words. Your reputation as a mind-gamer and manipulator of words is already all over the internet, don't worry.

I'll get Barry, Bob, and Ethan back on here, and straighten all of you out!


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## TheSonicTruth

bigshot said:


> The best sounding Queen releases I've heard are the Night At The Opera SACD and the Queen Video Hits 1 & 2 DVDs. Have you heard them? I'm told there is a Japanese 15 album SACD series that is better than the CDs on some albums. I don't have that myself though. The Video Hits DVDs are a revelation. Fantastic multichannel mix. Couldn't sound better.




Of course you hawk the latest versions of Queen's albums and other material - you're a salesman!

Anyone who proclaims the original to sound better than the remaster is 'wrong' in yours and Gregorio's estimation.


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## bigshot (Dec 9, 2019)

The Queen Video Hits 1 & 2 DVDs were released in 2002. The SACD was released in Japan originally around 2010. These were fresh remixes done for the audiophile market. The first CD release of Night At The Opera was in 1989. It isn't the best sounding version by a long shot. Until the SACD came out, The MFSL CD released in 1992 was the best. This album has been remastered for just about every medium and market multiple times over the years. I'm citing the versions I have heard that sound the best. I happen to know a little bit about the various releases of this music. I have the original LP release, the MFSL LP, the original CD release, the MFSL CD and the SACD of Night At The Opera. Each one has been an improvement on the one before.

You've got the idea that the first release of everything was the best sounding and remasters are all bad. You flat out don't know what you're talking about. It's a case by case basis depending on the album. You can't generalize like that.

Also, I wasn't talking to you. I was talking to the poster who was interested in the best sounding releases of Queen.


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## gregorio

TheSonicTruth said:


> [1] You just love to twist words.
> [2] Your reputation as a mind-gamer and manipulator of words is already all over the internet, don't worry.
> [3] I'll get Barry, Bob, and Ethan back on here, and straighten all of you out!
> [4] Of course you hawk the latest versions of Queen's albums and other material - you're a salesman!



1. How does being a hypocrite help your case, especially in this subforum? You then wrote: "_Anyone who proclaims the original to sound better than the remaster is 'wrong' in yours and Gregorio's estimation._" - Which isn't just twisting words, it's an outright lie!
2. Sure it is.
3. Of course you will.
4. Of course he is, bigshot is obviously a salesman for Queen's record label .... and apparently also a salesman for all the other record labels of all the other remasters he's mentioned! 

Why do you always choose this route: Thread cr@p any post/thread you think you can bend to your hobby horse of remasters, regardless of how off-topic. Base your arguments on ignorance and falsehoods, while dismissing the actual facts, regardless of the number of times and different ways they're explained to you. Dismiss those who know the facts as "salesmen" for every label ever to have issued a remaster. Resort to insults, hypocrisy or outright lies and progressively make yourself appear more and more foolish? And, you don't just do that here, according to you, you've done exactly the same thing on a number of other (pro audio) sites, with unsurprisingly exactly the same results! Are you doomed to forever making yourself appear ignorant and foolish and effectively being a thread cr@pper/troll? Will you never learn, or at the very least change your approach so you don't keep getting this exact same result?

For your own benefit, as well as the benefit of everyone else and this thread: ENOUGH, post on-topic or don't post at all!!!

G


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## castleofargh

*modo violence: *
It's been going on for 2 pages, OP asked several times to get back on topic and should be somewhat in charge of his own thread(although @gregorio, trying to have the last word on the off topic is obviously going to trigger people to reply. As the philosopher said "let it go! let it go!"). 
Maybe there should be a limit like that for off topics, and after a given limit like 2 or 3 pages, I just mindlessly delete stuff and kick people out if they keep on feeding the same off topic?

Anyway. Back to bit depth please.


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## castleofargh

I deleted 11 posts.


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## TheSonicTruth

castleofargh said:


> I deleted 11 posts.




You forgot to delete the salesman posts about Queen material. They had nothing to do with bit depths either.


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## Davesrose (Dec 11, 2019)

bigshot said:


> Also, I wasn't talking to you. I was talking to the poster who was interested in the best sounding releases of Queen.



Lets not also forget the soundtrack to Bohemian Rhapsody.  I bought it on streaming UHD, and the iTunes version is Dolby Atmos.  The Live Aid scenes are really something to behold.  It has a great sense of ambiance of panning around in "3D" (when it comes to audience and acoustics coming from around and above).  It's a great comparison to a Queen BD recording I have of "Rock Montreal": where the main feature was filmed with 35mm (and not sure what source audio was).  The visual and audio dynamics of the main feature is great.  It also includes the recording of Live Aid: where the visual is pretty lacking and looks to be upscaled analog video....but audio is still good and worth it for any Queen fan to compare that original recording to the modern recreation.

When it comes to the whole topic of this thread, I think it's another academic subject that has no final answer.  My favorite ever Sony Discman was 1bit processing, and I also have heard DACs I like that are 24bit upscaling: there's all different methods for processing, and you can have great sounding sound at 1bit, 16bit, 24bit, or 32bit.  I'm more of an authority with visual computer graphics, and see that the premise of this thread was that it's easier to see bit depth with images.  That's pretty simplistic: I think that reference was with early computers that would display monochrome (1bit) vs 4bit on up to what jpeg uses (8bit per channel).  Now in this age, people are starting to understand HDR (much to my enjoyment as a photographer).  Now consumers are arguing about 10bit per channel HDR vs Dolby Vision 12 bit per channel color depth (where there is "tone mapping"...adjusting contrast to native range)...visuals are rightly so needing precedence.  However, with my background in photography, I do like full contrast range.  I do find that some color grading for HDR tends to be too high contrast. (perhaps something analogous to compressed audio).


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## bigshot

Davesrose said:


> Lets not also forget the soundtrack to Bohemian Rhapsody..



thanks for the tip. I haven’t seen that yet. I’ll check it out!


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## gregorio

Davesrose said:


> [1] When it comes to the whole topic of this thread, I think it's another academic subject that has no final answer.
> [2] My favorite ever Sony Discman was 1bit processing, and I also have heard DACs I like that are 24bit upscaling: there's all different methods for processing, and you can have great sounding sound at 1bit, 16bit, 24bit, or 32bit.
> [3] I'm more of an authority with visual computer graphics, and see that the premise of this thread was that it's easier to see bit depth with images. That's pretty simplistic: I think that reference was with early computers that would display monochrome (1bit) vs 4bit on up to what jpeg uses (8bit per channel). Now in this age, people are starting to understand HDR ...



1. You're of course free to think whatever you want but the actual reality/facts prove that there IS a "final answer". I can understand how/why you could think there isn't though, which is why I'll respond to point 3 before point 2 ...

3. There are many similarities between digital photography and digital audio, as well as many similarities between how we see/perceive images and how we hear/perceive sound and therefore, we can potentially have many valid analogies between the two. However, there are also many differences between the two (some of which are quite profound) and therefore, potentially many analogies between them that are only partially valid and some that are quite profoundly invalid! I believe this is the trap you may have fallen into, which explains your conclusion of there being "no final answer". Unfortunately, I am NOT an authority with digital photography/computer graphics, so my terminology and description of digital imaging may not be entirely correct but I'm going to try to give a couple of examples: 

One of the most major differences is what we're actually converting into digital data to start with. With photography our source "format" is light, which is waves/packets (photons) of electromagnetic energy, that we can convert into digital data with sensors. With audio, our source "format" is sound pressure waves, which is mechanical/kinetic energy travelling through a medium BUT, we can only convert electromagnetic energy into digital data with sensors, not mechanical energy, so we CANNOT convert sound into digital data! The solution is simple in theory and didn't need discovering because it already existed, nearly 150 years ago and 50 years before digital audio was first conceived: We first convert this mechanical energy into electromagnetic energy (specifically electricity), a process called transduction, then we can convert this electromagnetic energy to digital data and of course do the reverse conversion and transduction to reproduce the sound waves. However, this has consequences/limitations compared to digital imaging (which doesn't involve transduction) because transduction is highly inefficient (due to the laws of motion/kinetics) and therefore requires relatively massive amounts of amplification, which in turn causes even more limitations (due to other laws of physics, such as thermal noise).

Another major difference is the different response of our eyes and ears, for example: Our ears have a freq response range of about 20 kilo-hz and can resolve that range into about 10,000 different pitches. Our eyes have a freq response range of about 320 tera-hz and can resolve that range into about 10,000,000 different colours. So 16bit, which can represent ~65,000 different colours, is about 150 times fewer colours than the human eye can differentiate but about 6.5 times more pitches than the human ear can differentiate. So, 16bit is definitely "low-res" for the human eye and 24bit, with 16,000,000 colours, is about 1.6 times greater than required for visual "hi-res". However, 16bit for the human ear is already 6.5 times greater than required for audio "hi-res"! This isn't an entirely fair comparison though, because we do not use bits to directly represent the frequencies (pitches/colours) in digital audio but to represent the amplitude of the transduced electrical voltage (from which freq is derived), which is another of the differences between digital audio and digital imaging. So as mentioned above, this would effectively represent a "partially valid" analogy and demonstrates the dangers of digital visual and audio analogies! 

2. There is no 24bit upscaling in digital audio! Maybe in digital imaging there is, maybe you can interpolate colours between the ~65,000 values available and write those colour values to the 16,000,000 (24bit) available values but digital audio doesn't work that way. If you "upscale" 16bit audio to 24bit nothing changes, there is no "upscaling", you just get 16bit audio in a 24bit container, the extra 8bits (LSBs) are just padded with zeros. 

G


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## castleofargh

Davesrose said:


> Lets not also forget the soundtrack to Bohemian Rhapsody.  I bought it on streaming UHD, and the iTunes version is Dolby Atmos.  The Live Aid scenes are really something to behold.  It has a great sense of ambiance of panning around in "3D" (when it comes to audience and acoustics coming from around and above).  It's a great comparison to a Queen BD recording I have of "Rock Montreal": where the main feature was filmed with 35mm (and not sure what source audio was).  The visual and audio dynamics of the main feature is great.  It also includes the recording of Live Aid: where the visual is pretty lacking and looks to be upscaled analog video....but audio is still good and worth it for any Queen fan to compare that original recording to the modern recreation.
> 
> When it comes to the whole topic of this thread, I think it's another academic subject that has no final answer.  My favorite ever Sony Discman was 1bit processing, and I also have heard DACs I like that are 24bit upscaling: there's all different methods for processing, and you can have great sounding sound at 1bit, 16bit, 24bit, or 32bit.  I'm more of an authority with visual computer graphics, and see that the premise of this thread was that it's easier to see bit depth with images.  That's pretty simplistic: I think that reference was with early computers that would display monochrome (1bit) vs 4bit on up to what jpeg uses (8bit per channel).  Now in this age, people are starting to understand HDR (much to my enjoyment as a photographer).  Now consumers are arguing about 10bit per channel HDR vs Dolby Vision 12 bit per channel color depth (where there is "tone mapping"...adjusting contrast to native range)...visuals are rightly so needing precedence.  However, with my background in photography, I do like full contrast range.  I do find that some color grading for HDR tends to be too high contrast. (perhaps something analogous to compressed audio).


Bits on a picture are allocated in a different way compared to bits in PCM. So reducing the number of bits will have a radically different impact and "bit" just becomes a false friend to anybody who isn't already familiar with how each digital system operates.


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## board (Dec 12, 2019)

old tech said:


> Tell me more...  I recall he piked out on James Randi challenge on cables but haven't heard of this one.



Sorry for the late response with this one, and I hope it's not too off-topic, castleofargh.
Although I find Michael Fremer to be one of the most obnoxious loudmouths in the entire audio world, I do actually mostly side with him on those two issues:

As far as I could read, the cable challenge with James Randi ended because Pear Audio wasn't willing to provide the cables, and no one else was willing to provide the Pear Cables either, and apparently James Randi didn't want to spend the $7000 on them himself. Fremer then suggested they use his own Tara Labs cables ($25,000). Randi first considered accepting this suggestion, but then in the end declined and announced that the challenge was over.
So Fremer actually didn't weasel out of it as far as I could understand. I think Randi had been told that the cables could change the sound, which is true - cables can change the volume level or the frequency response. Ethan Winer also mentioned this in his Null Tester video on Youtube, and someone on Hydrogen Audio also succesfully ABX'ed two sets of speaker cables. His subsequent measurement showed a marked difference in frequency response. The Audio Critic also published measurements of cables that showed different frequency responses.
So, I mention this because I think that what should have happened was that James Randi gave Fremer the chance to test the Monster cables against his Tara Labs, _given_ that they first measured the cables and found that they measured the same within audibility for frequency response, volume level, capacitance, inductance and resistance. Remember that Fremer often brags about how he's able to hear things that can't be measured. Such a test could prove or disprove it.

As for the other story, Fremer has recounted this several times, as he was apparently quite traumatized by it. I wish he had submitted himself to a mental institution to get better and had stayed there since.
All jokes aside, apparently he and John Atkinson was blind testing amplifiers at an AES event, and Fremer got 5 amplifiers out of 5 correct. John Atkinson got 4 out of 5 correct. Stanley Lipshitz then proclaimed that they were just "lucky coins" and then dismissed their results.
If you want to see it in Fremer's own words, here's one of his rundowns (I've seen him tell the story, full of self-pity, several other times):

https://www.stereophile.com/content/blind-listening-letters

Again, what I think should have happened was that they did further testing. As far as I understand, the test was to only play each amplifier once, and then you could of course be a lucky coin. If they had done an ABX test with 16 trials for each amplifier it would have been very difficult to dimiss someone as a lucky coin if they had had a very good result.
Personally, I believe there are audible differences between many amplifiers (but not all), and this is most likely due to an altered frequency response because of the speaker load. As far as I know, very few amplifiers will remain flat within 0.1 dB when subjected to a real-world speaker load. And of course speaker loads differ from speaker to speaker. That would explain a lot about so-called "synergy".
There might also be other things than frequency response that affect the sound of an amplifier. I can't say. But Bob Carver's amplifier challenge with Stereophile was a very interesting read, as he found other sources of audible differences, although he was trying to emulate a tube amp, so distortion was a big issue, and so did phase seem to be as well, surprisingly.
I can send you a link to this if you like.

What I found very, very disheartening about these two stories is that it also shows that objectivists can be as stubborn and close-minded as subjectivists .


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## Davesrose

gregorio said:


> 1. You're of course free to think whatever you want but the actual reality/facts prove that there IS a "final answer". I can understand how/why you could think there isn't though, which is why I'll respond to point 3 before point 2 ...
> 
> 3. There are many similarities between digital photography and digital audio, as well as many similarities between how we see/perceive images and how we hear/perceive sound and therefore, we can potentially have many valid analogies between the two. However, there are also many differences between the two (some of which are quite profound) and therefore, potentially many analogies between them that are only partially valid and some that are quite profoundly invalid! I believe this is the trap you may have fallen into, which explains your conclusion of there being "no final answer". Unfortunately, I am NOT an authority with digital photography/computer graphics, so my terminology and description of digital imaging may not be entirely correct but I'm going to try to give a couple of examples:
> 
> ...



I don't believe I've fallen in a trap, as you put it.  I don't know why you always assume other people are ignorant of facts, and there's not some clarity in reading what they are trying to write.  For example, if you took my post in context, you would see that I talked about DACs which process audio in different manners (using 1bit, 16bit, 24bit sampling). I implied they can all be valid in producing  audio that's subjectively good and not limiting with our hearing.

When it comes to photography, there are a few things you're overlooking.  I'm not sure why you say audio needs to be converted to digital to begin with: photography has to as well, and it's more complex.  The most common digital sensor uses a Bayer filter (an array of 1 red, 1 blue, to 2 green filters covering a photosite).  Every color pixel has sub-pixels of photo diodes (a photosite), which similar to a microphone, conducts electricity based on intensity of light (instead of sound).  Black and white photography would be more analogous to sound recording, as it doesn't need the use of filters (or with color film, there were three seperate photosensitive layers).  Behind the sensor is an analog to digital converter.  Over the years, digital cameras have improved in both resolution and dynamic range (with photosites shrinking, and improvements in circuit designs of the ADC).  Also, like audio, image files are now considered by dynamic range.  In the early days of computer images, computer hardware was pretty limited, and there could be color palates of 4, 16, 32, or 256 colors total.  But a millennial wouldn't have any experience with the type of images that couldn't be photo realistic.  A standard jpeg has a possibility of 16 million colors....but only 8 stops of light (or 256 shades of gray).  That's pretty limiting for exposing a scene that might have a higher dynamic range (for example, a sunny day in which you want to expose both the sky and subjects in the shade).  The human eye is thought to be capable of seeing up to 20 stops of light.  Current cameras can expose 14 stops of light (or 16,384 shades of light) in one exposure.  For computer raytrace rendering, 32bit per channel files are used for more realistic simulations of light (which can get up to 4.29 billion shades of tone).  There has been processing (either automatic or manual adjustments) to shrink higher DR images to 8bpc space (which was the only standard for monitors).  Now TVs and monitors can get up to 10bpc space, and there is automatic or manual processing to shrink 16bpc, 14bpc, 12bpc files to that as well.  Just like mixing influences how the sound is heard....so too how processing (known as tone mapping) is done, can greatly influence how an image looks.


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## bigshot

If someone can actually hear things that other humans can't, they shouldn't be doing stunts with magicians or doing it at audio sales events. They should be submitting themselves for scientific testing to determine exactly what they can hear that other people can't. Then perhaps we could figure out how to make equipment audibly transparent for the last .0001%.


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## gregorio

Davesrose said:


> [1] I don't know why you always assume other people are ignorant of facts, and
> [1a] there's not some clarity in reading what they are trying to write. For example, if you took my post in context, you would see that I talked about DACs which process audio in different manners (using 1bit, 16bit, 24bit sampling). I implied they can all be valid in producing audio that's subjectively good and not limiting with our hearing.
> [2] When it comes to photography, there are a few things you're overlooking.
> [3] I'm not sure why you say audio needs to be converted to digital to begin with:
> ...



1. I don't necessarily assume other people are ignorant of the facts! If someone makes some assertion that is incorrect, I assume that EITHER they're ignorant of the facts, that they're not ignorant of the facts but don't really understand them or that they're not ignorant of the facts, do understand them but are erroneously dismissing them for some reason. What rational alternative am I missing?
1a. I didn't state your post didn't have some clarity and I didn't dispute this assertion, only some specific facts within your assertion, your analogy with digital imaging and your conclusion/assertion of there being "no final answer".

2. More than a few I should imagine!

3. But I didn't say that! What I effectively (tried to) say is that to begin with we do NOT even have audio, we have sound pressure waves that needs to be transduced into audio and only then can it be converted to digital. Therefore:
3a. "No", because photography does not have to be transduced as well. Both digital audio and (as I understand it) digital imaging involve an analogue stage and then an ADC but a major difference is that digital imagining involves converting between different types of the same form of energy (light and electricity which are both electromagnetic energy), while digital audio recording involves converting between two different forms of energy (mechanical and electromagnetic).
3b. No, a photodiode is very significantly different to a microphone. As far as I'm aware, photodiodes are microscopic devices with no moving parts (solid state) which operate at the quantum level converting a photon/s into an electron/s. Microphones do NOT conduct electricity based on the intensity of sound, they generate electricity based on the "intensity" of movement of a diaphragm. A microphone therefore has to first convert variations in sound pressure into the mechanical motion of a diaphragm and then convert that mechanical motion of the diaphragm into electricity. So, we're dealing with relatively huge mechanical devices, subject to all the limitations of the laws of physical motion/transduction, all of which results in relatively massive inefficiency (compared to image sensors) in the generated analogue signal, which then of course is the input for conversion to digital data. In the practical application of digital audio, it's this inefficiency which defines system limitations, not the ADC, DAC or number of bits (beyond 16). A somewhat better analogy with an image sensor would have been a tape recorder, which also converts between different types of the same form of energy (electrical and magnetic) but it's still a rather poor analogy as tape recorder performance is still reliant on mechanical forces (physical properties of the tape itself, plus friction, tape alignment, motor/speed, etc.).

4. Another significant difference! Over the years, digital audio has changed significantly but the resultant output resolution and dynamic range has barely changed at all. Even in the earliest days of consumer digital audio (CD) the hardware was capable of 16 bits, near perfect resolution and a dynamic range in excess of both the limitations imposed in practice by microphones or that would be experienced in the real world (at a gig).

5. And another reason to be wary of comparing digital imaging with digital audio, that I've already mentioned. With digital imaging,16 million colours (24bit) still does not cover all the capabilities of the human eye. If, as you say, it only provides 8 stops and the human eye is capable of 20 stops, then it's still a long way from the capabilities of the human eye. For the human ear though, 16bit is already beyond it's capabilities and a long way beyond "comfortable".

G


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## Davesrose (Dec 12, 2019)

gregorio said:


> 1. I don't necessarily assume other people are ignorant of the facts! If someone makes some assertion that is incorrect, I assume that EITHER they're ignorant of the facts, that they're not ignorant of the facts but don't really understand them or that they're not ignorant of the facts, do understand them but are erroneously dismissing them for some reason. What rational alternative am I missing?
> 1a. I didn't state your post didn't have some clarity and I didn't dispute this assertion, only some specific facts within your assertion, your analogy with digital imaging and your conclusion/assertion of there being "no final answer".
> 
> 2. More than a few I should imagine!
> ...



The difference is I clearly was referring to audio processors, and you conflated my short statement with file bit depth. Light is not the same form of energy as electricity. The point was that although not the same, there is still a conversion to electric signal and an analog to digital conversion happening in digital photography.  And again, modern images are thought of in bits per channel: not total bits of image (for example, there are images that are a total of 32bit with 8bpc RGB and alpha transparency).  And I can leave it at that.


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## bigshot (Dec 12, 2019)

I don't see much relationship between resolution in photography and sound fidelity. With photographs you can get right up close and squint at a photo and see more resolution. With sampling rate, past the point your ears can hear, there is nothing you can do to perceive more zeros and ones. And with photos you can turn up the brightness to see detail in shadows, but with sound, if you turn up the volume to hear the wider bit depth, you get blasted deaf by the peaks.

But in either case, once you get beyond a certain point, for the purposes intended, it becomes pointless overkill.


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## Davesrose

bigshot said:


> I don't see much relationship between resolution in photography and sound fidelity. With photographs you can get right up close and squint at a photo and see more resolution. With sampling rate, past the point your ears can hear, there is nothing you can do to perceive more zeros and ones. And with photos you can turn up the brightness to see detail in shadows, but with sound, if you turn up the volume to hear the wider bit depth, you get blasted deaf by the peaks.
> 
> But in either case, once you get beyond a certain point, for the purposes intended, it becomes pointless overkill.



I wasn't talking about resolution.  Dynamic range is a separate subject.  With digital imaging, resolution is the number of pixels or dots in a given area.  It's as different as resolution in sound is compared to dynamic range.  Instead of it meaning the range from the softest to loudest peak, DR in imaging is the value range of blackest point to brightest point.  There is no such thing as infinite resolution: if you were able to go up close to a billboard (which can be as little as 20dpi), you would start seeing softness before just seeing individual points.  Also, content creators need larger dynamic range of their source files to be able to pull up detail in shadows or recover blown highlights (especially when converting HDR images to monitor color spaces).  Try post processing an image that doesn't have enough DR, and you'll either see noise or black splotches in shadows as well as complete white splotches in highlights.  This is just with photography: HDR is also used in 3D rendering for realistically simulating environmental conditions (where it's applicable to have DR that can exceed what the human eye is able to accommodate).  Digital displays are still in their infancy with having HDR, and it will be interesting to see if images appear even more "lifelike" as contrast ranges continue to improve (and color grading might not side with high key contrast for some content).  I realize there is not a 1:1 comparison of digital sound reproduction to image reproduction.  However, it is valid to understand respective technology and see what analogies there are.


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## bigshot (Dec 12, 2019)

But when dynamic range falls below a certain point, you can't see it or hear it without grossly adjusting the brightness or volume level. Dynamic range is only important if you are editing images or mixing a recording. After you finalize an image or export a mix, you don't need it at all. The eyes and ears can only detect a certain amount of dynamics at one time, The whole point of editing an image or mixing audio is to get the result within the comfortable range of the eyes and ears, and that is well within the range of a normal consumer image or sound format.


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## Davesrose

bigshot said:


> But when dynamic range falls below a certain point, you can't see it or hear it without grossly adjusting the brightness or volume level. Dynamic range is only important if you are editing images or mixing a recording. After you finalize an image or export a mix, you don't need it at all. The eyes and ears can only detect a certain amount of dynamics at one time, The whole point of editing an image or mixing audio is to get the result within the comfortable range of the eyes and ears, and that is well within the range of a normal consumer image or sound format.



Dynamic range still isn't useless to the consumer: there still needs to be a minimum.  We haven't reached the DR limits of vision with imaging, and there are very different applications where resolution comes into play.  A 46 mega pixel camera is over-kill for an image that gets scaled down to say a 1024 web image. But a person may still want that original size for making a high quality print at a large size.   With a static image, it is easier to layer different exposures to get detail in a dark lit room as well as detail in a bright window.  With single exposures or video, you may have to compromise and blow out parts of the scene to adequately expose for the subject.

While imaging still has room for improvement, I would agree that we've reached the limits of fidelity with audio in relation to human perception....and I leave it to others who may want to expose themselves to sound peaks over 100dB or want to argue what processing method (in relation to 1bit to 16bit to 24bit) is "best".


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## bigshot

I think people should submit themselves to peaks over 100dB. If they expose other people to that, it might be classified as assault. For the purposes of listening to music in the home, CD quality sound is already overkill.


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## Davesrose (Dec 13, 2019)

bigshot said:


> I think people should submit themselves to peaks over 100dB. If they expose other people to that, it might be classified as assault. For the purposes of listening to music in the home, CD quality sound is already overkill.



At least one of the advantages of DR with imaging is that you still won't ruin your eyes staring at a TV (yes, even now with HDR displays vs older old SD NTSC screens in which moms said you'd ruin your eyes).  As a small aside, I remember a time in which as a kid I was staring in front of a microwave and my mom exclaimed I was ruining my eyes.  Even then, I had known enough about radiation and shielding. When it comes to music, CD can be a great standard.  Now it is more convoluted when it comes to streaming and compression.  At least now that high bandwidth is becoming a norm, we're not having any issues with compression artifacts with music let alone "acceptable" 4K HDR Atmos video.


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## old tech

Davesrose said:


> At least one of the advantages of DR with imaging is that you still won't ruin your eyes staring at a TV (yes, even now with HDR displays vs older old SD NTSC screens in which moms said you'd ruin your eyes).  As a small aside, I remember a time in which as a kid I was staring in front of a microwave and my mom exclaimed I was ruining my eyes.  Even then, I had known enough about radiation and shielding. When it comes to music, CD can be a great standard.  Now it is more convoluted when it comes to streaming and compression.  At least now that high bandwidth is becoming a norm, we're not having any issues with compression artifacts with music let alone "acceptable" 4K HDR Atmos video.


Off topic, but decades ago that whole thing about microwave ovens and lethal radiation was a widely accepted myth.  I still remember the concerns people had with door seals and worrying about leaking radiation - even the possibility of putting radiation into the cooking food from normal use.  Of course, microwave radiation is non ionising and only likely to cause heat discomfort if one was very close to a badly leaking door.


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## Davesrose (Dec 13, 2019)

old tech said:


> Off topic, but decades ago that whole thing about microwave ovens and lethal radiation was a widely accepted myth.  I still remember the concerns people had with door seals and worrying about leaking radiation - even the possibility of putting radiation into the cooking food from normal use.  Of course, microwave radiation is non ionising and only likely to cause heat discomfort if one was very close to a badly leaking door.



There might have been more issues with the earliest microwaves, but I was a child of the 80s.  At that point, I think the microwave was a fixture that was completely safe.  My mom is now known as an actual Luddite (she's actually destroyed 2 computers from physical means).   I think she did think microwaves could continue to project radiation right in front: where in fact the radiation is blocked by the metal block screen you see in the window (acting as a Faraday device).


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## old tech

Davesrose said:


> At least one of the advantages of DR with imaging is that you still won't ruin your eyes staring at a TV (yes, even now with HDR displays vs older old SD NTSC screens in which moms said you'd ruin your eyes).  As a small aside, I remember a time in which as a kid I was staring in front of a microwave and my mom exclaimed I was ruining my eyes.  Even then, I had known enough about radiation and shielding. When it comes to music, CD can be a great standard.  Now it is more convoluted when it comes to streaming and compression.  At least now that high bandwidth is becoming a norm, we're not having any issues with compression artifacts with music let alone "acceptable" 4K HDR Atmos video.


Off topic # 2.  I find your posts around photography and video quite interesting, it is not a subject I know much about.  Tell me something, I upgraded my monitor the other week to a Dell 32" 4k screen.  Why is it that I don't notice any difference to my previous 31" UHD monitor, even with video content?  Do I need 4k content to appreciate any difference?


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## Davesrose

old tech said:


> Off topic # 2.  I find your posts around photography and video quite interesting, it is not a subject I know much about.  Tell me something, I upgraded my monitor the other week to a Dell 32" 4k screen.  Why is it that I don't notice any difference to my previous 31" UHD monitor, even with video content?  Do I need 4k content to appreciate any difference?



There's going to be some issues here.  First, UHD now is essentially 4k for consumer standards...it also includes close to 8K as well.   But the minimum horizontal pixels for "UHD" is 3840px while cinema standards are different (they're not locked into 16:9 aspect, and they have horizontal resolution of at least 4096px).  I'm not sure how there can be resolution differences between the base "UHD" (still around 4K at 3840px) vs "4K" (maybe at best 4096px wide).  If it's really as minute as a small resolution shift...then that wouldn't show much difference.  I suspect that instead of resolution, you might be wondering about HDR: which can show clear differences.  You mention a Dell screen: are you a Windows user?  If so, you can have HDR10 color space quite easily with the latest versions of Windows 10.  If you're up to date, you should be able to see HDR settings with display settings.  It will let you adjust contrast for SDR vs HDR....and another great thing about Windows is that I find no problems with playing movie files that are TrueHD base Atmos output.


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## old tech

Davesrose said:


> There's going to be some issues here.  First, UHD now is essentially 4k for consumer standards...it also includes close to 8K as well.   But the minimum horizontal pixels for "UHD" is 3840px while cinema standards are different (they're not locked into 16:9 aspect, and they have horizontal resolution of at least 4096px).  I'm not sure how there can be resolution differences between the base "UHD" (still around 4K at 3840px) vs "4K" (maybe at best 4096px wide).  If it's really as minute as a small resolution shift...then that wouldn't show much difference.  I suspect that instead of resolution, you might be wondering about HDR: which can show clear differences.  You mention a Dell screen: are you a Windows user?  If so, you can have HDR10 color space quite easily with the latest versions of Windows 10.  If you're up to date, you should be able to see HDR settings with display settings.  It will let you adjust contrast for SDR vs HDR....and another great thing about Windows is that I find no problems with playing movie files that are TrueHD base Atmos output.


Cheers, that makes sense. The native resolution of the monitor is 3840 x 2160. I have the current version of windows 10 but use the HDR settings on the Nvidia control panel.  

I thought the screen type would also have an impact on picture quality.  I know it is not like for like, but to my eyes my 42" 1080p plasma TV has better picture quality than the monitor.  A bit like I would expect a 4k OLED TV to have better picture quality than a 4k LCD TV.


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## Davesrose

old tech said:


> Cheers, that makes sense. The native resolution of the monitor is 3840 x 2160. I have the current version of windows 10 but use the HDR settings on the Nvidia control panel.
> 
> I thought the screen type would also have an impact on picture quality.  I know it is not like for like, but to my eyes my 42" 1080p plasma TV has better picture quality than the monitor.  A bit like I would expect a 4k OLED TV to have better picture quality than a 4k LCD TV.



Yep, that resolution is going to be the standard for "4K" UHD.  Cinema cameras can now record in higher resolutions, but keep in mind that digital intermediates can be lower resolutions (as trying to layer all the VFX and such is so intense that many productions still process in 2K HDR for their intermediates and then upres to 4K still).  The Japanese did set standards for the 8K UHD broadcast standards....but there's still much room to be the final outcome.  I think when it comes to cinema cameras...RED continues to set the standard: they continue to offer higher resolutions and score high for RAW DR video codecs.

When it comes to best display, I've actually read that TVs are now better at HDR than computer monitors.  I did invest in a Panasonic plasma HDTV early on, and it's stll going strong.  It's just that this year, I finally did upgrade my main TV a larger OLED TV.  When it comes to comparisons, I don't have any regrets: OLED easily reproduces high IQ for SD vs HD vs UHD.  It's also better with full 4K HDR content.  But I've also got the disclaimer that source makes a difference.  A highly compressed SD video is going to look like crap: my LG 4K player does a good job of playing well mastered DVDs, BDs, and 4K disks....but still you can see and possibly hear some differences with comparable early masters with artifacts.  For most my video playing, I stream material from an Apple TV 4K and play local content from an Intel NUC (that is great with lossless audio and HDR content).


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## TheSonicTruth (Dec 13, 2019)

old tech said:


> Cheers, that makes sense. The native resolution of the monitor is 3840 x 2160. I have the current version of windows 10 but use the HDR settings on the Nvidia control panel.
> 
> I thought the screen type would also have an impact on picture quality.  I know it is not like for like, but to my eyes my 42" 1080p plasma TV has better picture quality than the monitor.  A bit like I would expect a 4k OLED TV to have better picture quality than a 4k LCD TV.





Davesrose said:


> Yep, that resolution is going to be the standard for "4K" UHD.  Cinema cameras can now record in higher resolutions, but keep in mind that digital intermediates can be lower resolutions (as trying to layer all the VFX and such is so intense that many productions still process in 2K HDR for their intermediates and then upres to 4K still).  The Japanese did set standards for the 8K UHD broadcast standards....but there's still much room to be the final outcome.  I think when it comes to cinema cameras...RED continues to set the standard: they continue to offer higher resolutions and score high for RAW DR video codecs.
> 
> When it comes to best display, I've actually read that TVs are now better at HDR than computer monitors.  I did invest in a Panasonic plasma HDTV early on, and it's stll going strong.  It's just that this year, I finally did upgrade my main TV a larger OLED TV.  When it comes to comparisons, I don't have any regrets: OLED easily reproduces high IQ for SD vs HD vs UHD.  It's also better with full 4K HDR content.  But I've also got the disclaimer that source makes a difference.  A highly compressed SD video is going to look like crap: my LG 4K player does a good job of playing well mastered DVDs, BDs, and 4K disks....but still you can see and possibly hear some differences with comparable early masters with artifacts.  For most my video playing, I stream material from an Apple TV 4K and play local content from an Intel NUC (that is great with lossless audio and HDR content).



After ten years of doing basic setups and full calibrations on consumer TVs ranging from WW2-based CRT tube sets to the latest OLEDs, as well as desktop monitors,  I've come to the conclusion that the picture menu settings matter far more than how many lines or dots of resolution a TV or monitor is capable of.  Not to mention, well-produced content in the first place.

That's right - I said it: Settings/calibration makes the biggest impact on enjoyment of your display(and are also good for your eyes and for extending the life of the display).

Most people don't even know their TV _has _a menu, let alone know what things like Contrast, Brightness, and color actually do.    Typically, the sets are just left in Retail mode, or 'Vivid' or 'Dynamic', with as many as a dozen so-called enhancers engaged.  This airport runway light bright over colored cartoon clusterf-- is what the general public has come to associate with HD, and UHD, 4K, 8K, etc.  It's why they think the calibrated image 'looks wrong' to them.

Just taking the TV out of 'Dynamic' mode, and turning off crap like 'Skin Enhancer', 'Motion sensing', 'Noise Reduction'(one TV had 4 flavors of NR - I disabled them all), 'Super-Duper Contrast'(I made that one up!) - the picture already looked more natural.  And that was _before_ running alignment patterns for the basic settings(Contrast, Brightness, Sharpness, etc.).  As many as a dozen enhancers under the Advance Menu where people are afraid to venture: All non-standard settings.  _All disabled._

And the rest of what I do I'll keep secret.


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## old tech (Dec 13, 2019)

TheSonicTruth said:


> After ten years of doing basic setups and full calibrations on consumer TVs ranging from WW2-based CRT tube sets to the latest OLEDs, as well as desktop monitors,  I've come to the conclusion that the picture menu settings matter far more than how many lines or dots of resolution a TV or monitor is capable of.  Not to mention, well-produced content in the first place.
> 
> That's right - I said it: Settings/calibration makes the biggest impact on enjoyment of your display(and are also good for your eyes and for extending the life of the display).
> 
> ...


I agree that calibration of the display is important.  In a previous life I calibrated TV sets on the side during my uni years.  And you're right, most people were underwhelmed with the result but after watching the re-calibrated screen for a couple of weeks (and re-calibrating their brains) they would never go back to the previous settings.

Anyway, the Dell monitors are calibrated from the factory (they even come with a unique print out to show the results) so any additional tweaking would be a marginal improvement at best.


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## TheSonicTruth

old tech said:


> Anyway, the Dell monitors are calibrated from the factory (they even come with a unique print out to show the results) so any additional tweaking would be a marginal improvement at best.



Even though, Dell still does not know what environment these monitors will ultimately be used in.  Dark bedroom, sunny office, all have an impact on appearance of the image.

Brightness and contrast should still be calibrated for local lighting conditions, even if color and grayscale are certified and can usually be left alone.


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## TheSonicTruth

Delete please!  Flubbish internet speed!


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## gregorio (Dec 13, 2019)

Davesrose said:


> [1] I wasn't talking about resolution. Dynamic range is a separate subject.
> [2] With digital imaging, resolution is the number of pixels or dots in a given area. ... There is no such thing as infinite resolution: if you were able to go up close to a billboard (which can be as little as 20dpi), you would start seeing softness before just seeing individual points.
> [3] Also, content creators need larger dynamic range of their source files to be able to pull up detail in shadows or recover blown highlights (especially when converting HDR images to monitor color spaces). Try post processing an image that doesn't have enough DR, and you'll either see noise or black splotches in shadows as well as complete white splotches in highlights.
> [4] I realize there is not a 1:1 comparison of digital sound reproduction to image reproduction. However, it is valid to understand respective technology and see what analogies there are.



It's possible I have misinterpreted your post and am therefore going to inadvertently misrepresent it. If that's the case, I apologise in advance but it's still worthwhile as it goes to the heart of the OP:

1. In the case of digital audio, resolution and dynamic range are effectively exactly the same thing.

2. This really is a fundamental difference between digital imaging and digital audio! With digital imaging (as I understand it) we have a fixed output, an image is recreated using a fixed number pixels which when recreated correspond to (for example) a fixed number of LEDs. The more pixels/LEDs the higher the resolution but obviously we cannot have an infinite number of pixels, as that would require an infinite amount of data, and we cannot have an infinite number of LEDs, as that's a physical impossibility and therefore with digital imaging, as you say, "There's no such thing as infinite resolution", the only question is how many points (pixels/LEDs) we have and under what conditions that number exceeds the capabilities of the human eye. This is completely different to how digital audio works, the analogue signal output reconstructed from digital audio data does not have ANY fixed points, is not reproduced by a finite array of say LEDs and having more fixed data points (pixels) does not have any effect on resolution. Therefore, there IS "such a thing as infinite resolution" in digital audio, in fact, the whole principle of digital audio is based on infinite resolution (Shannon/Nyquist)!

Despite my ignorance of digital imaging, I'll attempt an analogy: Let's say we have a perfect circle which we want to capture and reproduce as a display graphic. We can capture/measure various points on the circumference of our circle as pixel data and then output that pixel data to say LEDs. The more pixels and corresponding LEDs we have, the more accurate (higher resolution) our reproduced circle will be. There is another way though, we can measure just 2 points on the circumference of our circle, which we store as data, mathematically define a perfect circle that bisects these two points and we now have infinite resolution! As I understand it, these two different methods define raster graphics and vector graphics respectively and digital audio is analogous with vector graphics, not raster graphics. This analogy fails in the final step though, because all displays have a fixed/finite number of LEDs, so our vector graphic has to be rasterised accordingly and we're stuck with that finite resolution defined by the number/density of LEDs. While with sound/audio reproduction we do not have LEDs (or any audio equivalent), we can in effect directly output that vector graphic without rasterisation, thereby maintaining infinite resolution! The limiting factor with sound capture and reproduction is therefore not digital audio but the laws of physics pertaining to the analogue input and output signals (EG. Thermal noise and transducer inefficiency).

3. Again, this is a massive difference between digital audio and digital imaging. 32bit float audio post processing (mixing) has been around for 20+ years which theoretically could encode a dynamic range of 1673dB. However, as a sound wave is the compression and rarefaction (variations in pressure) of air molecules, at 194dB the rarefaction portion of the wave would be a total vacuum and as we can't have more than a total vacuum, we can never have a sound wave greater than 194dB, beyond that point we can only have a shock wave. Being even more silly, a shock wave of 1100dB would so massively compress the air molecules that a (5kg mass) black hole would form! In digital audio, our processing environment massively exceeds what can ever actually exist in the real world, let alone what transducers or human senses are capable of.

4. In theory I would agree but in practice, as they're quite different, complex technologies, it's difficult to understand both of them and therefore analogies between them tend to be either invalid or only valid up to a point (and are then invalid again). So even in the latter case, when an analogy might be a useful aid to explaining/understanding a specific aspect of digital audio, it's more than likely to lead to a misunderstanding of other aspects and of digital audio as a whole.


Davesrose said:


> [1] We haven't reached the DR limits of vision with imaging, and
> [1a] there are very different applications where resolution comes into play. A 46 mega pixel camera is over-kill for an image that gets scaled down to say a 1024 web image. But a person may still want that original size for making a high quality print at a large size.
> [2] While imaging still has room for improvement, I would agree that we've reached the limits of fidelity with audio in relation to human perception....and I leave it to others who may want to expose themselves to sound peaks over 100dB or want to argue what processing method (in relation to 1bit to 16bit to 24bit) is "best".


1. This is largely addressed by my point 3 above, we exceeded the DR limits of hearing with digital audio decades ago and 20+ years ago exceeded what can even be reproduced according to the laws of physics.
1a. There's no analogy for this with audio!

2. I would point out that sound peaks and dynamic range (and therefore number of bits) are unrelated. In the real world of music gigs, an audience member might get peak levels of 120dB at an exceptionally loud EDM/rock/pop gig (if they're close to the speakers) but a dynamic range of only 40dB, for which 8 bits or so would be sufficient. While at a large, loud symphony gig, sitting close to the orchestra you might experience 96dB or so but a dynamic range of nearly 60dB, for which 11bits or so would be sufficient. From the perspective of an audience member, there is no real world music circumstance that exceeds (or even comes close to) the dynamic range of which 16bit is capable. In other words, 16bit digital audio already exceeds what actually exists in the real world (of music gigs), regardless of human perception!

G


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## Davesrose (Dec 13, 2019)

gregorio said:


> 1. In the case of digital audio, resolution and dynamic range are effectively exactly the same thing.



Wow, and here I thought DR was about loudness, and resolution about frequency



gregorio said:


> The limiting factor with sound capture and reproduction is therefore not digital audio but the laws of physics pertaining to the analogue input and output signals (EG. Thermal noise and transducer inefficiency).



And even after my basic descriptions, you don't think photography has similar issues as well???  One limiting factor of DR with the camera level is getting the most out of the black point (where there is a cut off with noise floor) vs white point (the value of saturation at exposure).  Different brands have different approaches at setting black point and white point (based on how their own sensor/ADC is at having a SNR and full saturation point).



gregorio said:


> 3. Again, this is a massive difference between digital audio and digital imaging. 32bit float audio post processing (mixing) has been around for 20+ years which theoretically could encode a dynamic range of 1673B.



I'm amazed how much you've written and taken various posts out of context.  Of course an audio file is not the same as an image file: but I would have hoped that you might have read enough of my posts to understand that there are many image formats that have more data than just one R,G,B channel.  A source audio file will have different tracks of audio recordings.  The same can be true for photographs (as in different layers), and video (that can have different tracks of video or audio sources).  A 3D scene file is quite different. The file size itself can actually be pretty small as it comprises vector 3D mesh, procedural shaders, animation graph (that only needs to plot a change in movement), animation scripts (that simulate complex animation at render), and links to images for textures.  An audio source, in comparison, can be larger as it has plots of data for each track based on a set time interval.  That's ironic given 3D graphics can be way more processor intensive in rendering compared to other media.  This even though the workflow with 3D projects is to render "passes" (separate objects and/or color or contrast settings).  The final role in 3D projects is to composite all these passes (and possibly layer video and audio sources).  Forgive me for nerding out, but I am passionate about cinematography.  I learned from ILM that the basis of our current workflow of rendering in passes originates from the original Star Wars films....in which the first use of computers was a prerecorded motion track for cameras.  ILM also funded the pioneers of digital imaging (Photoshop, and quite a few 3D technologies).  By the 80s, ILM would even use this process of filming passes for TV broadcasts of Star Trek Next Generation.  In a given scene in outer space, they would record different passes of the Enterprise model with just internal lights, outer lights, set lights, or other objects in that frame.  Given the requirements and time involved with TV in that era, the source VFX filmed in Vistavision would then be scanned in for analog video editing.  What's great about the situation of the show filming its VFX on film, is that the remastered HD version looks great (CBS spent the money to find and scan all VFX footage to composite in digital HD).

It's also not haphazard that camera manufacturers chose a Bayer sensor pattern ( 1 red, 1 blue, to 2 green) as it corresponds to component video space (where you have double the density of one channel for more efficient resolution and DR).  Lastly, when it comes to 32bpc image formats, those are more specialized.  People capturing images on their iPhone are not concerned with full 32bpc images.  Since most photographs taken today are with default apps on cell phones, most image files are 8bpc jpegs.  It's the photographers who want full DR of a given scene (that merge at least 3 different RAW exposures) and then tone-map to display DR, or it's the 3D artist that is using HDR textures for projecting raytracing (for realistic light simulation).  I've also found that while 32bpc image files take up more space, they are noticeably faster at rendering (so with 3D processing, 32bit float doesn't just mean higher DR, but more efficient rendering).

I'm expanding on information about visual technology for those who are genuinely interested on the subject.  Even consumers who have decided on a switch to HD, to now UHD, probably do have a passing interest in how that visual content is created.


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## bigshot (Dec 13, 2019)

Why wouldn't the depth of the noise floor be related to resolution? I can see someone comparing color depth to bit depth, but I can see bit depth being involved in pixel sharpness too. I think images and sound are apples and oranges personally. However you compare factors involved in resolution, it's going to be a stretch. I guess it just depends on which direction you stretch it.

I know people here love to use analogies, but when it starts getting into microwave ovens and CRT vs OLED and unladen swallows, I tend to glaze over. When you've accomplished the practical reality of perfect sound recording for the purposes of listening to music with human ears, what point is there of going further? I guess it's like nuclear bombs... Is it better to be able to blow up the world 12 times instead of just 3? OH DAMN! I JUST DID IT TOO! STOP ME BEFORE I ANALOGY AGAIN!


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## Davesrose (Dec 13, 2019)

bigshot said:


> Why wouldn't the depth of the noise floor be related to resolution? I can see someone comparing color depth to bit depth, but I can see bit depth being involved in pixel sharpness too. I think images and sound are apples and oranges personally. However you compare factors involved in resolution, it's going to be a stretch. I guess it just depends on which direction you stretch it.
> 
> I know people here love to use analogies, but when it starts getting into microwave ovens and nuclear bombs and CRT vs OLED and unladen swallows, I tend to glaze over. When you've accomplished the practical reality of perfect sound recording for the purposes of listening to music with human ears, what point is there of going further?



Your question seems to be about images.  There is going to be some apples to oranges comparison with image formats vs sound.  There still is some correlation with aspects like file compression introducing artifacts, and factors of file structure based on DR and resolution.  But the fundamental question of DR vs resolution with imaging is that they can be factors for having a high quality image...but are fundamentally different subjects.  With exposure, the optimal quality in DR is one in which you don't see noise in your blackest shadow value, and have a full range of contrast with no blown highlights.  Resolution is about optimizing pixel pitch for your given situation (and file size will also increase with your intended resolution to image size requirements).  A higher quality, large scale image viewed up close is going to require good resolution and DR for the best perceived detail.  The catch for these factors is that an increase in DR and resolution requires more file space, and can be over-kill if your final output is a small resolution jpeg.  One reason why 4K video streaming is becoming a popular format is that there's a newer compression format (h.265) that can reduce file size, but still maintain good quality UHD resolution, Dolby Vision color space, and DD+ Atmos standards for the common broadband speeds now.


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## gregorio

Davesrose said:


> Wow, and here I thought DR was about loudness, and resolution about frequency


Maybe you've learnt something then? DR is not about loudness, as I stated in my previous post. DR in the digital domain is defined by bit depth, due to the resolution of different bit depths defining the digital noise floor, as explained in to OP. Frequency is defined by the sampling rate but provided it is at least two times the audio frequency, resolution is effectively infinite. 


Davesrose said:


> And even after my basic descriptions, you don't think photography has similar issues as well???


Not according to your descriptions! According to your descriptions, even the highest bit depths and UHD video formats can't even represent the full range the human eye is capable of, let alone levels trillions of times greater then it's even possible for a sound wave to exist!

G


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## vatch

16-bit, 44.1 is still just as good for playback if not better as less errors are likely generated with unnecessary calculations..


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## vatch

The real limitation of electronics is about 20-bit anyway, you're not getting anything accurate beyond that as far as dynamic range goes as the noise inherent in electronics and capacitors is at that level; the rest is just noise.


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## bigshot (Dec 13, 2019)

The thing about sound and images that does correlate is that for the purposes of the intended use for the file, there is a point of transparency where more zeros and ones aren't going to do anything but increase file size. That applies to compression too. If your purpose is to look at a picture on your computer monitor, odds are a medium sized JPEG that matches your screen resolution is all you need. You can increase the file size, but at your intended viewing distance with the entire image visible, it won't make a lick of difference. That is true of audio too. If you want to listen to a Mozart piano concerto in your living room, an AAC file at 256 VBR ripped from a CD is all you need.

The problem with audiophiles (and armchair photo nuts) is that they rarely take into account the intended purpose, they have no clue about the thresholds of human perception, and they keep creating "what ifs" in their head to continually push the goalposts back over and over again.

The truth is that the quality of an image lies in its composition, lighting and exposure, not the file size. And the quality of recorded music depends on the musicality and musicianship of the musicians and the balance of the mix. We've gotten to the point where "good enough" is already far into the range of overkill. We should be happy with that, not trying to think up excuses why we might need more.

The part that can still use improvement is the last step in the line... the transducers and the display. Not the files themselves.


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## vatch

That is accurate.


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## Davesrose (Dec 13, 2019)

gregorio said:


> Maybe you've learnt something then? DR is not about loudness, as I stated in my previous post. DR in the digital domain is defined by bit depth, due to the resolution of different bit depths defining the digital noise floor, as explained in to OP. Frequency is defined by the sampling rate but provided it is at least two times the audio frequency, resolution is effectively infinite.



I already knew about this, and it certainly was not based on your previous posts.  Such as your last in which you claim audio DR being relational to one given time instead of full system (where, for example accepted definition of DR for 16bit audio is 95dB).



gregorio said:


> Not according to your descriptions! According to your descriptions, even the highest bit depths and UHD video formats can't even represent the full range the human eye is capable of, let alone levels trillions of times greater then it's even possible for a sound wave to exist!



I should stop responding to you, as you're either being purposely obtuse or still not understanding some fundamentals about photography.  The topic was recorded DR of source.  With digital imaging, we're still limited to realized DR of the conversion of light source to ADC...then also source file to final output.  A captured exposure is limited to acceptable SNR to highest saturation point (which with ADC, also has issues of amplification of signal).  Refer back to previous posts, and you'll see I stated many digital cameras can now record 14bpc at base ISO (some are even better: the best RED cameras actually can record 16bpc).  So how then are there photo formats that are 32bpc?  It' because you can set at least 3 exposures (one "normal", one "over-exposed", one "under-exposed") and then merge them in a single frame.  Because that process requires time, it's incompatible with situations like video (where you do have to expose a frame at certain fps).


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## bigshot

A noise floor of -95dB is never going to be audible under commercially recorded music. It is overkill for any normal purpose of listening I can think of. (and most abnormal ones too!)


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## castleofargh

The audio and video analogies are crap because even if we carefully pick the variables and explain the context for both, those contexts will be different! 
The very way the bits are used is different. In a picture for a given color channel, the minimum and maximum value of however many bits you use, will always mean both extremes of the predefined spectrum! To put it in the most obvious and intuitive way for those who aren't familiar with the stuff, consider a digital B&W picture. There is only that one "color" channel going from pure black to pure white. If you use 8bits to give a pixel its "color", the max value at 255 will be pure white and zero pure black. If you use 1bit encoding, now 1 will be pure white and zero pure black. It's the same for each color channel and also the same idea for the total color spectrum(as each channel works that way, the sum of colors also does). The difference with more bits, lies in how many increment values are available within that predetermined range.

Have fun using that behavior to describe bits in PCM files.  


@Davesrose . I'm sorry but the more stuff you bring up, the messier it looks. 
It started with the 1bit DAC that sounded good to you(and probably does), which has really little to do with 16 or 24bit audio files. When discussing 16 vs 24bit PCM, it is obviously assumed that we're comparing files of similar sample rate. A one bit DAC or most DACs, won't keep the original sample rate. And will use and abuse noise shaping so that the moronic bit value does not lead to a SNR of 6dB. Those stuff are not mysterious in any way, but of course if we keep looking only at a bit value out of context while ignoring the rest, it looks weird or even impossible. And it's not like 32bit DACs are 32bit, so now to add to the confusion about how a DAC will decide to skin the PCM cat, we have marketing entering the fray while the chip will stick to just a handful of bits and run as a typical delta sigma DAC(like your 1bit DAC).
So your anecdote was fine, but also not actually saying anything at all about PCM bit depth. And since, we got camera sensors, TV screens, processing, color channels, HDR... Is there a point to all this? Because to be clear, the argument isn't that we can't pull some analogies from the video world. Of course we can. It's that most of the time we shouldn't to avoid misunderstandings. IMO that series of posts is a solid case supporting that we should pick such anecdotes with more care.


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## Davesrose

castleofargh said:


> The audio and video analogies are crap because even if we carefully pick the variables and explain the context for both, those contexts will be different!
> The very way the bits are used is different. In a picture for a given color channel, the minimum and maximum value of however many bits you use, will always mean both extremes of the predefined spectrum! To put it in the most obvious and intuitive way for those who aren't familiar with the stuff, consider a digital B&W picture. There is only that one "color" channel going from pure black to pure white. If you use 8bits to give a pixel its "color", the max value at 255 will be pure white and zero pure black. If you use 1bit encoding, now 1 will be pure white and zero pure black. It's the same for each color channel and also the same idea for the total color spectrum(as each channel works that way, the sum of colors also does). The difference with more bits, lies in how many increment values are available within that predetermined range.
> 
> Have fun using that behavior to describe bits in PCM files.
> ...



What started the latest round of picture analogies is my response to post number 1 on this very thread: where Gregorio claimed it's easy to see bit depth with images.  My preface for that was it might be common for older people who had experience with color spaces that were below 256 colors.  From what I could tell (granted I haven't read every page), this analogy was never fully addressed.  With my engagement with Gregorio, I can tell he hasn't considered image capture and current photo and video formats...so while the topic isn't specifically sound related, I think other members might find my posts informative for the latest technologies in photography and video.  If you want the thread to continue with *this format* or *this mastering process* is best, then I'll let it be


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## bigshot

Well, it's a good time to start comparing bit depth to sports cars and wine then!


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## old tech

gregorio said:


> Therefore, there IS "such a thing as infinite resolution" in digital audio, in fact, the whole principle of digital audio is based on infinite resolution (Shannon/Nyquist)!


Now I am confused.  As you point out, resolution is effectively dynamic range - which in turn is directly related to SNR.  So isn't resolution in digital audio a fixed value directly related to bit depth (which can be perceptually increased with dither noise shaping)?

In other words, if a digital file had infinite resolution would it not by definition also have an infinite dynamic range and an infinite SNR?


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## TheSonicTruth (Dec 13, 2019)

old tech said:


> Now I am confused.  As you point out, resolution is effectively dynamic range
> - which in turn is directly related to SNR.  So isn't resolution in digital audio a
> fixed value directly related to bit depth (which can be perceptually increased
> with dither noise shaping)?
> ...




Confused?

Two words: _Ethan. Winer._

Cuts to the chase, explains everything in simple terms, and won't try to tell you the sky is green and grass is blue, like a couple of guys on here do...!


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## bigshot (Dec 13, 2019)

Within the range of the frequency response and the noise floor, dictated by sampling rate and bit depth, according to Nyquist sound is recreated *perfectly*. There isn't "more resolution" to be had. It's perfect. All you can achieve by increasing the sampling rate and bit depth is a wider range of frequencies and a deeper noise floor which are also recreated *perfectly*.

So therefore, if the intended recipient of the sound is a human, 16/44.1 reproduces the sound perfectly. You can increase the sampling rate and bit rate, but it doesn't add any resolution to the sound. All it adds is information that exists outside of the range of hearing. In order to effectively add resolution, you have to be able to perceive it.

The word "sound" assumes that it is possible to hear. Beyond the range of hearing it's just generalized "data".


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## Davesrose (Dec 13, 2019)

Instead of "infinite resolution", the Nyquist-Shannon sampling theorem refers to finite resolution.  Gregorio also tried to tell me that audio sampling is like vector graphics, which it's not.  With vector graphics, you only have a point in XY (or Z)  space when there's a change in angle.  With audio sampling, it's based on time intervals.


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## old tech

bigshot said:


> Within the range of the frequency response and the noise floor, dictated by sampling rate and bit depth, according to Nyquist sound is recreated *perfectly*. There isn't "more resolution" to be had. It's perfect. All you can achieve by increasing the sampling rate and bit depth is a wider range of frequencies and a deeper noise floor which are also recreated *perfectly*.
> 
> So therefore, if the intended recipient of the sound is a human, 16/44.1 reproduces the sound perfectly. You can increase the sampling rate and bit rate, but it doesn't add any resolution to the sound. All it adds is information that exists outside of the range of hearing. In order to effectively add resolution, you have to be able to perceive it.
> 
> The word "sound" assumes that it is possible to hear. Beyond the range of hearing it's just generalized "data".


Yes but it is not infinite resolution...  I agree that any more resolution than 16/44 is not going to be perceptible to humans and also beyond the range of most recorded music, I just cannot get my head around resolution being infinite as that would measure as an infinite SNR/dynamic range and would imply and infinite bit depth and perfect implementation.  I appreciate that this is academic as infinite resolution does not exist even in the natural world, I'm just trying to fill gaps in my (laymanish) understanding of digital audio.

Funnily enough, there are quite a few analog types (particularly vinylphiles) that believe records have infinite resolution, yet they cannot provide a convincing reason why then does its SNR and dynamic range is substantially less than CD.  Not helped of course by misinformation sites like howstuffworks claiming the same nonsense.


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## bigshot (Dec 14, 2019)

How do you get more infinite than perfect? This may be a theological question!

the point is that within the range, it can’t have any more resolution because it’s perfect.


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## old tech

bigshot said:


> How do you get more infinite than perfect? This may be a theological question!
> 
> the point is that within the range, it can’t have any more resolution because it’s perfect.


I think you are missing the point. It is not a theological question either. Forget about the bounds of human perception, the question is how is it possible to have infinite resolution without an infinite SNR and dynamic range?


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## gregorio

Davesrose said:


> [1] What started the latest round of picture analogies is my response to post number 1 on this very thread: where Gregorio claimed it's easy to see bit depth with images. My preface for that was it might be common for older people who had experience with color spaces that were below 256 colors.
> [2] With my engagement with Gregorio, I can tell he hasn't considered image capture and current photo and video formats...
> [3] so while the topic isn't specifically sound related, ...



1. You're effectively arguing against yourself here, or rather, effectively arguing that you didn't really pay attention to the OP before you decided to dispute it. Sure, it's considerably more difficult today to see the difference between different bit depth/resolution images but the OP was written nearly 12 years ago, when consumer digital photography, video formats and online digital image formats were significantly lower resolution than today! However, it is still possible today to tell the difference in some typical consumer viewing situations, while with 16bit vs 24bit digital audio, there were no typical consumer listening situations where the difference could be differentiated, either when 24bit was first introduced to consumers or at any time since!

2. Again, what you "can tell" - is erroneous. I'm certainly no expert but I certainly HAVE considered current video formats in particular, as my job requires a certain level of familiarity.

3. Exactly and that's the problem! You seem to believe that digital imaging is more "sound related" than is actually the case and are therefore using analogies which are as misleading as they are analogous.


Davesrose said:


> [1] Instead of "infinite resolution", the Nyquist-Shannon sampling theorem refers to finite resolution.
> [2] Gregorio also tried to tell me that audio sampling is like vector graphics, which it's not.
> [2a] With vector graphics, you only have a point in XY (or Z) space when there's a change in angle. With audio sampling, it's based on time intervals.



1. This statement is false, the sampling theorem refers to both finite and infinite resolution. The concept of infinite resolution was described in Shannon's seminal 1948 paper "A Mathematical Theory of Communication" (upon which the Nyquist/Shannon sampling theorem is predicated), which states: "_If a function of time is limited to the band from 0 to W cycles per second it is completely determined by giving it's ordinates at a series of discrete points spaced 1/2W seconds apart_" and in his article of the same year ("Communication in the presence of noise") Shannon provides: "_A mathematical proof showing that this is not only approximately, but exactly, true  ..._" - What do you think "completely determined" means, if not infinite resolution?

2. Again, your statement is false! What I actually tried to tell you is that while there are some similarities, the analogy between vector graphics and audio sampling ultimately fails (is invalid)!! Following on ...
2a. That's not entirely true, which is why there is SOME validity to my analogy. The statement quoted above (from "A mathematical theory of communication") was not innovative, it was effectively stated by Harry Nyquist some 20 years earlier, what was innovative was Shannon's mathematical proof of the statement and various other aspects of implementing digital audio, one of the most important of which was approaching the issue and formulating a solution in terms of Euclidean geometry (quote: "_Geometrical representation of the signals_"). IE. The representation of a signal as a point in an N-dimensional Euclidean space. In fact, Shannon's summary of his article states "_A method is developed for representing any communication system geometrically. Messages and the corresponding signals are points in two "function spaces," and the modulation process is a mapping of one space into the other._". 



old tech said:


> Yes but it is not infinite resolution... I agree that any more resolution than 16/44 is not going to be perceptible to humans and also beyond the range of most recorded music, I just cannot get my head around resolution being infinite as that would measure as an infinite SNR/dynamic range and would imply and infinite bit depth and perfect implementation. I appreciate that this is academic as infinite resolution does not exist even in the natural world, I'm just trying to fill gaps in my (laymanish) understanding of digital audio.



That "gap" is tricky to fill because it requires more than a "laymanish" understanding of digital audio. We experience acoustic sound and can relate that experience to analogue audio precisely because it's analogous. However, digital audio is not analogous to acoustic sound (or analogue audio) and therefore requires a conceptualisation (thinking about it) that is different. This requires going beyond "layman", and is somewhat difficult because it's not especially intuitive. It is possible to explain in terms that are "beyond layman" but still "laymanish", but bare in mind that such an explanation is rather like an analogy that is only somewhat accurate:

With acoustic sound we have sound waves, a signal that includes a noise floor. In the case of a music performance, that noise floor is created by the breathing and movement of both the musicians and the audience, extraneous sound entering the venue and one or two other variables. Analogue audio is also a signal that includes a noise floor, given a theoretically ideal/perfect analogue device that noise floor is defined by thermal noise (the collision of electrons within electrical circuits). A way to conceptualise digital audio is: A system that, unlike acoustic or analogue signals, does NOT include any noise floor at all (IE. Has infinite SNR). What it does include, that acoustic sound and analogue audio do not, is quantisation error, a type of signal distortion that provides a limit to "resolution". However, this quantisation error can be completely eliminated by a process intrinsic to all digital audio, dither. So now we have a perfect signal, with no distortion and unlimited resolution. Unfortunately though, as far as what we're going to hear after the DA conversion is concerned, the result of this dither process is white noise. So we seem to be back where we started, a dynamic range/resolution that is NOT infinite, that is limited by an included noise floor (dither noise). However, that's not really the case because dither noise is not an intrinsically "included noise floor", it's effectively separate. This might appear to be a purely semantic statement, as we can't remove this dither noise (without also loosing the elimination of quantisation error) but it's not just semantics, it's fundamental to how digital audio works. In other words, what we effectively have with digital audio is a signal with infinite resolution but you can't hear all of it because it's obscured by dither noise. It's important to conceptualise digital audio as two entirely separate but overlaid signals, an infinite resolution signal and a dither noise signal, because although we can't get rid of dither noise, we can process it entirely independently (of the infinite resolution signal). The demonstration (and practical implementation) of this fact is noise shaped dither. 16bit digital audio has an theoretical max resolution/dynamic range limit of ~96db (6.02db x 16bits) but using a typical noise shaped dither algorithm, to process/move the dither noise independently, we can extend this dynamic range/resolution to ~120dB (in the critical hearing band). Question: How is this possible, that extra 24dB of dynamic range (4 bits of resolution) can't exist in 16bit digital audio, where has it come from? The answer is: It was always there, the resolution is effectively infinite at virtually any bit depth but you'll have to eliminate the quantisation error and move the dither noise which obscures it. Incidentally, the most aggressive noise shaped algorithms I've used, allow a dynamic range/resolution of ~150dB with 16bit. In practice of course this can't actually be realised (unless you deliberately screw-up the gain staging big time, specifically to create a signal to test for it), because at -120dB below peak (and even at ~96dB), the noise floor of the analogue signal chain is higher and the noise floor of the original acoustic signal is higher still.

Not sure if this helps?

G


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## Davesrose (Dec 14, 2019)

gregorio said:


> 1. You're effectively arguing against yourself here, or rather, effectively arguing that you didn't really pay attention to the OP before you decided to dispute it. Sure, it's considerably more difficult today to see the difference between different bit depth/resolution images but the OP was written nearly 12 years ago, when consumer digital photography, video formats and online digital image formats were significantly lower resolution than today! However, it is still possible today to tell the difference in some typical consumer viewing situations, while with 16bit vs 24bit digital audio, there were no typical consumer listening situations where the difference could be differentiated, either when 24bit was first introduced to consumers or at any time since!
> 
> 2. Again, what you "can tell" - is erroneous. I'm certainly no expert but I certainly HAVE considered current video formats in particular, as my job requires a certain level of familiarity.
> 
> ...



How can one claim to need to know a passing knowledge of photography for their line of work, and then demonstrate lack of knowledge of what dynamic range is (in relation to photography), or what standards there are for video formats?  Especially 12 years ago, when 16 million colors where so common place, and cinema standards were moving to higher resolution and DR.

As for resolution of sound, why are you ignoring your quote, which clearly states it's a series of points spaced at a time apart (where that means a finite increment)?  There may be different approaches for interpolation and improving sound....but sampling refers to to specified intervals.


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## SoundAndMotion (Dec 14, 2019)

@gregorio , do you know the limits of your knowledge? Do you know that some of what you write above is absolutely true and some is absolutely nonsense? That is why I mentioned Dunning Kruger a little while back. I can't get into it too much at this moment, but I'll get back to you.
Quick summary: 1. Your quote from Shannon does not imply the ridiculous concept of infinite resolution in digital audio.
2. Noise shaping in digital audio requires oversampling, that is, you must have more samples than Shannon describes.
3. Euclidean space and function space are not the same thing. Do you know what a function space is?

I'll help you out to understand this (or explain to others who are misled by what you say) when I get a chance, soon.


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## SoundAndMotion (Dec 14, 2019)

@bigshot , is there a link for the BigShot Dictionary, because your definitions of "perfect" and "sound" don't match any I know of.
An original signal to which you have added noise due to a finite resolution cannot be recreated perfectly. You have added noise!!! No longer "perfect"!
"Sound" is not limited to what you can hear, e.g. ultrasound is sound and you can't hear it.


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## bigshot (Dec 14, 2019)

I normally don’t get into dumb arguments over definitions, but the term “ultra” in Latin means “beyond”. Whatever the definition you want to use, ultrasonic stuff isn’t what you listen to when you play your stereo. It’s as useless as teats on a bull hog. You’re better off without it. And if it’s there and it isn’t causing you trouble, you can safely ignore it. It sure isn’t worth arguing about.



old tech said:


> I think you are missing the point. It is not a theological question either. Forget about the bounds of human perception, the question is how is it possible to have infinite resolution without an infinite SNR and dynamic range?



There is infinite inwards... Within the parameters of frequency response and noise floor dictated by Nyquist it is infinite resolution. You can't "blow it up and see the halftone dots". In this sense, it is like vector graphics. It doesn't matter what scale you look at it, within the boundaries of Nyquist, it is perfect.

Infinite outwards is a different story... if I say that salt shaker is perfect, it doesn't mean the pepper shaker is perfect, or the table it's sitting on, or the room, or the house, or the city... Infinite outwards is when theology comes in... "Can God create a rock so big even He can't pick it up?" Infinite outwards is a rabbit hole that audiophiles love to chase down... How perfect is perfect? How much is enough?

Snatch the pebble from my hand, Grasshopper.


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## TheSonicTruth

bigshot said:


> Infinite outwards is a different story... if I say that salt shaker is perfect, it
> doesn't mean the pepper shaker is perfect, or the table it's sitting on, or the room,
> or the house, or the city... Infinite outwards is when theology comes in... "Can God
> create a rock so big even He can't pick it up?" Infinite outwards is a rabbit
> ...



Y'know, I've begun to wonder exactly what 'bigshot' looks like .....

Now I have a clearer idea:  !


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## bigshot

I look like a little dog in a top hat and tails!


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## old tech (Dec 14, 2019)

bigshot said:


> There is infinite inwards... Within the parameters of frequency response and noise floor dictated by Nyquist it is infinite resolution. You can't "blow it up and see the halftone dots". In this sense, it is like vector graphics. It doesn't matter what scale you look at it, within the boundaries of Nyquist, it is perfect.


I can't be infinite backwards if SNR/DR doesn't measure as infinity. 

No one is suggesting that you can "blow it up and see half tone dots", but rather that there is always quantisation errors or implemented dithered noise which put a limit on resolution.  If I broadly understood @gregorio 's post, then the "infinite resolution" part cannot in practice exist separately to the noise and/or distortion part.  Intuitively it makes sense and goes a long way to explain why increasing the bit depth or dithering and shaping the noise to outside the range where human hearing is most sensitive will increase resolution (and hence, increase SNR/DR), but infinite resolution? - I would be keen to see some real world measurements of digital audio no matter how much the bit depth is or how well it is noise shaped, that show SNR/DR approaching infinity.


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## old tech (Dec 14, 2019)

gregorio said:


> That "gap" is tricky to fill because it requires more than a "laymanish" understanding of digital audio. We experience acoustic sound and can relate that experience to analogue audio precisely because it's analogous. However, digital audio is not analogous to acoustic sound (or analogue audio) and therefore requires a conceptualisation (thinking about it) that is different. This requires going beyond "layman", and is somewhat difficult because it's not especially intuitive. It is possible to explain in terms that are "beyond layman" but still "laymanish", but bare in mind that such an explanation is rather like an analogy that is only somewhat accurate:
> Not sure if this helps?
> G


Thanks for the explanation, it does somewhat expand my "laymanish" understanding of audio, though I did already understand the concept behind digital audio.  I appreciate that a key difference between analog and digital signals is that noise and error is intrinsic to an analog signal while it exists somewhat separately with digital. Dealing with noise in analog production, eg pre/de emphasis (like Dolby) will increase its resolution, just like noise shaping can increase resolution of a digital production.  In that sense they are similar, though analog will always be at a disadvantage because dealing with the noise also messes with the signal.


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## castleofargh

old tech said:


> I can't be infinite backwards if SNR/DR doesn't measure as infinity.
> 
> No one is suggesting that you can "blow it up and see half tone dots", but rather that there is always quantisation errors or implemented dithered noise which put a limit on resolution.  If I broadly understood @gregorio 's post, then the "infinite resolution" part cannot in practice exist separately to the noise and/or distortion part.  Intuitively it makes sense and goes a long way to explain why increasing the bit depth or dithering and shaping the noise to outside the range where human hearing is most sensitive will increase resolution (and hence, increase SNR/DR), but infinite resolution? - I would be keen to see some real world measurements of digital audio no matter how much the bit depth is or how well it is noise shaped, that show SNR/DR approaching infinity.


It's just a different way to look at the same phenomenon, of course you never end up with anything perfect IRL. But thinking of a sampled signal as the perfect original(when within Nyquist theorem's conditions!!!!!!!) plus whatever amount of noise added by quantization error, is accurate. It's also a model that makes a few digital behaviors more intuitive IMO.

About your question, if you count on real life examples, the S9038PRO sabre chip brags about a dynamic range of 140dB in mono on a full moon with the wind in their back while buried 2kilometers underground. And they give a more realistic THD+N at -122dB. That's nominal conditions and only the chip itself, most complete DACs will do worst.
So your request rapidly falls back to pure math simulation. And then, so long as we're within Nyquist theorem's conditions, we already know that bringing quantization errors near zero will also bring near infinite resolution. I mean, the quantization noise is literally the difference between the perfect sample and the quantized approximation.


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## old tech

castleofargh said:


> It's just a different way to look at the same phenomenon, of course you never end up with anything perfect IRL. But thinking of a sampled signal as the perfect original(when within Nyquist theorem's conditions!!!!!!!) plus whatever amount of noise added by quantization error, is accurate. It's also a model that makes a few digital behaviors more intuitive IMO.
> 
> About your question, if you count on real life examples, the S9038PRO sabre chip brags about a dynamic range of 140dB in mono on a full moon with the wind in their back while buried 2kilometers underground. And they give a more realistic THD+N at -122dB. That's nominal conditions and only the chip itself, most complete DACs will do worst.
> So your request rapidly falls back to pure math simulation. And then, so long as we're within Nyquist theorem's conditions, we already know that bringing quantization errors near zero will also bring near infinite resolution. I mean, the quantization noise is literally the difference between the perfect sample and the quantized approximation.


Cheers for that.

That is what I sort of had understood, ie assuming Nyquist's conditions hold, we would have infinite resolution (or close to it) if there were no quantisation errors.  I certainly agree that resolution can be practically perfect within sample rate and bit depth parameters, particularly in regard with human end use, but not infinite.  Your example of the Sabre chip "only" being able to achieve a DR of 140db under the most contrived conditions illustrates the point.  Now 140db is spectacular DR and way beyond any relevance to reproducing music but even so, it is nowhere near infinity.


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## castleofargh

A file can be sold claiming a resolution of 24/96 when the recording conditions never reach that, and the playback chain is never able to come close to output that. So maybe we can just as well decide that a sampled signal is perfect, and ignore quantization noise as merely another extra stuff we ignore along the way so we can continue to discuss fantasy HiFi XL.


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## old tech

castleofargh said:


> A file can be sold claiming a resolution of 24/96 when the recording conditions never reach that, and the playback chain is never able to come close to output that. So maybe we can just as well decide that a sampled signal is perfect, and ignore quantization noise as merely another extra stuff we ignore along the way so we can continue to discuss fantasy HiFi XL.


I agree. My interest is in the science behind it, not some audiophile belief that resolution beyond some amount matters.

It is also being pendantic. Us claiming that digital audio production has infinite resolution is almost as bad as those analog guys making that claim.  I say almost as bad because the resolution of analog production doesn't exceed the equivalent of 14 bits digital at best, let alone a fantasy that it is infinite.


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## gregorio (Dec 15, 2019)

old tech said:


> [1] Dealing with noise in analog production, eg pre/de emphasis (like Dolby) will increase its resolution, just like noise shaping can increase resolution of a digital production. In that sense they are similar,
> [1a] though analog will always be at a disadvantage because dealing with the noise also messes with the signal.
> [2] That is what I sort of had understood, ie assuming Nyquist's conditions hold, we would have infinite resolution (or close to it) if there were no quantisation errors.
> [3] Your example of the Sabre chip "only" being able to achieve a DR of 140db under the most contrived conditions illustrates the point. ...
> [3a] Now 140db is spectacular DR and way beyond any relevance to reproducing music but even so, it is nowhere near infinity.



1. No, they're not really similar at all. Pre/de emphasis does NOT increase the resolution of the signal and doesn't attempt to, what it does is decrease the amount of noise that will be added to the signal downstream (EG. That occurs from transmitting the analogue signal). If we take a theoretically perfect implementation of pre-de emphasis, what you would end up with is slightly lower resolution, IE. Exactly the same signal/resolution with some additional thermal noise. In contrast, dither linearises all the quantisation errors, effectively giving us perfect accuracy (infinite resolution) and then it's a case of moving the resultant dither noise to reveal that resolution (with noise-shaping).
1a. Other forms of noise reduction for acoustic/analogue do, as you say, "mess with the signal", IE. Introduce distortion, and the more noise you try to remove, the more distortion you introduce. Introducing distortion is not increasing resolution, it's reducing resolution. The trick of applying noise reduction is to reduce the amount of noise somewhat (typically by no more than a few dB), without the added distortion becoming audible/objectionable.

2. And that's why dither is an intrinsic part of digital audio, because it results in "no quantisation errors" and therefore effectively "infinite resolution" (but with dither noise).

3. Unfortunately, that figure doesn't really tell us much. Where it's measured/calculated makes a big difference because a DAC (or ADC) chip is by definition partly an analogue device and therefore subject to the physics of analogue signals (EG. At least some thermal noise) but of course now we're effectively talking about analogue, not digital.
3a. Yes, in practice we cannot achieve infinity resolution or even close to infinity because we always have to enter the analogue realm (and then the acoustic realm).


old tech said:


> Us claiming that digital audio production has infinite resolution is almost as bad as those analog guys making that claim.



Not really, although we do have to be careful about what we mean, for example that we are in fact talking about digital audio and not about analogue audio, EG. Not about ADCs or DACs, which are partly analogue devices. And obviously, we're not talking about a digital audio system, which of course isn't actually a digital audio system, it's an analogue/acoustic system with some digital components/processes, which is therefore constrained by the limits of analogue circuitry and the laws of motion in creating an acoustic signal. And even when talking specifically about digital audio, there are still conditions to such a claim, for example, infinite resolution within a limited/specified audio frequency bandwidth. Furthermore, at the dynamic ranges and bit depths employed in digital audio, we run into the problem of what audio frequencies actually are, and therefore exactly what is meant by an infinite resolution of them. For example, a sound wave is a pressure wave travelling/propagating through a medium (air in our case), IE. A sound wave is the compression and rarefaction (movement) of air molecules, but what happens when we have a massive dynamic range? In the case of say 24bit (or 16bit with aggressive noise-shaping) the amount of energy represented at the bottom of the dynamic range (EG. -144dB) cannot compress and rarefy enough air molecules by a sufficient amount to propagate a sound wave and therefore a sound wave doesn't exist. In other words, a sound wave is itself a finite entity and therefore can't have infinite resolution. To rephrase your statement, we could say that digital audio production has a resolution/DR which exceeds the DR possible for a sound wave to actually exist and therefore effectively has infinite resolution. This is in stark contrast to analogue guys making that claim because in the case of an analogue audio production, we have numerous analogue units/signal paths each of which introduce cumulative distortion/noise, which brings the effective dynamic range of an analogue music production down to about 80dB at best (but more likely <60dB) and -80dB represents way more energy than is needed to propagate a sound wave. Digital audio also has numerous processes, each of which also introduces cumulative distortion/noise but professional audio production environments are 64bit float, so each process is adding noise/distortion at around the -350dB level and even a thousand or more still would not add up to enough for a sound wave to exist (and would therefore still effectively be infinite resolution).

G


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## old tech

@gregorio thanks for the explanation.  Thinking of resolution as being infinite in terms of reproducing a sine wave is a good way of looking at it.  What did my head in was that infinite resolution doesn't exist in the natural world, ie even the propagation of air molecules in the atmosphere will result in some noise and distortion along the way. so how could it in digital production when there is quantisation errors or alternatively, dithered noise that obscures the true signal (however below human thresholds that may be)? 

I'm a bit surprised though that noise reduction cannot increase resolution (whether it be analog or digital processes). I thought that it was a trade-off, ie if the denominator (N) could be reduced by a greater amount than the reduction effect in the numerator (S) then, assuming distortion is kept at bay, by definition SNR must increase.  I recall SNR specs on end tape recorders usually quoting a higher SNR if NR is used (like Dolby A or DBX etc ).  I forget the brand, but a final reiteration of a well known studio tape recorder claimed to achieve nearly 90db with NR.


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## bigshot (Dec 15, 2019)

Noise reduction just reduces noise. It doesn't extend signal down below the noise floor.

I'm not going to consider any sound reproduction format to be infinitely perfect unless it covers the range of 100GHz to 10THz, the point where sound becomes light.

Now that we've answered this one, can we get an answer to how many angels can dance on the head of a pin?


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## TheSonicTruth

bigshot said:


> Noise reduction just reduces noise. It doesn't extend signal down below the
> noise floor.
> 
> I'm not going to consider any sound reproduction format to be infinitely perfect unless it covers the
> ...




I hereby declare you well enough to be relocated from Acutes back to Chronics in Nurse Ratched's ward.  Congratulations!


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## Arjey

So I have a question. 24 bit is useless for listening to music. Does that make DSD useless too? Because I could swear I can hear a difference..
Or is there a difference, because DSD isn't just more bits? What's the difference between DSD and other formats?


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## Hifiearspeakers

Arjey said:


> So I have a question. 24 bit is useless for listening to music. Does that make DSD useless too? Because I could swear I can hear a difference..



Let me correct your first question. “So I have a question. 24 bit is useless for listening to music, in the opinion of some”


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## bigshot (Dec 15, 2019)

On SACDs the DSD track is generally at a higher volume level than the redbook layer. That skews comparison tests if you don't do careful level matching. Even worse, there are quite a few SACDs that have different masterings on the two layers, and I even found one that had a completely different mix. It is very difficult to do a controlled test between SACD and CD. I struggled for a month to find an SACD that was a fair comparison. I finally found one with a DSD recording that had the exact same master on the redbook layer. A sound engineer friend of mine and I racked them up side by side and couldn't detect any difference between them at all.

If someone has a report of a well conducted blind listening test that found otherwise, I would like to hear about it. But I doubt any such thing exists.

If you'd like to set up a controlled test to find out for yourself, I would be happy to help you with advice and tips on how to do that.


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## Arjey

Just an unrelated to the topic question. Is there a way to remove noise from old records? I know about the feature in audacity and other apps, but it doesn't really work.. I like some old jazz, but it has so much noise, it's basically impossible to listen to


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## old tech

Arjey said:


> So I have a question. 24 bit is useless for listening to music. Does that make DSD useless too? Because I could swear I can hear a difference..
> Or is there a difference, because DSD isn't just more bits? What's the difference between DSD and other formats?


As I understand it, DSD is only 1 bit but sampled in the megahertz range to create a wide enough band to shape all that 1 bit noise into (this is an oversimplification...)

I don't doubt that you hear a difference between DSD and other formats, I often do as well.  

The question is what is causing the difference? Typically the main reasons are that different or higher quality masters were used for the DSD version or just plain expectation biases - ie you expect the DSD version to sound better so your brain is fooled into perceiving the sound as better.  Then there are a myriad of other reasons such as not level matching the listening samples, not testing blind and so on.

All I know is that after more than 30 years that DSD and other hi res formats have been available to the public, why are we still debating this while, for example, no-one really debates whether LP records sound better than 8 Track cartridges?  Surely the jury would have been back long ago, yet there are no controlled tests that convincingly demonstrate that DSD or other hi res formats sound different to 16/44 and many studies supporting the claim that like 16/44, the hi res formats are equally transparent to the human ear (such as in the link below).

http://drewdaniels.com/audible.pdf


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## old tech (Dec 15, 2019)

bigshot said:


> Noise reduction just reduces noise. It doesn't extend signal down below the noise floor.



No one is saying that it does, rather it reduces the noise.  So you don't accept that the signal to noise ratio is a ratio?


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## TheSonicTruth (Dec 15, 2019)

old tech said:


> As I understand it, DSD is only 1 bit but sampled in the megahertz range to create a wide
> 
> enough band to shape all that 1 bit noise into (this is an oversimplification...)
> 
> ...




I think what needs to be done is to collect a number of subject listeners, a dozen, maybe fifty, and have them listen to several ten-second snippets of the same piece of music, but telling them NOTHING regarding delivery format, playback equipment being used, or even the composer, performer, or title of the piece.  If conducted on sets of headphones instead of speakers, that might be the only brand name the subjects are exposed to.

The proctors/test officiators know that snippet A is a 256kbps MP3, B is a 256kbps AAC, C is a 16/44.1k WAV, and D is a 24/96k WAV.  But the test listener subjects are going in CLUELESS in this regard.

Have them all write down which snippet they think sounds 'best', or at least different among them.  IE A, B, C, D, or, No Difference.


Then, have a laugh at the results.  I'm predicting equal numbers of subjects pick each of the samples, along with a few votes for 'No Difference'.  That's what happens when all EXPECTATIONS, as in 'expectation bias', are removed by not telling the subjects ANYTHING, except when and where to report for the listening test.


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## gregorio

old tech said:


> I'm a bit surprised though that noise reduction cannot increase resolution (whether it be analog or digital processes). I thought that it was a trade-off, ie if the denominator (N) could be reduced by a greater amount than the reduction effect in the numerator (S) then, assuming distortion is kept at bay, by definition SNR must increase. I recall SNR specs on end tape recorders usually quoting a higher SNR if NR is used (like Dolby A or DBX etc ). I forget the brand, but a final reiteration of a well known studio tape recorder claimed to achieve nearly 90db with NR.



Although operationally different, noise reduction like Dolby A or DBX is conceptually the same as pre/de emphasis and the RIAA curve used on LPs (which is a form of pre/de emphasis). Pre/De emphasis is an EQ curve applied to the master destined for LP (or some other media) and then an inverse of that curve is applied on playback, while Dolby NR and DBX are more sophisticated, based on companding rather than just EQ. Within the industry we often do not refer to this as noise reduction (NR) but as noise suppression, meaning that we're not reducing the amount of noise or distortion in the signal itself, we're just suppressing the amount of noise that will be added by some downstream two way process (such as cutting and playing an LP or recording to and playing back from magnetic tape). In other words, noise suppression doesn't increase resolution, it decreases resolution but not by as much as would be the case if you didn't employ it. Noise suppression effectively died with digital audio because, for example, copying a redbook master to a CD and then reading that data back from the CD should provide a bit perfect copy, IE. No noise is added by the two way process, so there's no additional noise to suppress and all you'd get is the lower resolution of the suppression process itself. The same is effectively true of processing/mixing digital audio, we use a much higher bit depth for processing/mixing so that the added noise (or quantisation error) is always well below the DR/resolution of the source recordings and destination media.



Arjey said:


> So I have a question. 24 bit is useless for listening to music. Does that make DSD useless too? Because I could swear I can hear a difference..
> Or is there a difference, because DSD isn't just more bits? What's the difference between DSD and other formats?



In practice, there really isn't much of a difference between DSD and other formats. DSD is just Sony's name for the delta-sigma method of digital audio, where a high sample rate (in the megahertz range) is used and then one or just a few data bits are employed to encode the difference in the analogue waveform (between one sampling time period and the next). The PCM method of digital audio doesn't encode the difference between the waveform at different instants of time, it encodes an absolute value of the waveform at each sampling instant, it therefore requires many more bits of data than delta-sigma but lower sample rates. However in practice, all professional PCM ADCs (for the last 25+ years or so) also employ very high sample rate (in the megahertz range) delta-sigma encoding using just a few bits to convert the analogue signal to digital data, so a PCM ADC and a DSD ADC are effectively the same, except that once converted to digital audio, a PCM ADC converts the delta-sigma data into PCM data (a process called "decimation") while a DSD ADC doesn't, it leaves it as delta-sigma data and applies noise-shaped dither. Also, it's not really possible to mix/process DSD recordings, so the vast majority of SACDs have been converted to PCM (typically 24/96 or 24/192), mixed/processed, then converted back to DSD and copied to an SACD. And lastly, virtually all PCM DACs do the reverse of PCM ADCs and convert the PCM data to delta-sigma for conversion into analogue. In practice then, there really isn't any/much of a difference, only where and when the conversions between delta-sigma and PCM occur.

It was a bit of marketing genius on Sony's part though: A DSD ADC doesn't have a decimation section, a DSD DAC doesn't have to perform the reverse conversion and therefore they're simpler devices and cheaper to manufacture. However, Sony did not pass along this cheaper cost of ADC and DAC manufacture, in fact they did the exact opposite and actually priced them higher, which they justified by marketing DSD as higher fidelity/resolution (that should therefore be more expensive). In a sense this became a self-fulfilling prophesy: DSD DACs (and SACDs) were not portable and because they were relatively expensive, they were only purchased by those who took sound quality seriously and had the money to own sound systems that were significantly better than average. This gave the record labels and audio engineers the option of creating a DSD master specifically targeted at "significantly better than average" sound systems, rather than targeted at a range of sound systems (from relatively poor sound systems to significantly better than average) as is the case with CD. In other words, if the record label has exercised the option of creating two different masters (targeting different quality consumer reproduction systems), then indeed you should "hear a difference". However, this is nothing directly to do with bits or formats, because a DSD master (specifically targeted at better than average systems) could be distributed on a standard CD transparently (IE. With no loss of perceivable fidelity).



Hifiearspeakers said:


> Let me correct your first question. “So I have a question. 24 bit is useless for listening to music, in the opinion of some”



You haven't corrected the assertion though, you've effective done the opposite! Factually, a 24bit file format IS useless for listening to music, although some have an opinion which contradicts this fact.

G


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## bigshot (Dec 16, 2019)

Arjey said:


> Just an unrelated to the topic question. Is there a way to remove noise from old records? I know about the feature in audacity and other apps, but it doesn't really work.. I like some old jazz, but it has so much noise, it's basically impossible to listen to



There are two types of noise reduction... Impulse Noise Reduction, which makes pinpoint edits to remove quick clicks and pops, replacing them with pink noise; and broadband noise reduction, which reduces continual noise. Impulse Noise Reduction can be easily applied without a lot of fuss, but broadband requires a careful adjustment to avoid lopping off music with the noise. There is no one-size-fits-all magic box that just fixes it real time. You have to capture the LP to a digital file and apply filtering, balancing on a case by case basis.



old tech said:


> No one is saying that it does, rather it reduces the noise.  So you don't accept that the signal to noise ratio is a ratio?



I'm saying that noise reduction doesn't alter the signal at all to increase resolution. Ideally, it just removes noise. Some kinds, like the broadband NR I mention above is capable of significantly reducing resolution. I consider resolution to be related to distortion... the signal is altered. Within the bandwidth Nyquist covers, there is no real distortion.


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## old tech

gregorio said:


> Although operationally different, noise reduction like Dolby A or DBX is conceptually the same as pre/de emphasis and the RIAA curve used on LPs (which is a form of pre/de emphasis). Pre/De emphasis is an EQ curve applied to the master destined for LP (or some other media) and then an inverse of that curve is applied on playback, while Dolby NR and DBX are more sophisticated, based on companding rather than just EQ. Within the industry we often do not refer to this as noise reduction (NR) but as noise suppression, meaning that we're not reducing the amount of noise or distortion in the signal itself, we're just suppressing the amount of noise that will be added by some downstream two way process (such as cutting and playing an LP or recording to and playing back from magnetic tape). In other words, noise suppression doesn't increase resolution, it decreases resolution but not by as much as would be the case if you didn't employ it. Noise suppression effectively died with digital audio because, for example, copying a redbook master to a CD and then reading that data back from the CD should provide a bit perfect copy, IE. No noise is added by the two way process, so there's no additional noise to suppress and all you'd get is the lower resolution of the suppression process itself. The same is effectively true of processing/mixing digital audio, we use a much higher bit depth for processing/mixing so that the added noise (or quantisation error) is always well below the DR/resolution of the source recordings and destination media.
> G


Thanks for this.  NR was also used to reduce tape hiss and It still doesn't explain why some tape machines (and if I recall correctly) blank tapes would claim a higher SNR with Dolby than without.  A manufacturer would not know what pre or post recording processing might be done with the machine.


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## bigshot

The kind of Dolby used on tape machines assumed pre-emphasis and post-filtering.


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## gregorio

Arjey said:


> [1] Is there a way to remove noise from old records?
> [2] I know about the feature in audacity and other apps, but it doesn't really work.



1. Reduce "yes", remove "no". As a general rule, the more noise you remove, the more distortion you introduce. The trick is therefore to play with the settings of the NR application to reduce an amount of noise but not so much as to make the added distortion objectionable. This is a subjective determination and there's no one setting that always works because it's dependant on both the signal that we want to keep (the music) and the signal we want to remove (clicks, surface noise, etc.), both of which vary of course. The difficulty is that what appears to be a fairly obvious objective property is in fact not objective at all. The difference between terms such as "sound", "music" and "noise" seems fairly obvious but are effectively descriptions of human perception and personal value judgements rather than objective audio properties, which is why NR applications struggle to differentiate between them.

2. There are a number of applications that can reduce noise. The most widely used/regarded professionally are the CEDAR units and the iZotope RX suite of applications, neither are cheap and RX has a pretty steep learning curve. In practice, using a little of both usually gives better results than just one or the other and they are both significantly better than the feature built into audacity. Again though, it all depends on the input signal and the exact nature and amount of noise it contains. 



old tech said:


> [1] NR was also used to reduce tape hiss and It still doesn't explain why some tape machines (and if I recall correctly) blank tapes would claim a higher SNR with Dolby than without.
> [2] A manufacturer would not know what pre or post recording processing might be done with the machine.



1. It (my response) does explain why tape machines and blank tapes with noise suppression would claim a higher SNR but obviously not successfully  
Let's take a simplified example of pre/de emphasis noise suppression. The processes of cutting an LP and playing it back adds a significant amount of high freq noise (hiss). When we've finished creating our (freq balanced) master, we do one additional final step, ruin that freq balance by applying an EQ/boost of the high freqs, let's say a 10dB boost. This high freq boosted master is cut to LP, read/played by a consumer's TT and then an exactly opposite 10dB EQ/cut is applied. The net result is the same freq balanced master we started with before we applied the EQ boost and cut but we've reduced the high freq noise (hiss) added between the EQ boost and cut processes (the LP printing and playing process) by 10dB ... or rather, not quite by 10dB. Of course it's not exactly the same the master we started with, because we've used two EQ processes to get back to where we started, each of which adds some noise of it's own, let's say a total of 1dB of noise. So what we end up with is a loss of resolution, 1dB less dynamic range than the input signal (our master) but still a net improvement of 9dB less hiss than if we hadn't used it. This is the same concept as noise suppression on tapes and tape recorders; the recording, replay and media itself introduces significant amounts of noise (hiss), we can apply a process to the input signal and the inverse of that process on the output signal which reduces it. It's more sophisticated than a simple EQ boost and inverse EQ cut but the concept is identical, we loose a little resolution by applying these two equal and opposite processes but achieve a net improvement in the amount of hiss that would otherwise have been added.

2. In light of the above explanation, your statement doesn't make much sense. A tape recorder receives an input signal, applies the first stage of noise suppression to it and records this modified signal to the tape. When playing back the tape, the tape recorder applies the second (inverse) stage of noise suppression before output. So, a tape recorder manufacturer does of course know what pre and post recording processing is done, because they're the ones who've built it into their tape recorder!

G


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## old tech

gregorio said:


> 1. It (my response) does explain why tape machines and blank tapes with noise suppression would claim a higher SNR but obviously not successfully


Thanks again for the explanation.  I understand what pre-emphasis and its purpose is but (perhaps I'm just a bit thick in this area) I still can't get my head around why dolby, dbx, CX or any other noise reduction or noise suppression would not (if successful) raise SNR given the measure is a simple ratio or, if that is not how it works, on what basis do tape recorders and tape media claim a higher SNR when Dolby (or others) are employed.

Rather than using LP production/playback as an example, I might understand it better if you could explain it with a tape recording like most home consumers had done in the past.  Say we are making a tape copy of an LP at home, nothing exotic, RCA out from the amplifier to tape in on the tape deck.  The tape deck and the tape I am using both claim a higher SNR if I use Dolby.  So I make two tape copies, one with Dolby switched on and the other without Dolby.  I play the tapes back  - the one recorded with Dolby has Dolby switched on in playback and the recording without Dolby is played back with Dolby switched off.  

So according to the manufacturer of the deck and tape, the recording I am listening to with Dolby will have a higher SNR (and hence, resolution) than the recording without.  Is that really the case and if so, why?  If that is not the case, are the manfacturers telling porkies?



gregorio said:


> 2. In light of the above explanation, your statement doesn't make much sense. A tape recorder receives an input signal, applies the first stage of noise suppression to it and records this modified signal to the tape. When playing back the tape, the tape recorder applies the second (inverse) stage of noise suppression before output. So, a tape recorder manufacturer does of course know what pre and post recording processing is done, because they're the ones who've built it into their tape recorder!
> G



Apologies for that, I got a bit tangled there - must be the heat wave we are having over here.


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## gregorio

old tech said:


> [1] Thanks again for the explanation.
> [2] I understand what pre-emphasis and its purpose is
> [2a] but (perhaps I'm just a bit thick in this area) I still can't get my head around why dolby, dbx, CX or any other noise reduction or noise suppression would not (if successful) raise SNR given the measure is a simple ratio or, if that is not how it works, on what basis do tape recorders and tape media claim a higher SNR when Dolby (or others) are employed.
> [2b] Rather than using LP production/playback as an example, I might understand it better if you could explain it with a tape recording like most home consumers had done in the past.



1. You're welcome but apparently it still wasn't good enough 

2. In which case you do also understand the concept of Dolby, etc., noise suppression because they're conceptually the same.
2a. It would raise SNR, the same as my previous example with LPs and pre/de emphasis would raise SNR by 9dB.
2b. OK, but it's just a case of taking exactly the same example and exchanging the words "LP" and "TT" with the words "tape recorder" and "tape player" so: Lets say the tape recording, tape itself and tape player results in say 12dB of additional noise (hiss). We have an input signal into the tape recorder, we apply a 10dB high freq boost to it and record the result on the tape. We then play that tape but add a 10dB high freq cut before output. This process of boosting and then cutting HF adds lets say 1dB of noise (hiss). The end result (output of our tape player of the recorded input signal) therefore has 3dB less SNR than our input signal: +12dB of hiss added by the recorder, tape and player, +1dB of noise added by our two boost and cut processes and -10dB from the HF cut = +3dB noise (hiss). Therefore, we can say that this HF boost and cut process has improved SNR by 9dB (compared to not using it), because if we hadn't used it, we would end up with an SNR 12dB less than our input signal instead of 3dB less.

The only things different between my explanation and what actually happens is the example figures given and the fact that Dolby NR (and others) is not just a simple EQ boost and exactly opposite EQ cut but a more complex process based on compression of the input signal (pre-emphasis) and then the exact opposite application of expansion on the output signal (de-emphasis) but conceptually it's the same thing.

Hope this help?

G


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## old tech (Dec 17, 2019)

gregorio said:


> Hope this help?
> G


Yeah, got it now thanks.

Funnily enough, in our late teens/early adulthood, me and my circle of friends preferred the sound of Dolby for recording on high bias tapes but switching Dolby off on playback, particularly for our car stereos.  The enhanced top end sounded great, and without noticeable bass penalty.


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## TheSonicTruth

old tech said:


> Yeah, got it now thanks.
> 
> Funnily enough, in our late teens/early adulthood, me and my circle of friends preferred
> the sound of Dolby for recording on high bias tapes but switching Dolby off on playback,
> ...




Humans naturally gravitate toward a 'V'-shaped sound, mainly due to our natural sensitivity to mid- and upper-midrange frequencies.  I am unique in that I keep the tone controls on my car and home stereos flat, so what I'm playing comes through unaltered.


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## bigshot (Dec 17, 2019)

I used to add pre emphasis and play back without filtering too. I found that I could adjust the treble control to fine tune the high frequency filtering to sound better than flipping the Dolby switch and getting Dolby's overkill. Adjustable is always better than one-size-fits-all. Dolby was focused on removing hiss. It removed a little signal too along with it.


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## TheSonicTruth

bigshot said:


> I used to add pre emphasis and play back without filtering too. I
> found that I could adjust the treble control to fine tune the high
> frequency filtering to sound better than flipping the Dolby switch
> and getting Dolby's overkill. Adjustable is always better than
> ...




That could be the result of (albeit slight)non-standard encode and or decode.   Didn't the Dolby NR encode equipt(at the plant) require periodic calibration?

Also, I'm not sure if the decode process on cheaper cassette decks of the time, or in cars, conformed tightly to Dolby's standards.


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## bigshot

TheSonicTruth said:


> Humans naturally gravitate toward a 'V'-shaped sound.



That isn't what the Harman Curve looks like, and there's a lot of research and polling behind it.


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## TheSonicTruth

bigshot said:


> That isn't what the Harman Curve looks like, and there's a lot of
> research and polling behind it.



Yeah, a crooked smile with a bottom shelf and a peak in the upper mids.  But you get my point, no need to mince words bigshot.

Every car I'm driven that wasn't mine had either the bass & treble fully clockwise, or a combo of bass & treble boost and a mids cut.  That was my point.


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## bigshot

My car needs the opposite. It needs a curve like ^ But it has separate tweeters and a subwoofer.


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## gregorio

old tech said:


> [1] Yeah, got it now thanks.
> [2] Funnily enough, in our late teens/early adulthood, me and my circle of friends preferred the sound of Dolby for recording on high bias tapes but switching Dolby off on playback, particularly for our car stereos. The enhanced top end sounded great, and without noticeable bass penalty.



1. Great! I think maybe you were trying to complicate it, conceptually it's very simple. It's also a very good demonstration of the inherent problem with analogue audio, that no matter what you do, you always end-up with lower resolution. Using my example again, if the added hiss is greater than 12dB then our end result is a lower SNR by the same amount and if it's less, say 9dB then our end result would still be 1dB less SNR, due to the pre/de emphasis process itself. Analogue audio/music production was therefore always limited and the vast majority of music mixes had to be carefully planned, to minimise/not exceed a certain number of "bounce downs" or other processes and end-up with an unacceptable amount of noise.

2. I sometimes did that too, I don't think it was uncommon. There are several reasons for this: One is the very commonly overlooked fact that our hearing/perception of sound operates on the basis of relative levels/amounts, not absolute amounts and another is that most consumer tape recorders, players and the cassette tapes themselves often had such lousy mid and HF response that the added mid and HF compression/gain from the pre-emphasis (NR encoding) process effectively counteracted that lousy response, resulting in a flatter freq response without engaging the NR. So for most consumers the choice was between a flatter reproduction with more noise/hiss (by not engaging the NR decoding) or a mid/HF deficient reproduction with less hiss (by engaging the NR decoding) and some/many chose the former.



TheSonicTruth said:


> [1] Humans naturally gravitate toward a 'V'-shaped sound, mainly due to our natural sensitivity to mid- and upper-midrange frequencies.
> [2] I am unique in that I keep the tone controls on my car and home stereos flat,
> [2a] so what I'm playing comes through unaltered.



1. Sure we have a greater sensitivity to mid freqs but do you have any evidence that humans prefer ("naturally gravitate toward") a V-shaped sound? The evidence does not support that assertion. For example, the Harman studies demonstrated a preference for a flat response and if humans did prefer a V-shaped sound then that's what we'd expect to see in music masters, but we don't. Spectral analysis of the vast majority of music masters demonstrate a flat response or more commonly, a trend for a downward sloping response (similar to but not as extreme as pink noise). Certainly, V-shaped amplifier settings were very common but that was because most consumer systems had poor bass and HF response and so V-shaped settings compensated for this tendency, resulting in a flatter (not V-shaped) output. V-shaped consumer settings are somewhat less common today than they once were, because digital formats do not suffer the HF loss of analogue formats (particularly cassette tapes) so less HF compensation is generally required.

2. Firstly, that's certainly not unique, although of course many people do adjust their tone controls. And 
2a. Secondly, that's not really possible! If you've got an accurate full range speaker system and a very well controlled listening environment then "what you're playing" would "come through" somewhat unaltered (altered by relatively little) but of course very few people have accurate full range speaker systems and very well controlled listening environments, so changing their tone controls can somewhat compensate and provide a reproduction that is less altered by their listening circumstances. The situation is more complicated when listening in a car because the listening environment is not only generally much noisier than home listening environments but is constantly changing, often very significantly. For example, in my particular case, I have the tone controls set to add a little more bass (but flat in the mid and HF). In slow moving traffic at low speeds this setting results in a little too much bass but out on the highway (motorway), at higher speeds, it results in too little bass, because our perception is based on relative differences and at higher speeds (my car and other traffic), there is more low freq environmental noise (rumble, etc.). My setting is based on the response of my car speaker system and is a compromise between my two typical driving conditions. If the response of my car speaker system were a little more bass heavy to start with or if I only drove in the city in slowly moving traffic, then like you, I would set my tone controls to flat.



TheSonicTruth said:


> [1] That could be the result of (albeit slight)non-standard encode and or decode.
> [2] Didn't the Dolby NR encode equipt(at the plant) require periodic calibration?
> [3] Also, I'm not sure if the decode process on cheaper cassette decks of the time, or in cars, conformed tightly to Dolby's standards.



1. As far as I'm aware, not likely. The encode and decode circuitry built into tape recorders and players had to actually be purchased from Dolby, so there wasn't much opportunity for a recorder/player manufacturer to screw the encoding/decoding up.

2. Again, I don't think so. Some pro flavours of Dolby NR, Dolby A and SR for example, did require calibration but consumer versions (both encode and decode) were much simpler and designed not to need re-calibration.

3. It had to conform with Dolby standards. It was patented/copyrighted and the hardware had to be bought from Dolby themselves, as far as I remember. 

G


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## TheSonicTruth (Dec 18, 2019)

TheSonicTruth said:


> Post #5445: Every car I'm driven that wasn't mine had either the bass & treble fully clockwise, or a combo of bass & treble boost and a mids cut.  That was my point.





gregorio said:


> Sure we have a greater sensitivity to mid freqs but do you have any evidence
> that humans prefer ("naturally gravitate toward") a V-shaped sound?




From my own post, #5445, on top.  Perhaps 'smiley' EQ would be more accurate in describing it?  Either way, V-shaped, Smiley-shaped, it's all boom & sizzle to me. Not how I prefer to listen to music, recorded or otherwise.


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## bigshot

I think I'll stick with Harman.


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## gregorio

TheSonicTruth said:


> [1] From my own post, #5445, on top.
> [1a] Either way, V-shaped, Smiley-shaped, it's all boom & sizzle to me.
> [2] Not how I prefer to listen to music, recorded or otherwise.



1. So that's a "no" then, because ... 
1a. On a relatively flat (including the environment) system, setting the tone controls to a V/smiley shape would give an "all boom & sizzle" result but if the system isn't flat, say it has weak low and high freq response then V-shaped settings would give a flatter/ish response. So your post is NOT evidence that people prefer a "V-shaped" sound, it could just as easily be evidence that people prefer a flat sound and apply V-shaped tone settings to achieve that when driving.

2. Me neither. Again, that's why I effectively apply half a V-shaped tone control settings, not because I prefer "all boom" but because I prefer a flattish response and my raised bass setting somewhat achieves that in my car and my driving conditions.

G


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## TheSonicTruth

gregorio said:


> 1. So that's a "no" then, because ...
> 1a. On a relatively flat (including the environment) system, setting the tone controls to a V/smiley shape would give an "all boom & sizzle" result but if the system isn't flat, say it has weak low and high freq response then V-shaped settings would give a flatter/ish response. So your post is NOT evidence that people prefer a "V-shaped" sound, it could just as easily be evidence that people prefer a flat sound and apply V-shaped tone settings to achieve that when driving.
> 
> 2. Me neither. Again, that's why I effectively apply half a V-shaped tone control settings, not because I prefer "all boom" but because I prefer a flattish response and my raised bass setting somewhat achieves that in my car and my driving conditions.
> ...



2. Did you _measure _, as in, with a spectrometer, how flat your car response is while moving the tone controls?

Also, unless you've undertaken any sound-deadening action(inside your doors, the floor pan, or the trunk) in your vehicle, at least _half_ of what you're hearing _isn't _directly from your transducers.


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## gregorio

TheSonicTruth said:


> 2. Did you _measure _, as in, with a spectrometer, how flat your car response is while moving the tone controls?
> [3] Also, unless you've undertaken any sound-deadening action(inside your doors, the floor pan, or the trunk) in your vehicle, at least _half_ of what you're hearing _isn't _directly from your transducers.



2. Of course not. Firstly, it's not safe to measure while driving and Secondly, as I've already effectively mentioned twice, at what speed should I measure? At a higher speed I'll measure more bass due to environmental LF rumble (and therefore perceive less bass from the speakers), which is why I boost the bass (use a half a V-shaped setting). You're the one making the claim though, what did you measure with a spectrometer (and at what speed)?

3. Exactly! Haven't you read the posts you're responding to?

G


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## bigshot

Who worries about flat response in a car? I can't even imagine that. It would be an impossible task. You can get good sound in a car, but not studio quality. All you want is a good listenable balance. In my car, that means turning the bass and treble WAY down and working out a level that works with the subwoofer.


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## Glmoneydawg

You have headphones and home speakers to fiddle with....the car should be for enjoying music without fiddling with it...good music is enjoyable on the most primitive of systems.....relax and enjoy the drive/music,you can mess with your home system if you feel the need.


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## gregorio

bigshot said:


> [1] Who worries about flat response in a car? I can't even imagine that. It would be an impossible task.
> [2] You can get good sound in a car, but not studio quality.
> [3] All you want is a good listenable balance.
> [4] In my car, that means turning the bass and treble WAY down and working out a level that works with the subwoofer.



1. Not a flat response but a flatter response. It would be an impossible task to get a flat response, except maybe when the car is stationary in a quiet location. But when driving, how would you separate the traffic noise and the car's own noise from the speakers' output when measuring or get a perceptually flat response, as the traffic/car's own noise varies so much?

2. Not sure you can get even a "good sound" in a car. I suppose it depends on what you mean by "good" and maybe for one specific condition (say stationary for example). 

3. Even that is impossible to achieve, unless your driving conditions never change or only change very little.

4. So in answer to your first question: You do! Although we're talking about "flatter" (somewhat flattish) rather than anywhere particularly close to actually "flat". The point I was trying to make is that you don't like/prefer an inverted V-shaped sound and I don't like a half V-shaped sound (raised bass), we both like roughly the same thing, a flattish response but to achieve that requires completely different (almost opposite) settings because our cars, the sound systems in them and probably our driving conditions are very different (mine doesn't have a sub for example).



Glmoneydawg said:


> You have headphones and home speakers to fiddle with....the car should be for enjoying music without fiddling with it...



I think a lot of people do fiddle with their car systems. In my case, just a couple of times to get a somewhat flatter response but it's a compromise setting and not worth spending more than a few minutes on. I would think I'm probably not far from the average car owner in that regard, although I should imagine there are quite a few who don't even go that far and of course there's a very small number who go really mad and spend many thousands on custom installs. Never really got that myself, it's a bit like trying to train a donkey to win the Kentucky Derby in my opinion  

G


----------



## sander99

Smart DSP adapting to driving conditions? (Using information like speed, motor's RPM, sensors measuring the cars vibration, outside noise).

Funny coincidence: someone in the Smyth Realiser A16 thread is planning to do a PRIR measurement of his car audio system today.


----------



## old tech

Perhaps it is not possible to achieve a "good sound" in a car from the perspective of high fidelity, but a "good sound" from the perspective of subjective euphony is achievable.

The best sound system I had in a car was a Pioneer component system, back in the early 1980s with separate woofers, mids and tweeters.  It was a cassette tape system but it sounded great - even the component FM radio sounded really good.  It also had great sounding bass without the need of a subwoofer (which is just as well as they didn't exist in car audio back then).  Apart from the stereo, I think the car itself contributed to the good sound - it was a boxy early Falcon with a large boot.

I currently drive a Jeep Grand Cherokee and it has a crap factory sound system - despite multi speakers and sub.  So much so that I rarely play music in it but mainly listen to talkback and current affairs type of shows. I will definitely prioritise sound with my next car, even if it means fitting an after market unit.


----------



## bigshot

I wonder if people have audiophile telephones and balanced response intercoms?


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## TheSonicTruth (Dec 21, 2019)

old tech said:


> Perhaps it is not possible to achieve a "good sound" in a car from the perspective
> of high fidelity, but a "good sound" from the perspective of subjective euphony is achievable.
> 
> The best sound system I had in a car was a Pioneer component system, back in the early 1980s
> ...



A relatively recent Grand Cherokee? It's not the stereo - or the speakers.   That thing is a tin can compared to your old Falcon. Sheet metal panels half the gauge(if you're lucky) of what the Falcon was built out of.

Despite what certain others on here believe, I'm going to suggest you are listening to resonances from body panels as much as from the speakers themselves.  It will need acoustic sound-deadening to even get close to what you were used to in the older car.


----------



## old tech (Dec 22, 2019)

TheSonicTruth said:


> A relatively recent Grand Cherokee? It's not the stereo - or the speakers.   That thing is a tin can compared to your old Falcon. Sheet metal panels half the gauge(if you're lucky) of what the Falcon was built out of.
> 
> Despite what certain others on here believe, I'm going to suggest you are listening to resonances from body panels as much as from the speakers themselves.  It will need acoustic sound-deadening to even get close to what you were used to in the older car.



Current model.  The sheet metal panels are quite thick by modern car standards and the car is quieter and way better insulated than an old Falcon.  I think the Falcon having a large sealed boot is part of the story.

My previous car (a large Holden Commodore sedan) also had a mediocre stereo sound despite being the premium model.  It was better than the Jeep's stereo though.  It is not just a Jeep thing either, all the cars I looked at had mediocre sounding stereos (of course I didn't look at every car or high end cars for that matter) and the way they are integrated in the car interior makes it difficult and expensive to upgrade.

Even accepting that the Falcon had an aftermarket car stereo, the part that gets me is why is the sound quality in most new cars so poor compared to the earlier car after nearly 40 years of technical advances and cost reductions?


----------



## bigshot

Stock stereos are mediocre. You can do better custom, but not anywhere in the league of a halfway decent home speaker system. There is no reason to argue about car stereos because we are all well aware of the limitations.


----------



## TheSonicTruth (Dec 22, 2019)

bigshot said:


> Stock stereos are mediocre. You can do better custom, but not anywhere
> in the league of a halfway decent home speaker system. There is no
> reason to argue about car stereos because we are all well aware of the
> limitations.




Still, upgrading the speakers can help, a little.  But I feel that keeping the EQ/tone controls flat, and any 'enhancers', if present, disabled, keeps the signal path more transparent, even if the hypothetical _measured_ result might be anything but 'flat'.

Sometimes, if I have a full round of passengers, I'll drop the mid control a little so the music doesn't compete with the conversation, but that's it.


----------



## castleofargh

> *Discussion*
> 
> Respect the OP's (original poster's) intentions and try and keep comments on topic. If you feel a strong need to (continue to) discuss a particular, but off-topic subject in a thread, start your own thread and direct post a link in the thread rather than derail that thread. If it's not a discussion worthy for a thread, consider taking it to private messages.
> 
> If it is a one-off reply, you can also put the reply in a spoiler tag and rename it "Off-topic".


----------



## Harold999

Had my first real experience today with 24bits/96khz music. Subscrided to Tidal Masters.
I can't hear any difference between 320kbps vs Tidal Hifi vs Tidal Masters, so i can safe me the 10 extra bucks each month.


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## ChrisIsAwesome (Dec 29, 2019)

Okay, so despite any kind of technical knowledge i agreed with what is written on the first page and decided i would standardize my music library to 16/44.1. But before doing that i decided to run a quick test, first I've ever done.
Using Coldplay- Parachutes album and Asura - 360. Using AK SP2000, Andromeda. I tied variations of 16/44.1, 16/96, 24/16, 24/96 WAV all from the same original recording downloaded from HD music.

To my shock and annoyance.
I can pick it up pretty consistently, 24bit had more dimension to the sound stage while 16 bit was thinner.
44.1 vs 96khz, now this i'm not so sure about. I couldn't confidently pick a difference.

But to me 24/96 sound fuller and smoother than 16/44.1 definitely.  more analog vs digital kind of impression.
So now i'm annoyed and confused.

Edit: im hearing more depth, definitely


----------



## chef8489

ChrisIsAwesome said:


> Okay, so despite any kind of technical knowledge i agreed with what is written on the first page and decided i would standardize my music library to 16/44.1. But before doing that i decided to run a quick test, first I've ever done.
> Using Coldplay- Parachutes album and Asura - 360. Using AK SP2000, Andromeda. I tied variations of 16/44.1, 16/96, 24/16, 24/96 WAV all from the same original recording downloaded from HD music.
> 
> To my shock and annoyance.
> ...


Problem is it's not a valid test. You are not doing a controlled double blind test.


----------



## bigshot

Level matched, blind, direct A/B switched.


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## old tech (Dec 30, 2019)

ChrisIsAwesome said:


> But to me 24/96 sound fuller and smoother than 16/44.1 definitely.  more analog vs digital kind of impression.



How can it be an improvement if it sounds more "analog"?  Do you prefer a more noisy, less clear, less bass/treble, less accurate timbre and a narrowed instrument separation presentation to your music?

Would you prefer an analog looking TV picture over a digital broadcast?

Seriously though, even if you prefer a more muffled dull analog sound, how can you be sure that this is the difference you heard and that the difference is attributed to the formats?  Did you control for different masters, sound levels, expectation biases and so on (eg double blind testing using foobar for example)?


----------



## ChrisIsAwesome

I didn't say improvement anywhere, just the difference that i believe i hear. Although what others have said is true, not valid unless i do it blind test i suppose.
I'm just shocked and annoyed that i believe i can hear a difference, since its against my best interest. Who wants excessive file size. 

I like things to be neat and orderly and having everything standardizes to 16/44.1 would be nice i think. A neat music library.


----------



## old tech

ChrisIsAwesome said:


> I didn't say improvement anywhere, just the difference that i believe i hear. Although what others have said is true, not valid unless i do it blind test i suppose.
> I'm just shocked and annoyed that i believe i can hear a difference, since its against my best interest. Who wants excessive file size.
> 
> I like things to be neat and orderly and having everything standardizes to 16/44.1 would be nice i think. A neat music library.


I have several hi res files that sound different to 16/44 files of the same song, but I know that this is because of different masters.  I was able to confirm this through testing with Foobar as it uses the same hi res file and converts it to 16/44 for the comparison and I can't then hear a difference. Try it for yourself using the DBX plug in and your 24/96 file.

Just out of interest, why do you want to standardise your files to 16/44?  Do you intend making CDs or just short of storage space?  I have around 20,000 files in my collection and they vary from various lossy formats of different bit rates, 16/44, 24/44, 24/48, 24/96, 24/192 and DSD.  It doesn't matter to me as storage is not an issue, they all are indexed without any hassles and I stream them from a server.


----------



## ChrisIsAwesome

I use a DAP so limited space


----------



## TheSonicTruth

old tech said:


> Would you prefer an analog looking TV picture over a digital broadcast?



I personally prefer an accurate(IE calibrated) TV picture, regardless of broadcast format or resolution.

I've seen people spend money on a big 4K QLED, only to leave it in factory out-of-box settings, typically 'Vivid' or Dynamic, and it's unwatchable for anything save five minutes of news highlights.


----------



## bigshot (Dec 30, 2019)

If the movie I'm watching has a lousy color timing, I'll adjust beyond calibration. I just keep the calibration as a saved setting to use as a starting place. Most of the time it works fine at calibration, but sometimes certain transfers need some help. I also adjust a bit depending on whether the ambient light in the room is bright or dark.



ChrisIsAwesome said:


> I'm just shocked and annoyed that i believe i can hear a difference, since its against my best interest. Who wants excessive file size.



Welcome to the wonderful, unpredictable world of expectation bias! If you could control it, you could be happy with anything, regardless of what it sounds like!


----------



## castleofargh

ChrisIsAwesome said:


> Okay, so despite any kind of technical knowledge i agreed with what is written on the first page and decided i would standardize my music library to 16/44.1. But before doing that i decided to run a quick test, first I've ever done.
> Using Coldplay- Parachutes album and Asura - 360. Using AK SP2000, Andromeda. I tied variations of 16/44.1, 16/96, 24/16, 24/96 WAV all from the same original recording downloaded from HD music.
> 
> To my shock and annoyance.
> ...


Based on many trials done by many people, it is really, really, really unlikely that you would notice the difference caused by the loss of resolution itself(at or above 16/44) with music and normal listening conditions. It could happen if you tried to find a difference and created the nominal conditions for those differences to be noticed. Then of course the likelihood to perceive a change increases proportionally. But with typical music and typical listening levels, audible transparency is the expected result between 16/44 and 24/44 or higher. And even if you were to still perceive a difference on occasion, I have never seen any proper experiment suggesting that it would result in a subjective impressions as consistent as the one you describe.

So it seems more likely to me that you're fooling yourself(like we sadly all do too many times everyday), because you didn't perform a proper a conclusive listening test. Or the second possibility, although less likely, could be that the device doesn't handle 16/44 the way it handles higher resolution and somehow ends up causing an audible difference. Some measurements would obviously help settling the second possibility, while blind testing would help dealing with the first.
The desire to know the truth, would require more serious experiments than what you did. That much is a fact. But as far as personal enjoyment is concerned, if for whatever reason(audio or not), you do feel a consistent improvement with hires, then IMO you should stick to that. Pleasure isn't always explained in a rational way.


----------



## ChrisIsAwesome

castleofargh said:


> Based on many trials done by many people, it is really, really, really unlikely that you would notice the difference caused by the loss of resolution itself(at or above 16/44) with music and normal listening conditions. It could happen if you tried to find a difference and created the nominal conditions for those differences to be noticed. Then of course the likelihood to perceive a change increases proportionally. But with typical music and typical listening levels, audible transparency is the expected result between 16/44 and 24/44 or higher. And even if you were to still perceive a difference on occasion, I have never seen any proper experiment suggesting that it would result in a subjective impressions as consistent as the one you describe.
> 
> So it seems more likely to me that you're fooling yourself(like we sadly all do too many times everyday), because you didn't perform a proper a conclusive listening test. Or the second possibility, although less likely, could be that the device doesn't handle 16/44 the way it handles higher resolution and somehow ends up causing an audible difference. Some measurements would obviously help settling the second possibility, while blind testing would help dealing with the first.
> The desire to know the truth, would require more serious experiments than what you did. That much is a fact. But as far as personal enjoyment is concerned, if for whatever reason(audio or not), you do feel a consistent improvement with hires, then IMO you should stick to that. Pleasure isn't always explained in a rational way.



Thanks, i was curious to see if that consensus had changed at all since the original post in 2009. Not in regards to the science of file format as its well established, but more so with the hardware which has evolved. Unsure if hardware responded differently to different file resolution.
You've put my mind at ease, i'll go back to not worrying about these things.


----------



## old tech

ChrisIsAwesome said:


> Thanks, i was curious to see if that consensus had changed at all since the original post in 2009. Not in regards to the science of file format as its well established, but more so with the hardware which has evolved. Unsure if hardware responded differently to different file resolution.
> You've put my mind at ease, i'll go back to not worrying about these things.


Evolution of electrical hardware will not change the specifications of 16/44, 24/96.  What does need to evolve to hear the difference between these formats are our ears and brains. Perhaps one day in the future with genetic engineering and hearing hardware implants we may get to a point where musicians can write and play 24 bit music, mixing and mastering engineers can hear and work with music beyond 98db and a frequency response beyond 20 khz and consumers will be able to listen to it.  
Perhaps check back in this thread in 50 years time, rather than 10.


----------



## bigshot

Amps and players have been audibly transparent for some time. The only thing that might have improved is transducers, but those are still so far behind the electronics, it's really doubtful that you could hear the difference even with excellent headphones.


----------



## gregorio

ChrisIsAwesome said:


> [1] To my shock and annoyance. I can pick it up pretty consistently,
> [2] 24bit had more dimension to the sound stage while 16 bit was thinner. ... But to me 24/96 sound fuller and smoother than 16/44.1 definitely.   ... im hearing more depth, definitely
> [3] I'm just shocked and annoyed that i believe i can hear a difference, since its against my best interest.



1. Can you really though? As others have said, you need a controlled blind/double blind test because it's trivially easy to differentiate if you know in advance which is which. I cover this more in #3 below.

2. As castleofargh stated, what you describe here is NOT what is actually happening (as explained in the OP); 24bit doesn't have more dimension to the sound stage, is not fuller, smoother or have more depth. The ONLY difference is that 16bit has a tiny bit of extra noise, which you won't be able to hear unless you listen at a playback level well above comfortable/reasonable and even then, only under certain circumstances.

3. This really is the heart of the matter and indeed, at the heart of more than a fair bit of what goes on in the audiophile world. On the one hand: Generally no one wants to admit they've been fooled, are imagining things or are not hearing accurately (especially as critical listening is a requisite for audiophiles) and in addition, the belief in what one hears/senses is strong, for example the old cliche that "seeing is believing", despite countless examples of optical illusions and that everyone is aware images can be and routinely are manipulated. For many audiophiles both of these add up to an unassailable belief in what they are hearing and being unassailable, anything/everything which disputes this belief must therefore be wrong; demonstrated facts, proven science, blind testing, objective measurements, everything, it doesn't matter! On the other hand though: Being fooled and imagining things or more precisely, what we think/believe we're hearing being changed/affected by biases (and therefore being inaccurate) is not only NOT a bad thing, it's actually vital! For 500 years or so, virtually all western music has been based on bias, namely the manipulation of expectation bias, the expected continuity of rhythms, the expectation of chord and melodic progressions and the resolution of dissonance. Without expectation bias affecting what we think we're hearing, it would be impossible to appreciate western music, it would all just sound like semi-random noise with absolutely no meaning or emotional impact. In other words, without bias affected hearing, music doesn't exist! 



ChrisIsAwesome said:


> Thanks, i was curious to see if that consensus had changed at all since the original post in 2009. Not in regards to the science of file format as its well established, but more so with the hardware which has evolved. Unsure if hardware responded differently to different file resolution.



The hardware has evolved; in general (as with just about all modern technology), it's become better/more accurate or cheaper, or both. There's generally fewer incompetent digital audio devices, audibly transparent/perfect digital reproduction is even cheaper, the specifications of expensive devices have improved (though not audibly) and the issue of some devices operating better at higher sample rates, which was occasionally the case 10-15 years ago (to save money or through incompetence), is far less common today. What you believe you're hearing (a difference between 16/44 and 24/96) is possible, for example due to some seriously dodgy conversion software but that was relative unlikely even 10-15 years ago. So unless you're using conversion software that's ancient (and one of the dodgy ancient ones) and/or have made a serious error in the conversion settings, we can pretty much rule this out as a possibility.

G


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## ChrisIsAwesome

Thanks for taking the time everyone, you've put my mind at ease.


----------



## tmb821

I’m relatively new to the hifi world. At least in the headphone world. I have been on a quest to make music enjoyable. See, I have a pretty serious hearing impairment. So what most people hear, I just simply can’t. One of the biggest change I hear was going from a standard format to a 16bit/44.1khz format. It brought more clarity to the music. In all honesty I can’t tell a difference between the 16bit and 24bit formats that come up. But, maybe that’s because I can’t hear anyway. 
Then there is the fact that I apply a eq to all music I listen to. My eq looks like a upside down v. But, at least I can hear the mid ranges now. I just wanted to say all this to say thanks for all the info in this thread, very helpful to a deaf guy like me.  I say let all the people who say they can hear a difference in the bitrate have their higher bitrate.
And as far as car audio, it is possible to have very high quality sound, you just have to paaaaaay for it. Upwards of $30k is not unusual. I know, I did it in my younger years. It takes a lot of equipment and time and work, but the results are tremendous, at least to me. 
Again, thanks for all the info.


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## chennaxin95

Coming from computer vision side this is a great write up. I often mentally compare audiophile stuff with camera and displays, and the bit depth can be seen as how many colors your image sensor could capture.


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## Sonic Defender (Jan 3, 2020)

It makes me sad inside when I read comments from ardent 24bit fans who simply refuse to use proper testing saying such maddeningly things as "I trust my ears, that is good enough" as if that somehow constitutes evidence anymore than when your parents told you the tooth fairy was real constituted evidence of winged tooth bandits with spare change. I weep inside.


----------



## Davesrose (Jan 3, 2020)

chennaxin95 said:


> Coming from computer vision side this is a great write up. I often mentally compare audiophile stuff with camera and displays, and the bit depth can be seen as how many colors your image sensor could capture.



As someone with a professional background in computer graphics, I don't see it as such a close analogy.  As so many pages past highlight, audio bit depth can cover any type of resolution or dynamic range, and don't need improvements.  Since the beginning of digital imaging, digital cameras could capture a palette reaching millions of colors (3 channel 8bit per channel gives you 16 million colors).  However, they're really lacking when it comes to realistic light modeling as well as the limits of human vision: which requires a higher dynamic range.  Image sensors are in component color space: which places emphasis in dynamic range....14-16bit per channel space are the common exposure for base ISO (16bpc=65,536 shades of tone).  The extreme HDR formats are 32bit per channel (4.29 billion tones)....who knows if digital cameras will ever be able to record that kind of space in one exposure, but it's been usable for its application for quite a number of years.


----------



## chennaxin95

Davesrose said:


> As someone with a professional background in computer graphics, I don't see it as such a close analogy.  As so many pages past highlight, audio bit depth can cover any type of resolution or dynamic range, and don't need improvements.  Since the beginning of digital imaging, digital cameras could capture a palette reaching millions of colors (3 channel 8bit per channel gives you 16 million colors).  However, they're really lacking when it comes to realistic light modeling as well as the limits of human vision: which requires a higher dynamic range.  Image sensors are in component color space: which places emphasis in dynamic range....14-16bit per channel space are the common exposure for base ISO (16bpc=65,536 shades of tone).  The extreme HDR formats are 32bit per channel (4.29 billion tones)....who knows if digital cameras will ever be able to record that kind of space in one exposure, but it's been usable for its application for quite a number of years.


Hopefully it won’t take forever. Currently good camera can do 15-16 stops, while human eye is about 20 stops. The same bit depth argument can go for 8bit vs 10bit color video recordings. You probably don’t need 10bit recordings for watching them but when you start editing those clips the artifacts will show.


----------



## TheSonicTruth

chennaxin95 said:


> Hopefully it won’t take forever. Currently good camera can do 15-16 stops,
> while human eye is about 20 stops. The same bit depth argument can go for
> 8bit vs 10bit color video recordings. You probably don’t need 10bit recordings for
> watching them but when you start editing those clips the artifacts will show.



YES!  What audio(and video)-philes must realize is that those higher sampling rates and bit-depths are useful only on the production side.  Do everything right there, and the project will translate well at consumer specs.


----------



## Davesrose (Jan 3, 2020)

chennaxin95 said:


> Hopefully it won’t take forever. Currently good camera can do 15-16 stops, while human eye is about 20 stops. The same bit depth argument can go for 8bit vs 10bit color video recordings. You probably don’t need 10bit recordings for watching them but when you start editing those clips the artifacts will show.



And some of the best RED cameras will natively record 16bpc RAW.  Tonemapping 12bit to 10bit or 8bit is pretty seamless now.  I'm amazed by how much I can compress with new formats like h.265 and not see artifacts.  Early mastered DVDs, though (with MPEG-2), can show a ton of artifacts on a large sized TV.  We'll see how long it takes for TVs to improve contrast range: that's the current limitation.  I heard an interview with Dolby when they were coming out with Dolby Vision.  They mentioned that they tried using native 12stop/bit displays: which costs thousands of dollars and required special super cooling.


----------



## bigshot

Donald Fagin's "The Nightfly" was recorded and mixed at 16/44.1 and it is one of the best sounding albums ever produced. More data rate won't make it sound any better. It will just make it more flexible for filtering and balance leveling. Just like with photography, if you shoot for the "sweet spot" of your latitude, you can get great results even with limited range.


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## TheSonicTruth (Jan 3, 2020)

bigshot said:


> Donald Fagin's "The Nightfly" was recorded and mixed at 16/44.1and it is
> one of the best sounding albums ever produced. More data rate won't
> make it sound any better. It will just make it more flexible for filtering and
> balance leveling. Just like with photography, if you shoot for the "sweet spot"
> of your latitude, you can get great results even with limited range.




Is that true - that an album back in that era would have been produced at consumer spec?

I have been told, on here by the way, that 24bit/96k sampling was the studio norm going back to the eighties.


----------



## bigshot

Yes, they were working at 16/44.1 in the early 80s if they recorded digitally. I think Dire Straits' Brothers In Arms was 16/44.1 too.


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## 524419 (Jan 3, 2020)

bigshot said:


> Amps and players have been audibly transparent for some time. The only thing that might have improved is transducers, but those are still so far behind the electronics, it's really doubtful that you could hear the difference even with excellent headphones.


I can reliably hear the differences between 24/44, 24/96, and 24/192. Not to mention DSD Files.
Dynamics, and noise are pretty consistently what stands out between all these files, less noise on higher quality files, better dynamic range.
I hear the difference so much so, that I can no longer enjoy 16/44 files. They sound muddy and congested.
DSDs sound more analogue, "like I am there" than any other format.


----------



## chef8489

Diet Kokaine said:


> I can reliably hear the differences between 24/44, 24/96, and 24/192. Not to mention DSD Files.
> Dynamics, and noise are pretty consistently what stands out between all these files, less noise on higher quality files, better dynamic range.
> I hear the difference so much so, that I can no longer enjoy 16/44 files. They sound muddy and congested.
> DSDs sound more analogue, "like I am there" than any other format.


Prove it. Beside the fact dsd usually uses different masters than the 16/44. From what you wrote looks like you just read too much on the internet and tricking your ears with your brain.


----------



## 524419

chef8489 said:


> Prove it. Beside the fact dsd usually uses different masters than the 16/44. From what you wrote looks like you just read too much on the internet and tricking your ears with your brain.


You can believe whatever you want.
To my ears the difference is as clear as it can be, If I was you, I would get my ears checked, or invest in better gear.


----------



## TheSonicTruth (Jan 3, 2020)

bigshot said:


> Yes, they were working at 16/44.1 in the early 80s if they recorded
> digitally. I think Dire Straits' Brothers In Arms was 16/44.1 too.



Oh you "think" BIA was16/44.1?

So when(approximately) do you "think" they started using 24bit depth and sampling rates higher than 44.1kHz during RMM(Recording, Mixing, & Mastering)?

Because in a thread long dormant, I made the exact same statement as you did, above, regarding which specs were used when, and a certain other  high-profile member here corrected me emphatically.  So now the onus is on you and him to reconcile each other's facts.


----------



## megabigeye

This thread is great!  I get a pretty good laugh every few days.


----------



## chef8489

Diet Kokaine said:


> You can believe whatever you want.
> To my ears the difference is as clear as it can be, If I was you, I would get my ears checked, or invest in better gear.


My equipment is great as are my ears and I have been in this community for a really long time. Enough time to hold double blind tests at universities as well as see countless people like you that say and believe they can hear a difference. When true tests are done they cant hear a difference. Just you saying dsd sounds more analog to you. Analog is far inferior to digital. This is the science forum. It is up to you to provide proof that you can hear a difference, otherwise you are just talking bs as the science says otherwise. You dont like that, dont post in the sound science forum.


----------



## old tech (Jan 3, 2020)

TheSonicTruth said:


> *Oh you "think" BIA was16/44.1*?
> 
> So when(approximately) do you "think" they started using 24bit depth and sampling rates higher than 44.1kHz during RMM(Recording, Mixing, & Mastering)?
> 
> Because in a thread long dormant, I made the exact same statement as you did, above, regarding which specs were used when, and a certain other  high-profile member here corrected me emphatically.  So now the onus is on you and him to reconcile each other's facts.



BIA was recorded at 16/44 as were most digital recordings prior to the 1990s.  I read an interesting interview with Neil Dorsfman (Dire Strait's long time producer) which covered this aspect.  He also discussed how they had to experiment with new recording arrangements, miking ect because 16/44 was picking up studio noise (things like band members shuffling their feet) which analog never had the resolution to do so.  One misconception though is that the CD is DDD.  It is actually DAD as it was mixed through an analog work station.

What I find amusing though is all the vinylphiles paying top dollar for MFSL half speed 45 rpm records of this album rather than the original early 1985 CD, which is a bit perfect copy of the master.  Surely that proves that these vinylphiles do not value high fidelity but rather are chasing an artificial vinyl sound, ie the distortions which they find subjectively pleasing.


----------



## old tech

Diet Kokaine said:


> I can reliably hear the differences between 24/44, 24/96, and 24/192. Not to mention DSD Files.
> Dynamics, and noise are pretty consistently what stands out between all these files, less noise on higher quality files, better dynamic range.
> I hear the difference so much so, that I can no longer enjoy 16/44 files. They sound muddy and congested.
> DSDs sound more analogue, "like I am there" than any other format.


Of course, you believe you are exceptional and the science, objective testing does not apply to you.  Why are you posting here?

There is also a major logical inconsistency with your belief. You claim you can hear less noise and a better dynamic range with hi res file, yet you say DSD sound more analogue.  Given that analogue has more noise and less dynamic range than 16/44, surely that is evidence that your ears are not very good and your listening skills leave a lot to be desired.


----------



## bigshot

Diet Kokaine said:


> You can believe whatever you want.
> To my ears the difference is as clear as it can be, If I was you, I would get my ears checked, or invest in better gear.



You can believe whatever you want, but tests have shown that 16/44.1 is audibly transparent. If I was you, I would check my bias and invest in some research about how digital audio works. I’d suggest starting by reading the article at the link in my sig titled “CD Sound Is All You Need”. If you have any questions after reading that article, let me know.


----------



## bigshot

old tech said:


> What I find amusing though is all the vinylphiles paying top dollar for MFSL half speed 45 rpm records of this album rather than the original early 1985 CD, which is a bit perfect copy of the master.



That album is released on SACD too. The only reason for it is the multichannel mix.


----------



## TheSonicTruth

old tech said:


> BIA was recorded at 16/44 as were most recordings prior to the late 1980s.  I read an interesting interview with Neil Dorsfman (Dire Strait's long time producer) which covered this aspect.  He also discussed how they had to experiment with new recording arrangements, miking ect because 16/44 was picking up studio noise (things like band members shuffling their feet) which analog never had the resolution to do so.  One misconception though is that the CD is DDD.  It is actually DAD as it was mixed through an analog work station.
> 
> What I find amusing though is all the vinylphiles paying top dollar for MFSL half speed 45 rpm records of this album rather than the original early 1985 CD, which is a bit perfect copy of the master.  Surely that proves that these vinylphiles do not value high fidelity but rather are chasing a "vinyl sound", ie the distortions which they find subjectively pleasing.



Thank you for that history, but my question remains: When did higher sample rates and bit depths start to be used in production?


When you record something in digital, then apply processes(EQ, comp, etc) digitally to it, it's like a recording of a recording of a recording, and quantization noise and other artifacts slowly accumulate.

That is why 24bit and higher bit depths, and 92kHz and higher sample rates are used in the production process before exporting down to 16/44.1  for consumer mass-distribution.


----------



## old tech

TheSonicTruth said:


> Thank you for that history, but my question remains: When did higher sample rates and bit depths start to be used in production?
> 
> 
> When you record something in digital, then apply processes(EQ, comp, etc) digitally to it, it's like a recording of a recording of a recording, and quantization noise and other artifacts slowly accumulate.
> ...


1996 according to the AES audio timeline.
http://www.aes.org/aeshc/docs/audio.history.timeline.html


----------



## bigshot

If you record carefully and don’t require a lot of post processing in the mix, 16/44.1 is more than enough. It’s better than most 24 track analogue decks are capable of, so any analogue master is going to be more hobbled than digital ones.


----------



## aukhawk

A commonly-used storage format in studios in the late '80s was DAT.  That is a 16/*48* format.  I know, I was there.


----------



## Daiyama

Diet Kokaine said:


> I can reliably hear the differences between 24/44, 24/96, and 24/192. Not to mention DSD Files.
> Dynamics, and noise are pretty consistently what stands out between all these files, less noise on higher quality files, better dynamic range.
> I hear the difference so much so, that I can no longer enjoy 16/44 files. They sound muddy and congested.
> DSDs sound more analogue, "like I am there" than any other format.



blablabla 
Typical internet speech, anyone can write anything and proving nothing.....


----------



## old tech (Jan 4, 2020)

aukhawk said:


> A commonly-used storage format in studios in the late '80s was DAT.  That is a 16/*48* format.  I know, I was there.


That would have been around the time digital consoles first start appearing in studios eg the Yamaha DMP 7? 

I owned a DAT machine a long time ago and it also recorded and stored 16/44 - wasn't 16/48 generally more for video audio tracks prior to that? 

But you are right though, there are many 16/48 as well as 16/44 recordings prior to the mid 1990s (also 24/44 and 24/48 later on - not sure of the history on those depth/rates).


----------



## TheSonicTruth

bigshot said:


> If you record carefully and don’t require a lot of post processing in the
> mix, 16/44.1 is more than enough. It’s better than most 24 track analogue
> decks are capable of, so any analogue master is going to be more hobbled
> than digital ones.




I'm genuinely surprised to be hearing such from you.


----------



## gregorio

Diet Kokaine said:


> [1] I can reliably hear the differences between 24/44, 24/96, and 24/192. Not to mention DSD Files. Dynamics, and noise are pretty consistently what stands out between all these files, less noise on higher quality files, better dynamic range.
> [2] DSDs sound more analogue, [2a] "like I am there" than any other format.
> [3] If I was you, I would get my ears checked, or invest in better gear.



1. There is no dynamic or noise differences between any of those formats, they are all have identical dynamic range and noise, except DSD which has significantly less dynamic range than any of the others. So how can you reliably hear differences which do not exist?

2. That's a bit of a sad indictment, do you really think DSD sounds as bad as analogue?
2a. I don't understand, if you think analogue sounds "like I am there", why do you think they invented digital audio instead of just sticking with vinyl or cassette tapes?

3. You're the one "reliably hearing" differences that do not exist, so shouldn't it be you getting your "ears checked"?



old tech said:


> One misconception though is that the CD is DDD. It is actually DAD as it was mixed through an analog work station.



The CD codes were rather misleading. DDD doesn't mean an album was mixed digitally, the 3 letters ALL refer to the recording of the 3 stages: The first D indicates the audio was recording in digital, the second D indicates the mix was recorded in digital and the third D indicates the master was recorded in digital. So, you can mix and master with an analogue desk/equipment and it's still DDD (rather than DAA) providing you record the analogue mix and master in digital and indeed, this is the case with pretty much all DDD rock/popular music genres up until the millennium.



TheSonicTruth said:


> [1] Thank you for that history, but my question remains: When did higher sample rates and bit depths start to be used in production?
> [2] When you record something in digital, then apply processes(EQ, comp, etc) digitally to it,
> [2a] it's like a recording of a recording of a recording, and quantization noise and other artifacts slowly accumulate.
> [3] That is why 24bit and higher bit depths, and 92kHz and higher sample rates are used in the production process before exporting down to 16/44.1 for consumer mass-distribution.



1. 32kHz, 44.1kHz and 48kHz were the standard for many years, 96kHz started to be widely available to studios in the early 2000s. 20bit recording started to be employed in the early 1990s and 24bit again in the early millennium.

2. You're confusing "recording something in digital" and mixing ("applying processes") in digital. At the beginning of the 1990's (and even a few years earlier) there were digital mixers that processed internally at around 28bits but they weren't particularly common. By the late 1990's professional digital and virtual mixers were 32bit float or 48bit fixed, today they're mostly 64bit float.
2a. Not really. There are no other artefacts besides quantisation noise and that's defined by the bit depth of the mix environment. So with 48bit fixed, the quantisation noise is roughly at -288dB and around -370dB with 64bit float. You literally need hundreds or thousands of processors for quantisation noise to accumulate to anywhere near audible levels. This is independent of the recording bit depth, these quantisation noise figures are the same regardless of whether the recorded audio being mixed is 24bit or 16bit.

3. There are and have never been any 24bit professional mixers as far as I'm aware. Sampling rates are a bit more complicated; 48kHz is by far the most common sampling rate used professionally/commercially, as it's the worldwide standard for film and all SD and HD television. There are few situations where a higher sample rate is beneficial when processing but this is again independent of the recorded audio sample rate and even independent of the sample rate of the mix environment, the processor will internally upsample and downsample if necessary. The only legitimate use for recording at 96kHz or higher sample rates is for some sound effects design purposes and for scientific purposes (recording insects or bats for example). However, 96kHz and 192kHz are reasonably common as recording formats for music products because clients may want to market a "hi-res" version at some stage.

G


----------



## Arjey

Does this mean there is absolutely NO difference between 16/48 and 24/48?


----------



## castleofargh

Arjey said:


> Does this mean there is absolutely NO difference between 16/48 and 24/48?


The quantization noise rises to 16bit, and depending on the playback gear, you might see some slight increase in distortions. Both should remain inaudible under typical music listening, and normally do, as suggested by most listening tests.


----------



## gregorio

Arjey said:


> Does this mean there is absolutely NO difference between 16/48 and 24/48?



To elaborate on what castleofargh stated: Yes, there is a difference, 16bit has a 48dB higher noise floor than 24bit, which is defined by dither (noise). This puts the noise floor of 16bit at around -92dB, which is audible under laboratory and mastering studio conditions using very high (uncomfortably loud) listening levels, tests indicate roughly 14dB higher than comfortable levels. However, it's been standard/recommended practice to use noise-shaped dither (rather than standard dither) since before 24bit recording was even available to recording/mastering studios, which reduces the noise floor of 16bit by a further 25dB or so (to roughly -120dB). Therefore, the digital noise floor of 16bit as defined by (noise-shaped) dither is roughly 40dB below what would be audible under laboratory conditions when listening at comfortable levels (and 40dB is 100 times)!

G


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## TheSonicTruth (Jan 6, 2020)

gregorio said:


> To elaborate on what castleofargh stated: Yes, there is a difference, 16bit has a 48dB higher noise floor than 24bit, which is defined by dither (noise). This puts the noise floor of 16bit at around -92dB, which is audible under laboratory and mastering studio conditions using very high (uncomfortably loud) listening levels, tests indicate roughly 14dB higher than comfortable levels. However, it's been standard/recommended practice to use noise-shaped dither (rather than standard dither) since before 24bit recording was even available to recording/mastering studios, which reduces the noise floor of 16bit by a further 25dB or so (to roughly -120dB). Therefore, the digital noise floor of 16bit as defined by (noise-shaped) dither is roughly 40dB below what would be audible under laboratory conditions when listening at comfortable levels (and 40dB is 100 times)!
> 
> G



But obviously there is some benefit to using higher specs on the production side, or else they would not have bothered to start doing it, right?


One will more quickly raise the noise floor if the entire production process - RMM(Recording sessions, Mixing, Mastering) is done at 16/44.1 than at 24/92 or higher before exporting to consumer formats, correct?

Sort of analogous to a wedding photographer capturing the event portraits at the highest raw megapixels he can afford, and his equipment is capable of, _before_ any touch-up processing is done and smaller 8x10s, 5x7s, or wallet-sized copies of the special day's photos  are produced.


----------



## bigshot (Jan 6, 2020)

Arjey said:


> Does this mean there is absolutely NO difference between 16/48 and 24/48?



No audible difference for the purposes of playing back recorded music in the home. The differences are purely technical and would have no practical application for how you would be using music files.


----------



## 71 dB

TheSonicTruth said:


> But obviously there is some benefit to using higher specs on the production side, or else they would not have bothered to start doing it, right?



On the production side the benefit is you don't need to optimize the dynamic range in everything you do and you know the noise floor won't explode to your face no matter how much you process the sound. ===> *Clear benefits on the production side, but no benefits on consumer side (apart from "hi-res" placebo effect).*


----------



## 71 dB

castleofargh said:


> The quantization noise rises to 16bit, .



If one is to create a 16 bit version of a 24 bit file it would be "dumb" to just truncate the 24 bit data to 16 bit and get 16 bit quantization noise. Instead people use dither noise and instead 16 bit quantization noise they get 16 bit dither noise and these two are very different things. Quantization noise _correlates_ with the signal and causes non-linear distortion. Dither noise doesn't correlate with the signal at all and avoids non-linear distortion. Even having more power, dither noise appears less annoying to human ear and shaping the spectrum of dither can make it spychoacoustically much quieter. In fact shaped dither can achieve perceived dynamic range so large that one can call such 16 bit consumer audio total overkill by several bits.

We should also remember that producing music at larger than 100 dB dynamic range is not trivial. Good luck recording a singer at 120 dB dynamic range without noise! The recording boot is not noise free. The microphone pre-amp is not noise free. There will be noise significantly above the 24 bit quantization noise. Ironically this noise acts as dither and helps keeping things linear and distortion free. So, the 24 bit recording may have a noise floor at level not much below 16 bit dither meaning going from 24 bit to 16 bit doesn't mean increasing noise by 48 dB. Using 24 bit allows flexible use of the bits. For example, the four highest bits can be "allocated" to 24 dB headroom while the lowest 4 bits ensure the quantization happens well below the noise floor of the whole recording system.


----------



## gregorio

TheSonicTruth said:


> [1] But obviously there is some benefit to using higher specs on the production side, or else they would not have bothered to start doing it, right?
> [2] One will more quickly raise the noise floor if the entire production process - RMM(Recording sessions, Mixing, Mastering) is done at 16/44.1 than at 24/92 or higher before exporting to consumer formats, correct?



1. On the recording, mixing and mastering side yes, there is some benefit, on the reproduction side, no audible benefit at all. For example, during recording 24bit allows for plenty of headroom, which is useful because we obviously don't know what the peak level is going to be before the musician has actually performed but we do know exactly what the peak level of the master is, so we don't need any headroom.

2. If the mixing and mastering were done at 16bit the noise floor would theoretically increase more quickly than if it were done at 24bit but even at 24bit it would increase unacceptably, which is why professional mixers have never been 24bit devices (let alone 16bit). Even the early ones were about 28bit but by the time there were widely used they were 32bit float or 48bit, as I've already explained just a few posts previously.

G


----------



## 71 dB

Arjey said:


> Does this mean there is absolutely NO difference between 16/48 and 24/48?



For consumers placebo effect is all "difference" there is. People may think 24/48 sounds better than 16/48 because intuition says more bits must mean better fidelity, but 16 bits is already a few bits more than we actually need in consumer audio (the threshold is ~13 bits).


----------



## Arjey

So, if I have a bunch of music in 24 bit, and I only use it for listening, no mixing or anything.. Is it safe to say that a can just convert everything to 16 bit (to save some space) and I won't hear a difference?

And a different question.
24 bit can go up to 192khz. I know most people don't here even over 15khz. I personally can hear up to ≈19khz (I'm pretty young, 19 y.o., maybe that's why I hear so high frequencies and hate sibilance).
Then there are low frequencies, people don't really hear 1-4hz as much as feel the vibration.
So my question is, say I have Really good headphones, a Really good source, and a Really good recording.. will I be able to "feel" the difference between 48 vs 192khz? I just want it to sound as natural and realistic as possible.

The real world has unlimited (almost) frequencies, thought we can't even hear them.
But for example: dog whistles. We can't here them, but there where a couple of time I was walking with a friend in a park with his dog, and I was at a distance, and he just blew the whistle (I was looking in a different direction), but for some reason I just know that "something" is different, and I look in that direction. And there he is, blowing a whistle that I can't hear.
Something like a "sixth sense"..
So what I'm asking is, will converting a 24/192 or DSD to 16/48 "remove" that slight sixth sense of airyness or realism?

Sorry for the weird way of putting this question, to which you guys most likely don't have an answer, because it's absurd, and the answer will most likely be: "depends on what you believe.." but still. I think is an interesting thing to think about)


----------



## 71 dB (Jan 6, 2020)

Arjey said:


> 1 --- So, if I have a bunch of music in 24 bit, and I only use it for listening, no mixing or anything.. Is it safe to say that a can just convert everything to 16 bit (to save some space) and I won't hear a difference?
> 
> 2 --- And a different question.
> 24 bit can go up to 192khz. I know most people don't here even over 15khz. I personally can hear up to ≈19khz (I'm pretty young, 19 y.o., maybe that's why I hear so high frequencies and hate sibilance).
> ...



1 --- Yes. You can even convince yourself about it by producing that 16 bit version and then upload both the 24 bit version and the 16 bit version on a wave editor such as Audacity, invert the 16 bit version and mix it with the 24 bit version so that you get the difference (make sure your result is 24 bit or higher) and the try to hear the result. You'll notice at you hear nothing unless you turn the volume up crazy and that is the difference signal only. It is totally masked when you listen to the music! That's when you go "of course I can't hear the difference between 16 bit and 24 bit.

2 --- People can only _sense_ low frequencies (such as 1-4 Hz) but around 16 Hz starts the sensation of sound (combination of both). Sampling rate affects the upper limit of frequency range so the difference between 48 kHz and 192 kHz sampling rates is above human hearing range and doesn't matter. 48 kHz is enough and so is 44.1 kHz. The truth is 192 kHz sampling rate is totally pointless in consumer audio. If you study the ultrasonics of bats then you need 192 kHz, but music listening is not about bat science. Whatever you "feel" between 48 kHz and 192 kHz is probably due to placebo effect that affects your perception because you "believe" 192 kHz must sound better. When you have the knowledge of digital audio and what is enough you'll be able to remove the effect of placebo.

3 --- Maybe you knew "something" is different, because you heard the change in the behavior of the dog who did hear the whistle? Maybe dog whistles produce some sound in the human hearing range, say in the range of 10-20 kHz? I have to admit I don't know much about dog whistles. Listening tests suggests that downsampling hi-res audio shouldn't remove anything relevant to human hearing.

According to Wikipedia "To human ears, a dog whistle makes only a quiet hissing sound." and  "some [dog whistles] are adjustable down into the audible range."


----------



## Arjey

71 dB said:


> 1 --- Yes. You can even convince yourself about it by producing that 16 bit version and then upload both the 24 bit version and the 16 bit version on a wave editor such as Audacity, invert the 16 bit version and mix it with the 24 bit version so that you get the difference (make sure your result is 24 bit or higher) and the try to hear the result. You'll notice at you hear nothing unless you turn the volume up crazy and that is the difference signal only. It is totally masked when you listen to the music! That's when you go "of course I can't hear the difference between 16 bit and 24 bit.
> 
> 2 --- People can only _sense_ low frequencies (such as 1-4 Hz) but around 16 Hz starts the sensation of sound (combination of both). Sampling rate affects the upper limit of frequency range so the difference between 48 kHz and 192 kHz sampling rates is above human hearing range and doesn't matter. 48 kHz is enough and so is 44.1 kHz. The truth is 192 kHz sampling rate is totally pointless in consumer audio. If you study the ultrasonics of bats then you need 192 kHz, but music listening is not about bat science. Whatever you "feel" between 48 kHz and 192 kHz is probably due to placebo effect that affects your perception because you "believe" 192 kHz must sound better. When you have the knowledge of digital audio and what is enough you'll be able to remove the effect of placebo.
> 
> ...


Thanks for the detailed reply! I guess I'll just stick with 48khz, though I don't really hear a difference between 48 and 44.1, but just to be safe) Also most films and many games use 48, and it's easier on the system (if it's old, at least) to keep playing the same sample rate.
Here's some interesting and detailed material on the matter. https://forums.stevehoffman.tv/threads/longish-why-i-prefer-48-khz-over-44-1-khz.347515/


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## old tech (Jan 6, 2020)

Arjey said:


> Thanks for the detailed reply! I guess I'll just stick with 48khz, though I don't really hear a difference between 48 and 44.1, but just to be safe) Also most films and many games use 48, and it's easier on the system (if it's old, at least) to keep playing the same sample rate.
> Here's some interesting and detailed material on the matter. https://forums.stevehoffman.tv/threads/longish-why-i-prefer-48-khz-over-44-1-khz.347515/


I like the SH forum, mainly for finding consensus on the best masterings of a CD or LP.

There are some very knowledgeable people who post in that forum but also a lot of misinformed woo, that 48 v 44 thread is a case in point.  A year ago there was a thread comparing 44.1 vs hi res (ie anything from 24 bits to DSD) and the overwhelming response was that no-one could hear a difference if the same master was used under blind level matched tests.  What was surprising was that the thread ended up being over 5 pages of good rational discussion but the mods then deleted the entire thread - probably because Steve Hoffman is trying to break back into the industry with AF SACDs, (which is based on a lie that the format itself sounds better than CDs).  They lost a lot of good members after that thread deletion.

That forum also seems to attract a lot of vinylphiles, digiphobes and the downright irrational.  So take care with information from that site - check the info with the more evidence based sound science forums such as this sub forum or at https://hydrogenaud.io/. Another good resource which contains many blind tests and discussions is at http://archimago.blogspot.com/

With your point about the dog whistle, bare in mind that they vary quite a bit in frequencies - some start at 20khz and others go up to 23 to 25 khz.  So it is possible that you could have heard it but I tend to agree more with 71 DB, that your perception was subconsciously more around the reaction of the dog and/or a fleeting sight of the whistleblower in your periphery. You really need a controlled test to be sure.


----------



## bigshot (Jan 6, 2020)

I find that every forum has nut jobs (even this one!) You just have to read what people say and look for the people who seem to have practical experience rather than the armchair experts who speak purely in theory. It also helps to figure out whether someone is talking about something they've experienced rather than just repeating common knowledge. If you make note of the good posters and what their specialty is, you can glean information from posts even if your own subjective taste is different than theirs.

But some forums actively promote the nut jobs. I've found a few of those and I don't spend much time there. The forum mentioned falls into that category for me.


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## TheSonicTruth (Jan 6, 2020)

gregorio said:


> which is why professional mixers have never been 24bit devices (let alone 16bit).
> Even the early ones were about 28bit but by the time there were widely used they
> were 32bit float or 48bit, as I've already explained just a few posts previously.



AHHH... So you were saying that even 35-ought years ago, they had _higher_ than 24bit mixers.  I didn't pick up that inference the first time.  Well what about recording?  Still stuck with 16/44.1 for sessions?  Then even a 64bit mixer wouldn't do a whole lot of good.  Kind of like importing a 128k mp3 and exporting a 320k mp3 from it.  Not very useful.


----------



## old tech (Jan 6, 2020)

TheSonicTruth said:


> But obviously there is some benefit to using higher specs on the production side, or else they would not have bothered to start doing it, right?
> 
> 
> One will more quickly raise the noise floor if the entire production process - RMM(Recording sessions, Mixing, Mastering) is done at 16/44.1 than at 24/92 or higher before exporting to consumer formats, correct?
> ...


I wonder whether you see the irony in your preference for the sound quality of 1980s CDs.


----------



## TheSonicTruth (Jan 6, 2020)

old tech said:


> I wonder whether you see the irony in your preference for the sound
> quality of 1980s CDs.



I wonder if you(and not to mention countless others) see the irony in comparing the sound quality of 1980s CDs to cassettes or 78rpm records.  Tired of hearing that schitt, seriously.


----------



## old tech

TheSonicTruth said:


> I wonder if you(and not to mention countless others) see the irony in comparing the sound quality of 1980s CDs to cassettes or 78rpm records.  Tired of hearing that schitt, seriously.


I actually prefer the sound quality of most CDs made in the 1980s to their contemporary counterpart.  The irony is that those later "remastered" CDs which you constantly criticise are the product of 24 bit processing.  The flat transfers which feature on most 1980s CDs are the product of 16 bit and analog workstation limitations.


----------



## TheSonicTruth

old tech said:


> I actually prefer the sound quality of most CDs made in the 1980s to their
> contemporary counterpart.  The irony is that those later "remastered" CDs which you
> constantly criticise are the product of 24 bit processing.  The flat transfers which feature on
> most 1980s CDs are the product of 16 bit and analog workstation limitations.



Well, it's unfortunate that we'll never get to experience those flat transfers done at 24bit.  But at least the '80s era CD releases are closer to the original by virtue of being mostly flat transfers.  The CDs made from 24bit processing sadly suffered the final step: the brickwall limiting that just made them louder thsn the 80s transfers.


----------



## gregorio (Jan 7, 2020)

Arjey said:


> [1] So, if I have a bunch of music in 24 bit, and I only use it for listening, no mixing or anything.. Is it safe to say that a can just convert everything to 16 bit (to save some space) and I won't hear a difference?
> [2] And a different question. 24 bit can go up to 192khz. ...So my question is, say I have Really good headphones, a Really good source, and a Really good recording.. will I be able to "feel" the difference between 48 vs 192khz? I just want it to sound as natural and realistic as possible.
> [3] The real world has unlimited (almost) frequencies, thought we can't even hear them.
> [3a] But for example: dog whistles. We can't here them, but there where a couple of time I was walking with a friend in a park with his dog, and I was at a distance, and he just blew the whistle (I was looking in a different direction), but for some reason I just know that "something" is different, and I look in that direction.



1. True.

2. Higher sample rates only make a difference in capturing very high freqs, for very low freqs they make no difference. You won't be able to hear or feel the difference between 48 and 192kHz with really good headphones. There's a chance you might if using not so good headphones though, because they're liable to produce distortion in the audible range in response to very high/ultrasonic frequency content (this is called IMD, Inter-Modulation Distortion).

3. But we're not dealing with the real world, we're dealing with the music world. Musical instruments have been designed by humans for human hearing and produce relatively little or in some cases no freqs beyond 20kHz. There are some exceptions, mainly metalophones like gamalan or glockenspiel for example but at typical audience listening distances they still have relatively little ultrasonic content and have not been reliably differentiated with 48 vs 192kHz recordings.
3a. As mentioned, some dog whistles aren't quite ultrasonic, they're just within the limits of human hearing baring in mind they output extremely high sound pressure levels. Even the truly ultrasonic dog whistles can produce distortion just within the limits of human hearing when blown hard.



TheSonicTruth said:


> [1] AHHH... So you were saying that even 35-ought years ago, they had _higher_ than 24bit mixers.  I didn't pick up that inference the first time.
> [2] Well what about recording?  Still stuck with 16/44.1 for sessions?
> [2a] Then even a 64bit mixer wouldn't do a whole lot of good.  Kind of like importing a 128k mp3 and exporting a 320k mp3 from it.  Not very useful.



1. The first widely used digital mixer was introduced in 1987 (Yamaha DMP7), I'm not sure what internal processing it had but believe it was more than 24bit. However, it wasn't used in the music recording industry, it was used in live sound/music reinforcement and broadcast. This was due to the functionality it offered (total recall, flying faders, automation, etc.) over analogue desks, not the sound quality, which wasn't great. Around 1990 Yamaha introduced the DMC1000, which did have good sound quality and was occasionally used in the music recording industry but still mainly in broadcast and live sound. Trevor Horn used one on Seal's first album (Crazy, Killer, etc.) but I don't recall if he used it exclusively, certainly digital mixing wasn't used for mastering until well into the millennium. The DMC1000 had 28bit internal processing, later (mid 1990s) digital desks used in music recording were 32bit float, Sony's Oxford desk for example.

2. I've already said! It was mainly 16/44.1 for recording up until the millennium, although 20bit became more widespread starting in the early 1990's up until 24bit around the millennium.
2a. No, you're not getting it, even though you yourself brought it up! Sure, there's no benefit whatsoever of importing 16/44 into a 64bit mixer but then the whole point of a digital mixer is to mix, not just to import! And as you mentioned, each and every processor within a digital mixer adds quantisation noise which accumulates. Baring in mind the average rock song (and all other non-acoustic popular genres) employs dozens of processes/processors, the quantisation noise would become unacceptable, regardless of whether the original tracking is at 16bit or 24bit. 64bit processing allows thousands of processes before the accumulated quantisation noise would become audible, which covers any eventuality.



old tech said:


> [1] The irony is that those later "remastered" CDs which you constantly criticise are the product of 24 bit processing.
> [2] The flat transfers which feature on most 1980s CDs are the product of 16 bit and analog workstation limitations.



1. As I've mentioned, "those later remastered CDs" were not the product of 24bit processing, as there's never been 24bit processing used in commercial music production to my knowledge. It was the product of 32bit float or 48bit fixed processing.

2. To clarify, with the exception of some/a few classical recordings, the process was: Recording on multi-track tape, constantly replaying that tape out to an analogue desk (with "outboard" gear) where it was mixed. When all the desk's (and outboard gear's) parameters we adjusted to produce the desired mix, the result was "bounced" (recorded) back down to tape. There was no such thing as an analog workstation. Digital recording did not change this workflow at all, it was still multi-track tape, constantly replayed out to an analogue desk, etc. The only difference was that the multi-track tape recorded digital audio (and then replayed it through the recorder's DACs) rather than analogue audio. The only workflow change was in editing, as digital audio tape couldn't be spliced and was a rather more involved and time consuming process. Digital workstations started to be used in the mid 1990s but not as workstations, they we're used for editing because they massively reduced the editing time, as well at it being more accurate and non-destructive (and bit depth is irrelevant to editing as there's no processing involved). It wasn't until the very end of the 1990's that they started being used as workstations (IE. For recording, editing and mixing), the first No.1 done this way was Ricky Martin's "Living La Vida Loca" in 1999, although outboard analogue gear was still employed and the mastering was still analogue. Fully ITB (In The Box, no analogue outboard gear) didn't start really taking over in the commercial music world until the mid 2000's, quite a few years after 32bit or 48bit fixed mix environments/processing was standard, with mastering being the last bastion to hold out for a few years more.

Baring all the above in mind, I don't really understand what you mean by "flat transfers" or what TheSonicTruth is trying to say with his response?

G


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## TheSonicTruth (Jan 7, 2020)

gregorio said:


> introduced in 1987 (Yamaha DMP7), I'm not sure what internal processing
> it had but believe it was more than 24bit. However, it wasn't used in the
> music recording industry, it was used in live sound/music reinforcement and broadcast.
> This was due to the functionality it offered (total recall, flying faders, automation, etc.)
> over analogue desks, not the sound quality, which wasn't great.



So what about these early digital desks was behind such 'not so great' sound quality?  Converters of the day?




gregorio said:


> Digital workstations started to be used in the mid 1990s but not as
> workstations, they we're used for editing because they massively reduced the
> editing time, as well at it being more accurate and non-destructive (and bit depth
> is irrelevant to editing as there's no processing involved). It wasn't until the very
> ...



Ahh, so it's songs like that which misled the public that that digital formats themselves were behind such super-compressed & loud releases!  They failed to realize that it was the human processing, not the format.




gregorio said:


> Baring all the above in mind, I don't really understand what you mean by "flat
> transfers" or what TheSonicTruth is trying to say with his response?



Isn't it common knowledge that the very earliest CD releases(1980 up to about 85-86) - the pop ones anyway, not jazz or classical - were produced straight from the same stereo masters used for LP records, minus the RIAA treatment which was applied later on at the vinyl cutting lathe, and with little or no additional processing(EQ, comp, etc)?

And what did you not understand from my response in post #5526?


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## TheSonicTruth (Jan 7, 2020)

gregorio said:


> 2. I've already said! It was mainly 16/44.1 for recording up until the millennium,
> although 20bit became more widespread starting in the early 1990's up until
> 24bit around the millennium.



So, in my infinite, and admittedly Dyslexic wisdom, I had it completely the wrong way round.  It was the _recording_ sessions(and legacy flat transfers for that matter) that were mainly 16/44.1 up until 2000, and the _mixing desks themselves_ that were 32-float and higher.   Was 24-bit & higher _session recording_ still possible before 1990??  It would have been better, and I'm sure it was available.  My take is that the barriers to wiDespread adoption of it were, thirty years ago, the usual suspects: storage space, and affordability.


For one, I would love to have owned a CD issue of 'THRILLER' or any of the late-70s Roth-era Van Halen albums made from 24- or higher - bit 96k sampling transfers of those original stereo masters!  The same for James Taylor, Donna Summer, Earth Wind & Fire, Fleetwood Mac, and so on.  If only....


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## pinnahertz

TheSonicTruth said:


> Was 24-bit & higher _session recording_ still possible before 1990??


"Possible"?  Yes.  Practical and done in practice?  No. 


TheSonicTruth said:


> It would have been better, and I'm sure it was available.


I'm sure you are sure of both.  But you're also wrong on both counts. 
24 bit quantization, in the real world, doesn't improve resolution at all.  The "steps" aren't smaller because they aren't there in the first place.   All more bits can ever do is result in a lower noise floor, but at -144dBFS theoretical, it's below the practical noise floor of every other device in the system including the ADC's own input stage.  In 2020 there remain no ADCs with true 24 bit noise floors, nor mics, preamps, and most importantly, rooms, with noise floors even anywhere close, typically worse than 16 bits.


TheSonicTruth said:


> My take is that the barriers to wisespread adoption of it were, thirty years ago, the usual suspects: storage space, and affordability.


We recorded digits on tape back then.  Studio-grade recorders were most often DASH machines, two and multi track, and always 16/48 with the highest rate being 50k, mostly not used because resampling to 44.1 for CD release raised the noise floor 3dB.  

The typical biggie HDD was 200mB.  For computer -based editing, it took a stack of Ultra SCSI Fast/Wide drives to edit a project, and there was only enough room for the raw good takes plus fade files used in cross-fade edits, and only one project in the machine at a time.  Early computer editing (like the Studer Dyaxis) were sluggish, pricy, temperamental beasts.  I watched a ProTools demo in the early 1990s that showed a 1 second cross-fade taking 22 minutes to render.  Editing was a major pain.


TheSonicTruth said:


> For one, I would love to have owned a CD issue of 'THRILLER' or any of the late-70s Roth-era Van Halen albums made from 24- or higher - bit 96k sampling transfers of those original stereo masters!  The same for James Taylor, Donna Summer, Earth Wind & Fire, Fleetwood Mac, and so on.  If only....


They'd sound exactly the same as what you have now.  None of that is demanding material at all, and none comes even close to needing 16 bits. 

Nice to see things don't change here.  All the time I've been away, and it's the same arguments by the same people ignoring the same experts.


----------



## gregorio

TheSonicTruth said:


> [1] So what about these early digital desks was behind such 'not so great' sound quality?  Converters of the day?
> [2] Ahh, so it's songs like that which misled the public that that digital formats themselves were behind such super-compressed & loud releases!  [2a] They failed to realize that it was the human processing, not the format.
> [3] Isn't it common knowledge that the very earliest CD releases(1980 up to about 85-86) - the pop ones anyway, not jazz or classical - were produced straight from the same stereo masters used for LP records, minus the RIAA treatment which was applied later on at the vinyl cutting lathe, and with little or no additional processing(EQ, comp, etc)?
> [4] And what did you not understand from my response in post #5526?



1. I don't know for certain but an educated guess would be not so much the "converters of the day" but more the sophistication of the algorithms employed, both in terms of the limited amount of DSP power available at the time and in terms of the programming knowledge to make the most efficient use of it. The DMC1000 had many times the DSP capabilities of the DMP7 and was developed with the help of engineers from Deutsche Grammophon (who had been using their own custom built, very simple but good SQ digital mixers since about the release of CD for their classical releases).

2. I'm not sure about that. The really hyper-compressed releases started a few years before then.
2a. True, that always been the case.

3. I'm not sure that is the case. I'm sure it was sometimes but probably not all the time and much more so in recent times. LP cutting requires certain other conditions besides just the application of the RIAA curve.

4. What you mean by "flat transfers done at 24bit" and "CD's made from 24bit processing". 



TheSonicTruth said:


> [1] So, in my infinite, and admittedly Dyslexic wisdom, I had it completely the wrong way round. It was the _recording_ sessions that were mainly 16/44.1 up until 2000, and the _mixing desks themselves_ that were 32-float and higher.
> [2] Was 24-bit & higher _session recording_ still possible before 1990?? It would have been better, and I'm sure it was available.
> [3] My take is that the barriers to wisespread adoption of it were, thirty years ago, the usual suspects: storage space, and affordability.



1. Correct, although bare in mind that the vast majority of non-acoustic music recordings were recorded in 16/44 but mixed in analog up until 2000 (and even somewhat beyond). Digital mixing desks in the 1990s were all higher than 24bit but were not as widely used in the commercial music as analog desks.

2. No, it wasn't. The first commercial 24bit recording available as far as I recall was Pro Tools in 1997 (which had 48bit internal processing) but the converters weren't that good, better 24bit converters (from Apogee and Prism) became available a couple of years later and were common around the millennium. The first 20bit recording was Yamaha's DMR8 released in 1990 which was still tape based, although other Japanese manufacturers (Mitsubishi, Otari) released optical disk based 20bit recorders around the same time or a year or two later. Again though, these weren't really widely used, the commercial music industry largely stuck with 16bit Sony DASH and Otari until they too did 24bit versions in mid/late 1990's. I don't remember the exact time line and might be out by a couple of years here and there but there certainly wasn't commercial music 24bit recording before the mid/late 1990's.

3. Again, not so much. The Sony 3348 DASH (16bit, tape based) was used by many/most of the world's top commercial music studios in the late '80's, it weighed several hundred pounds, cost about $300,000 and it's main competitors (mainly Mitsubishi) were about the same. On top of that, the desks (analog or digital) were typically $300,000 - $750,000 so affordability wasn't much of an issue. When it came to DAWs, the only game in town commercially was Pro Tools and although storage space was an issue, the bigger issue was transfer speeds of HDs and available DSP power. To have usable track counts for 24/44.1 or 24/48 at the turn of the millennium required arrays of the latest SCSI standard HDs and multiple hardware DSP cards.

G


----------



## gregorio

Hey @pinnahertz I didn't see your reply before I posted mine, great to see you back here! Professional knowledge and expertise is a bit thin on the ground around here these days.

G


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## TheSonicTruth

pinnahertz said:


> "Possible"?  Yes.  Practical and done in practice?  No.
> I'm sure you are sure of both.  But you're also wrong on both counts.
> 24 bit quantization, in the real world, doesn't improve resolution at all.
> The "steps" aren't smaller because they aren't there in the first place.
> ...



Well, my 'argument' for higher specs is from a production(including recording sessions) context, not from one of the end-listener formats.  I apologize if my argument came off otherwise - a la audiophile, etc.


----------



## TheSonicTruth

gregorio said:


> LP cutting requires certain other conditions besides just the
> application of the RIAA curve.



Such as bottom end roll-off, and the center of remaining bass elements?  Of course!  But those should have no impact on the sound as heard on CD, other than just a little bass-shy.




gregorio said:


> 4. What you mean by "flat transfers done at 24bit" and "CD's
> made from 24bit processing".



Simple:  Recording of a band, or in the case of CD from legacy,  transfer of original master tapes to digital at higher than Red Book specs. mixing(in the case of the band's session tracks), and mastering, all at higher than Red Book.  Then dither down to 16/44.1 Red Book.


----------



## bigshot

Welcome back Pinnahertz!


----------



## pinnahertz

TheSonicTruth said:


> Well, my 'argument' for higher specs is from a production(including recording sessions) context, not from one of the end-listener formats.  I apologize if my argument came off otherwise - a la audiophile, etc.


It doesn't matter.  The recording process was 16/44.1/48/50 for the first two decades of digital audio, and it wasn't a limiting factor.  The practical limiting factors in "resolution" are all outside of quantization, and way outside quantization at 24 bits.  Until digital desks with adequate DSP arrived, most post was done in the analog world, then mixed back to 16/44.1.  

The one quality limiter of all PCM in the early days (technically still outside of quantization) was analog anti-aliasing and reconstruction filters, the earliest of which suffered from nonlinearities in the cutoff zone which resulted in intermodulation products being folded down into the mid-audible band.  Once oversampling filters arrived, the problem was mostly mitigated.  Increasing sampling frequency was initially viewed as the solution, but all that could do was relocate the problem.  The real work was on the filters, and there was a cafe of exotic retrofits for every major recording device.  All were compromises, trading one filter quality for another, nobody every made the perfect analog filter.  You could reduce intermod, but then raise aliasing, for example. Until we got to oversampling digital filters, that was a rough spot.  

24 bits couldn't help the intermod issue then, and still wouldn't.  

Again, 24bits does not increase resolution or reduce distortion, it lowers quantization noise only.  But system noise is already higher than LSB jitter in 24 bits by many, many dB, so it's a wash. 

Oh yeah, 24 bits does provide one very significant advantage that all of us in the industry recognize: higher numbers are better, and more bits makes everyone feel warmer and fuzzier.


----------



## pinnahertz

gregorio said:


> Hey @pinnahertz I didn't see your reply before I posted mine, great to see you back here! Professional knowledge and expertise is a bit thin on the ground around here these days.
> 
> G





bigshot said:


> Welcome back Pinnahertz!



Thanks guys!


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## 71 dB

Arjey said:


> Thanks for the detailed reply! I guess I'll just stick with 48khz, though I don't really hear a difference between 48 and 44.1, but just to be safe) Also most films and many games use 48, and it's easier on the system (if it's old, at least) to keep playing the same sample rate.
> Here's some interesting and detailed material on the matter. https://forums.stevehoffman.tv/threads/longish-why-i-prefer-48-khz-over-44-1-khz.347515/



48 kHz is totally fine as a "fixed" samplerate for all material. The "optimal" samplerate would be something like 56 kHz (because it allows somewhat relaxed anti-aliasing and reconstruction filters). If we can live with steep filters 44.1 kHz is just enough.


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## 71 dB

pinnahertz said:


> They'd sound exactly the same as what you have now.  None of that is demanding material at all, and none comes even close to needing 16 bits.
> 
> Nice to see things don't change here.  All the time I've been away, and it's the same arguments by the same people ignoring the same experts.



Turns out the 16 bit vs 24 bit myth can't be exploded… …it is too persistant! 

Surprised to see you back here.


----------



## pinnahertz

71 dB said:


> 48 kHz is totally fine as a "fixed" samplerate for all material. The "optimal" samplerate would be something like 56 kHz (because it allows somewhat relaxed anti-aliasing and reconstruction filters). If we can live with steep filters 44.1 kHz is just enough.


I wouldn't classify filter quality based on cutoff slope alone.  Much more to it than just that.  But yeah, higher sampling mitigates the filter.  I just don't think it makes any audible difference if the filters are clean to begin with.


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## pinnahertz

71 dB said:


> Turns out the 16 bit vs 24 bit myth can't be exploded… …it is too persistant!
> 
> Surprised to see you back here.


Ha ha!  It's only the people that are persistent. 
Thanks.


----------



## 71 dB

pinnahertz said:


> 1 --- I wouldn't classify filter quality based on cutoff slope alone.
> 
> 2 --- I just don't think it makes any audible difference if the filters are clean to begin with.



1 --- Of course not. What I mean here is higher sample rate gives more freedom for the shape of the cutoff slope and at around 56 kHz sample rate the benefits of this freedom are pretty much exhausted.

2 --- Well, I can hear very minor changes in sound (spatial width) when changing the reconstruction filter of my NAD C565BEE CD-player, It's kind of like moving the speakers a few inches closer or away from each other. The standard filter sounds most narrow while the other options gives a little bit wider soundstage of various degree. I also think I can just hear a tiny change in the amount of the highest frequencies, but this might be placebo or just me misinterpreting the change in spatial width as spectral change. All filters give equally high fidelity in my opinion, just a little bit different ways (moving the speakers an inch or two makes similar changes). So, in my opinion at 44.1 kHz the type of reconstruction filter does make an audible difference, but does it matter? Not really if you ask me. If your gear has only one fixed filter your ears will get used to it.


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## 71 dB

pinnahertz said:


> Ha ha!  It's only the people that are persistent.
> Thanks.



And it's not only this issue. The internet is about insisting one's beliefs ad nauseum and I am one to blame. 
I'm trying to learn to become a happier and better person, but it's a long process...


----------



## old tech

gregorio said:


> 1. As I've mentioned, "those later remastered CDs" were not the product of 24bit processing, as there's never been 24bit processing used in commercial music production to my knowledge. It was the product of 32bit float or 48bit fixed processing.



The point is the same though, 32bit or 48 bit float (or even 24 bit) would enable far more processing of the recording, pushing the sound wars into the stratosphere.  What I was trying to say is that there was a limit to how much compression etc when processing power was limited to analog equipment or 16 bit digital.  In other words, the greater processing power could and is (in many cases) abused.  CDs with DR2 for anyone?



gregorio said:


> 2. To clarify, with the exception of some/a few classical recordings, the process was: Recording on multi-track tape, constantly replaying that tape out to an analogue desk (with "outboard" gear) where it was mixed. When all the desk's (and outboard gear's) parameters we adjusted to produce the desired mix, the result was "bounced" (recorded) back down to tape. There was no such thing as an analog workstation. Digital recording did not change this workflow at all, it was still multi-track tape, constantly replayed out to an analogue desk, etc. The only difference was that the multi-track tape recorded digital audio (and then replayed it through the recorder's DACs) rather than analogue audio. The only workflow change was in editing, as digital audio tape couldn't be spliced and was a rather more involved and time consuming process. Digital workstations started to be used in the mid 1990s but not as workstations, they we're used for editing because they massively reduced the editing time, as well at it being more accurate and non-destructive (and bit depth is irrelevant to editing as there's no processing involved). It wasn't until the very end of the 1990's that they started being used as workstations (IE. For recording, editing and mixing), the first No.1 done this way was Ricky Martin's "Living La Vida Loca" in 1999, although outboard analogue gear was still employed and the mastering was still analogue. Fully ITB (In The Box, no analogue outboard gear) didn't start really taking over in the commercial music world until the mid 2000's, quite a few years after 32bit or 48bit fixed mix environments/processing was standard, with mastering being the last bastion to hold out for a few years more.
> 
> *Baring all the above in mind, I don't really understand what you mean by "flat transfers" or what TheSonicTruth is trying to say with his response?*
> 
> G



The recording process you describe still provides limitations on the degree of processing, ala the soundness wars.  However, I think we were mainly alluding to the reissue of back catalogs when CD sales started to take off in the early to mid 80s.  Of course I am no expert but back in the day a close friend of mine worked at Alberts and Disctronics and many of these releases were a flat transfer of whatever analog production tape was available.  Depending on the quality of the tape (some were great and others quite poor (condition and/or generation wise) which I think accounts for the large variability in sound quality of many 80s back catalogue CDs.  What I (and Sonic Truth, I think) were getting at is that many or most of these flat transfer CDs subjectively sound better than their later remasters simply because they were a flat transfer without any futzing or attempt to make them louder and more compressed/limited.[/QUOTE]


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## bigshot

Whether or not a remastered album sounds better or not depends more on how the processing is applied, than it does whether processing has been applied at all. Bad remastering will result in a worse sounding album. Good processing would result in a better sounding album. Compression and noise reduction have existed since the late 1930s, and I have 78s that have been over compressed and over scrubbed. It's nothing new. However, the tools used to compress and reduce noise have gotten much better since the 1980s. Sure, they can still be used to blunt and mangle music. But used correctly and with good taste, they can vastly improve old analogue masters. Mixing and mastering is a subjective process, and it serves different purposes with different formats and audiences. I've said this a million times to Sonic Truth (and he just doesn't hear what I am saying)... There is good remastering and bad remastering. You can't generalize that most remastering is bad and most 80s CDs are better. There are too many examples of just the opposite being true. You have to judge albums on a case by case basis. The words "24 Bit Remastering" are nothing more than sales pitch. The phrase doesn't indicate whether the album sounds better, worse or exactly the same. You have to listen to it to find that out.


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## TheSonicTruth (Jan 7, 2020)

old tech said:


> What I (and Sonic Truth, I think) were getting at is that many or most of these
> flat transfer CDs subjectively sound better than their later remasters simply
> because they were a flat transfer without any futzing or attempt to make them
> louder and more compressed/limited.




THANK YOU, ot!  Glad I am not alone in this regard, nor completely nuts, lol!

Not always "better", in the case of those first-issue CDs, but: generations _closer _to the sound of the original musical intent of the given album and its artists.  As I keep repeating on here, alas to deaf ears, but "better" is purely subjective, from a audio perspective. Don't get me wrong: take two identical make/model# stereo speakers, and the one with intact driver surrounds will definitely sound _better_ than the one with torn or rotted away surrounds.  Common sense! But that's not what we're on about, here.

Back to point: Sadly, many of those first-issue era CDs did suffer from some of the distortions Gregorio mentioned a page back, post #5531 in this conversation, something all the recording bits and stratospheric sampling rates on Earth would have hardly put a dent in, noise-wise, but still, I prefer them to those reissues from during the 'Remaster-Mania' Era(Late 1990s-2000s).


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## bigshot (Jan 7, 2020)

I've got lots of CDs from the 1990s and 2000s that sound better than they ever sounded before. I don't know how you can generalize unless you're specifically speaking about albums that were poorly remastered. There were plenty of those too. They were pumping out lots of titles back then... some good, some bad.


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## old tech (Jan 7, 2020)

bigshot said:


> Whether or not a remastered album sounds better or not depends more on how the processing is applied, than it does whether processing has been applied at all. Bad remastering will result in a worse sounding album. Good processing would result in a better sounding album. Compression and noise reduction have existed since the late 1930s, and I have 78s that have been over compressed and over scrubbed. It's nothing new. However, the tools used to compress and reduce noise have gotten much better since the 1980s. Sure, they can still be used to blunt and mangle music. But used correctly and with good taste, they can vastly improve old analogue masters. Mixing and mastering is a subjective process, and it serves different purposes with different formats and audiences. I've said this a million times to Sonic Truth (and he just doesn't hear what I am saying)... There is good remastering and bad remastering. You can't generalize that most remastering is bad and most 80s CDs are better. There are too many examples of just the opposite being true. You have to judge albums on a case by case basis. The words "24 Bit Remastering" are nothing more than sales pitch. The phrase doesn't indicate whether the album sounds better, worse or exactly the same. You have to listen to it to find that out.


I don't think we are on different wavelengths... I agree with what you say. I'm not doubting that compression, NR, etc have always existed (and for good reasons), the point is that the original production masters would already have been mastered with it (ie compression, EQ and other 'tricks') that the original producer/artist wanted.  Sure, there are many remasters that do (subjectively) sound better and often they are the albums that were remastered with the aim of improving the original, mitigating their flaws etc rather than just making it loud and brickwalled because that is a modern trend.   Of course it is case by case but generally, mastering has got hotter and more limited/brickwalled since the early 1990s. Hi res processing is neither good or bad, it depends on how it is used. The part we were debating was whether (on balance) the introduction of "hi res" processing has mainly resulted in improvements or a decline in sound quality for the 5% of the population that value sound quality.


----------



## pinnahertz

old tech said:


> The point is the same though, 32bit or 48 bit float (or even 24 bit) would enable far more processing of the recording, pushing the sound wars into the stratosphere.  What I was trying to say is that there was a limit to how much compression etc when processing power was limited to analog equipment or 16 bit digital.  In other words, the greater processing power could and is (in many cases) abused.  CDs with DR2 for anyone?


Hmm.. Well let me relate something about processing that may shed some light.  A good part of my career has been spend in broadcasting where there is a very aggressive version of remastering processing going on in the final processor before transmission.  The idea is quite old, but the tools got pretty intense in the late 1970s, early 1980s, all analog of course.  The goal was to be loud, subjectively clean (the two are opposite vectors), and not go past legal modulation limits.  The degree varied by music type and format, but the general goal is always true.  The loudness war on-air is older than it is in recorded music.  So the analog tools got pretty fancy.  Then suddenly, we had digital processing, with tons more flexibility, stability, and the ability to save and recall massive settings.  You know what the goal was then?  To make that DSP based megaprocessor sound as good as it's analog predecessor.  And it took almost a decade to come even close to that goal.  Now we have the latest units with massive DSP engines, we can build extremely complex processors, and we have the ability to model nonlinear systems.  Yeah, some of the new digital processors are amazing, but I'll tell you, first hand, it's not easy to reach the goal with those tools.  The flexibility works against you.  In fact, there are some rather prominent products that are actually pretty far behind what we did analog 30 years ago, and not because we can't, it's because everybody want's to tweak the latest tools to push for louder and cleaner.  The two are still opposite vectors.  Nothing has changed but the number of ways and degree to which you can mangle the signal.   

So, 48 bit float would enable more processing? I don't disagree, but the result would be no better than what you could do with fewer  (or zero) bits because it's about the algorithm first and foremost, not about bit depth at all.    The most you can hope to gain in the digital realm is for tight and accurate peak control, and even then only if you fully include the entire system in the algorithm, even if it is predictive.  All of this is because 90% of all dynamics processing happens above -15dBFS where we have all bits active already.  Remember, higher bit depth only lowers quantization noise, it does not increase resolution.  Dynamics processing works in the area farthest from the noise floor by nature.

There is no "stratosphere" here.  There's no way to magically raise 0dBFS 3dB.   In the end, you can only do so much before the damage is just too great to live with.  Where that limit is changes with the individual making the judgment, not the DSP or bit depth.

So I'd have to say high bit depth processing doesn't gain you anything when it comes to loudness-war processing.  The ability perform effective loudness processing is not contingent on word bit depth, it's entirely dependent on how the specific process works and how it is employed.


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## TheSonicTruth (Jan 7, 2020)

bigshot said:


> Whether or not a remastered album sounds better or not dependsmore on how the processing is
> applied, than it does whether processing has been applied at all. Bad remastering will result
> in a worse sounding album. Good processing would result in a better sounding album. Compression
> and noise reduction have existed since the late1930s, and I have 78s that have been over compressed
> ...



But I'm not debating original Vs remaster on a 'which sounds better' basis.

I'm generally opposed to remastered reissues of classic pop genre(rock, rap, country, etc) because the sound of the reissued music has been _changed _from either what I'm used to it sounding like, or what I remember it sounding like, etc etc.  It's not whether the remastered version sounds better or worse, but rather that it sounds _different. _I'm sorry to anyone whom I might have previously and unintentionally misled, on this point, in the past.


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## TheSonicTruth (Jan 7, 2020)

pinnahertz said:


> A good part of my career has been spend in broadcasting where there is a very
> aggressive version of remastering processing going on in the final processor
> before transmission.



And that sound is exactly what it seems modern popular mastering - at the request of the client, or the mastering engineer's personal technique or style - is after.   While listening to NY City's Z-100 CHR FM station, Adele's 'ROLLING IN THE DEEP' came on.  I quickly got up, grabbed my '21' CD, and put it in the carousel.  Sure enough, that same song playing from my CD sounded *almost* as loud as it did  when switching between it playing on the radio station and on my CD deck.  This experience recalls an anecdotal tale I recall from the early 1980s, when Carly Simon referenced that same Z-100 station in a conversation with her engineer about how she wanted her then current album project to sound.  Not a good role model I think - a major metro Contemporary Hits Radio outlet, that is!


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## pinnahertz

TheSonicTruth said:


> And that sound is exactly what it seems modern popular mastering - at the request of the client, or the mastering engineer's personal technique or style - is after.   While listening to NY City's Z-100 CHR FM station, Adele's 'ROLLING IN THE DEEP' came on.  I quickly got up, grabbed my '21' CD, and put it in the carousel.  Sure enough, that same song playing from my CD sounded *almost* as loud as it did  when switching between it playing on the radio station and on my CD deck.  This experience recalls an anecdotal tale I recall from the early 1980s, when Carly Simon referenced that same Z-100 station in a conversation with her engineer regarding how she wanted her then current album project to sound.  Not a good role model I think - a major metro Contemporary Hits Radio outlet, that is!


My compliments on attempting the comparison!  

I'm not completely discounting your effort, but please recognize the flaw here.  An FM station must limit "peaks of frequent recurrence" to a level defined as 100% modulation.  All stations are dealing with the same rules in the US.  The loudness war is a fight for the highest apparent loudness without exceeding legal maximum peak level.  Competition in the market drives every station of a given music genre to attempt loudness equal or greater to it's neighbor up or down the dial.  That maximum peak level is pretty hard and firm, but average or RMS loudness levels are not regulated at all.  

On a CD, the maximum is also there, 0dBFS.  But there's rarely an instant comparison between CDs, like pushing the preset tuning buttons on a radio.  Still, it is the same war.

The problem is, we don't know what specific level 0dBFS is in any system (we just turn the volume control) and we don't know what specific level 100% FM modulation is.  They come from two different devices, and are almost certainly not the same.  So a direct loudness comparison is not actually possible without a LOT of setup effort.  Confounding the comparison further, the station is applying a rather huge amount of its own processing to a tune that has already had loudness processing applied. Part of the station's processing chain involves dynamic equalization, which is a large part of what makes up the "signature" of the station.  It's applied in multiple bands, 2 to as many as 30, but typically 5 or 6.  With all of that, plus broadband compression, high frequency limiting to protect from overmodulation, peak limiting, and yes, clipping, you'll have only two distantly related results.  

IF they were similar, that actually says something for the station.  But it could also be accidental.  More likely, the FM station will sound duller by comparison, and with less separation, unless you're listening to HD radio.  Duller comes from trying to control the HF boost of pre-emphasis, which is 17.5dB @ 15kHz re: 400Hz. You don't have that on CD or HD radio.  Hence the duller result on FM. 

Oh, and FM has it's own kind of anti-aliasing filter between 15 and 17kHz, keeping audio away from the 19kHz pilot tone.  Those filters have always been a problem all their own.


----------



## TheSonicTruth (Jan 7, 2020)

pinnahertz said:


> The problem is, we don't know what specific level 0dBFS is in any system
> (we just turn the volume control) and we don't know what specific level 100%
> FM modulation is. They come from two different devices, and are almost
> certainly not the same. So a direct loudness comparison is not actually possible
> without a LOT of setup effort. C



It's not that complicated.  I just _listened _while switching between CD and Tuner on my home system, while not touching the Volume control.  And based on my own _ears_, 'ROLLING IN THE DEEP' on my 21 CD sounded almost as loud as it did on WHTZ 100FM.  No complicated setup needed to determine that!

From a sound quality standpoint, no contest: the CD version blows away how her song sounded on FM, and to a degree on HD FM(which I think is one of the latest broadcasting gimmicks - why would I want to hear the same hyper-mastered pudding in glorious, static-free 'HD'??).


----------



## pinnahertz

TheSonicTruth said:


> It's not that complicated.  I just _listened _while switching between CD and Tuner on my home system, while not touching the Volume control.  And based on my own _ears_, 'HELLO' on my 21 CD sounded almost as loud as it did on WHTZ 100FM.  No complicated setup needed to determine that!


You missed the point entirely.  Your comparison doesn't include matched gains.  Without matching gains between the two, the comparison includes a form of bias.  Yes, complicated setup is required.  You need to establish 0dBFS through the system for the CD, and match 100% FM modulation to it within 0.1dB.  The CD is easy, matching to 100% mod, not so much.  

Sorry, that's life.  Your ears told you something that has a built-in, unknown, uncontrolled error. The chances of accidental CD/FM gain-match are near zero.


----------



## TheSonicTruth

pinnahertz said:


> Sorry, that's life. Your ears told you something that has a built-in, unknown,
> uncontrolled error.



You sure you aren't the White House Press Secretary?  Because with the above, you dang sure sound like one!  Telling me what I hear with my own two ears isn't real, or is some "mistake"?


----------



## pinnahertz

TheSonicTruth said:


> You sure you aren't the White House Press Secretary?  Because with the above, you dang sure sound like one!  Telling me what I hear with my own two ears isn't real, or is some "mistake"?


I'm confused.  

Your sig says, "if someone tells you "Use your ears" _- they're probably a politician...", _and now you say if I tell you _*not*_ to believe your ears, I'm a politician too?  

So, *anyone* who disagrees with you is a politician?  Ahem.  Couldn't be farther from one.

I'm telling you the facts.  If you're making a comparison between two sources, trying to detect differences or similarities, and you DON'T level match, you're going to get an erroneous answer BECAUSE you've introduced an artificial difference (level) that isn't part of the real difference you're trying to hear. But don't take my word for it, just do a bit of research on ABX/DBT.  It's all there, the principles apply even if you're not doing an ABX.   

Again, I applaud your attempt, it's far more than most would do.  I also point out the point of entry for error.


----------



## old tech

@pinnahertz apart from your undoubted knowledge and experience in audio science, you write like a diplomat.


----------



## pinnahertz

old tech said:


> @pinnahertz apart from your undoubted knowledge and experience in audio science, you write like a diplomat.


I'm gutted.


----------



## old tech

pinnahertz said:


> I'm gutted.


Why? A diplomat is better than a politician


----------



## Davesrose (Jan 8, 2020)

old tech said:


> Why? A diplomat is better than a politician



Wouldn't that depend on who they're trying to serve?  I'm sure given today's age, there are plenty of people who question whether a diplomat is from a certain lobby.


----------



## old tech

Davesrose said:


> Wouldn't that depend on who they're trying to serve?  I'm sure given today's age, there are plenty of people who question whether a diplomat is from a certain lobby.


Hmmm, you're right, I never thought of it that way.  I was referring to Pinnahertz's nice writing style rather than insinuating something else.


----------



## bigshot (Jan 8, 2020)

Pinnahertz, one of the knives need sharpening in that drawer.


----------



## bigshot (Jan 8, 2020)

old tech said:


> I'm not doubting that compression, NR, etc have always existed (and for good reasons), the point is that the original production masters would already have been mastered with it (ie compression, EQ and other 'tricks') that the original producer/artist wanted.



I'll let you in on a dirty little secret... After a decade or more, the people who made the album in the first place aren't interested in going back and revisiting it. They just want to endorse the royalty checks and let the label take care of the technical details. It may say in the liner notes that the remastering was overseen by X, Y and Z; but all that means is that they approved what was handed to them. That doesn't always mean that it is their intent. The other dirty little secret is that the first release may have been done under time and budgetary constraints that don't exist decades later. The remix or remaster may have more resources to do a better job. It all depends. Some stuff sounds great and some stuff sucks. That's the way it has always been, and it's the way it will continue to be.


----------



## 71 dB

TheSonicTruth said:


> But I'm not debating original Vs remaster on a 'which sounds better' basis.
> 
> I'm generally opposed to remastered reissues of classic pop genre(rock, rap, country, etc) because the sound of the reissued music has been _changed _from either what I'm used to it sounding like, or what I remember it sounding like, etc etc.  It's not whether the remastered version sounds better or worse, but rather that it sounds _different. _I'm sorry to anyone whom I might have previously and unintentionally misled, on this point, in the past.



Then again nobody forces you to buy/listen to the remastered version. Just stick to the original.


----------



## TheSonicTruth (Jan 8, 2020)

71 dB said:


> Then again nobody forces you to buy/listen to the remastered
> version. Just stick to the original.



I'm not concerned about me - *EDIT:* I already possess most of what I want in some original form. *END OF EDIT. *  I'm concerned about newer generations coming up, exploring music from my time and before.  They won't have access to that 'original' sound because those sources are off the shelves in stores!  Therefore, _they're _forced to buy(or download) the 'remastered' or remixed version because that's all that is available.

IE: The 2017 Sgt. Pepper 50th Anniv. remixes.  Young Beatle fans will buy or listen to those and just assume they're the canonical, authoritative versions.  They might not know, or care, that there are more authentic versions of Sgt. Pepper, and other albums, out there.


----------



## TheSonicTruth (Jan 8, 2020)

bigshot said:


> The other dirty little secret is that the first release may have been done
> under time and budgetary constraints that don't exist decades later.



Well guess what...  Those constraints, at the time of the release of that 1964 oldie or 1978 gem, resulted in those songs or albums sounding the way they did the first time most of us heard them.  And, for 'better' or for worse, that is what, not just a few of us, prefer.


----------



## SoundAndMotion (Jan 8, 2020)

Yesterday:


TheSonicTruth said:


> But I'm not debating original Vs remaster on a 'which sounds better' basis.
> 
> I'm generally opposed to remastered reissues of classic pop genre(rock, rap, country, etc) because the sound of the reissued music has been _changed _from either what I'm used to it sounding like, or what I remember it sounding like, etc etc.  It's not whether the remastered version sounds better or worse, but rather that it sounds _different. _I'm sorry to anyone whom I might have previously and unintentionally misled, on this point, in the past.


Today:


TheSonicTruth said:


> I'm not concerned about me - I'm concerned about newer generations coming up exploring music from my time and before.  They won't have access to that 'original' sound because those sources are off the shelves in stores!  Therefore, _they're _forced to buy(or download) the 'remastered' or remixed version because that's all that is available.


Please pick one: (you or the kids). Or perhaps we can expect another reason you dislike remasters tomorrow?


----------



## TheSonicTruth (Jan 8, 2020)

SoundAndMotion said:


> Yesterday:
> 
> Today:
> 
> Please pick one: (you or the kids). Or perhaps we can expect another reason you don't remasters tomorrow?


 

Go back and re-read #5565:   I edited that comment.


----------



## bigshot

TheSonicTruth said:


> Well guess what...  Those constraints, at the time of the release of that 1964 oldie or 1978 gem, resulted in those songs or albums sounding the way they did the first time most of us heard them.  And, for 'better' or for worse, that is what, not just a few of us, prefer.



That's fine, but that doesn't have anything to do with audio fidelity, and it's purely subjective. I guess when you listen to new music, you're free to choose the best sounding release. Just buy older used CDs of music you know and new ones for music you are just discovering.


----------



## bigshot (Jan 8, 2020)

TheSonicTruth said:


> IE: The 2017 Sgt. Pepper 50th Anniv. remixes.  Young Beatle fans will buy or listen to those and just assume they're the canonical, authoritative versions.  They might not know, or care, that there are more authentic versions of Sgt. Pepper, and other albums, out there.



Didn't the Beatles make something like 17 different versions of "Lucy in the Sky with Diamonds"? Is the one you're familiar with more "authentic" than the other 16? The Beatles didn't like what Phil Spector did with the "Let It Be" album. Since the Spector mix is the one you are familiar with, does that mean that Let It Be (Naked) or the Get Back sessions aren't canonical? Is "You Can't Always Get What You Want" supposed to have a boys' choir or not? It was originally released both ways. If an artist goes back and remixes his music and announces that the remix is what he intended to do when he originally released the album, but he just didn't have the resources to do it at the time; does that mean that your memory is a better guide for what young listeners should listen to than the performer himself?

I really don't think there is a black and white answer here. Dogma doesn't work. You have to take it on a case by case basis and judge with your own ears and tastes.


----------



## TheSonicTruth

bigshot said:


> That's fine, but that doesn't have anything to do with audio fidelity,
> and it's purely subjective. I guess when you listen to new music, you're free
> to choose the best sounding release. Just buy older used CDs of music you
> know and new ones for music you are just discovering.



But, at the risk of repeating myself-sk of repeating myself-sk of repeating myself-sk of repeating myse - _Hey honey will you lift the tonearm, please!!  _

Anywho, I'll say yet a dozenth time: It's not (always) about fidelity.  It's about how you remember it sounding the first time you heard it.

As far as canon goes - in your second reply you mentioned that Stones song, 'Can't Always Get..'.  The choir version is all the radio has played around these parts, for as long as I have been alive to hear the song.  Conversely, I have heard stations alternate betweem the single and album version of the Beatles' 'Let It Be' for years, also.   Or Elvis, 'Suspicious Minds' _without _those soaring trumpets? Sounds like all the air let out of the tires. lol!


----------



## SoundAndMotion

TheSonicTruth said:


> Go back and re-read #5565:   I edited that comment.


I read the changes; you still seem to contradict yourself. How would the newer generation know about how you remember the earlier versions?

I would have a lot more respect for you if you simply, honestly said that you prefer the versions that you first got used to, rather than contort yourself trying to make that perspective seem objectively true.

If you did, I'd comment that I'm also like that sometimes. 
For me, Sean Connery *is* James Bond. Although Roger Moore, Daniel Craig and the others play (Sean Connery) well, they are not real to me. 
For me, the late 60's Porsche 911 *is* the 911. The modern 911 is a bloated, power-steered copy.
More relevant to music: the first Ramones album should sound like one take, with Joey and Tommy right in the middle, Dee Dee hard panned left and Johnny hard panned right. The CD I have with a "normal" mix just sounds wrong to me.
These subjective preferences aren't wrong, but they can't be argued into being "right" for everyone.

Every time you argue about the creative choices made in producing some version of a song, I can't help but imagine you telling Monet to paint less blurry or Hemingway to write longer sentences.


----------



## bfreedma

TheSonicTruth said:


> But, at the risk of repeating myself-sk of repeating myself-sk of repeating myself-sk of repeating myse - _Hey honey will you lift the tonearm, please!!  _
> 
> Anywho, I'll say yet a dozenth time: It's not (always) about fidelity.  It's about how you remember it sounding the first time you heard it.
> 
> As far as canon goes - in your second reply you mentioned that Stones song, 'Can't Always Get..'.  The choir version is all the radio has played around these parts, for as long as I have been alive to hear the song.  Conversely, I have heard stations alternate betweem the single and album version of the Beatles' 'Let It Be' for years, also.   Or Elvis, 'Suspicious Minds' _without _those soaring trumpets? Sounds like all the air let out of the tires. lol!




When did the first version of a song YOU heard become the defacto artist approved standard?

This entire discussion is about your preferences, not audio fidelity or preservation of artist intent.  I get that you're passionate about this, but please, move on.


----------



## GearMe

TheSonicTruth said:


> I'm not concerned about me - *EDIT:* I already possess most of what I want in some original form. *END OF EDIT. *  I'm concerned about newer generations coming up, exploring music from my time and before.  They won't have access to that 'original' sound because those sources are off the shelves in stores!  Therefore, _they're _forced to buy(or download) the 'remastered' or remixed version because that's all that is available.
> 
> IE: The 2017 Sgt. Pepper 50th Anniv. remixes.  Young Beatle fans will buy or listen to those and just assume they're the canonical, authoritative versions.  They might not know, or care, that there are more authentic versions of Sgt. Pepper, and other albums, out there.



First off...for their own good, young Beatle fans should be summarily redirected to The Stones, The Who, The Grateful Dead, The Allman Brothers, The Jimi Hendrix Experience, and The Clapton  
(The Doors, The Doobie Brothers, The Byrds, The Animals, The Kinks, The Moody Blues...OK, the last one was a stretch!)

That aside, honestly who cares if the version they hear is canonical/authoritative/etc?  I just hope they're enjoying whatever version they're hearing -- _you know...if they're into that sort of music_ 

Seriously though, think of all the different versions of great songs by great groups out there!  Do we really need to agonize over stuff like this or _just Listen to the Music_?

Who knows, with a Little Help from their Friends, maybe they'll get a whole new appreciation for the variety of music genres, song versions, covers!!!
(thank goodness for covers...or I'd never listen to any Beatles songs)


----------



## TheSonicTruth (Jan 8, 2020)

SoundAndMotion said:


> the first Ramones album should sound like one take, with Joey
> and Tommy right in the middle, Dee Dee hard panned left and
> Johnny hard panned right. The CD I have with a "normal" mix just
> sounds wrong to me.



Doesn't that just bust your bubble?

You finally take home a CD issue of your absolute favorite album by your favorite artist, put it in your machine, and come to find they done some messed-up schitt in the mastering dept, or, as you mentioned, some weird new mix y'don't remember, makes you wanna yank it off your machine, open your window, and hurl that sucker up on your neighbor's roof across the street!!


----------



## 71 dB

TheSonicTruth said:


> I'm not concerned about me - *EDIT:* I already possess most of what I want in some original form. *END OF EDIT. *  I'm concerned about newer generations coming up, exploring music from my time and before.  They won't have access to that 'original' sound because those sources are off the shelves in stores!  Therefore, _they're _forced to buy(or download) the 'remastered' or remixed version because that's all that is available.
> 
> IE: The 2017 Sgt. Pepper 50th Anniv. remixes.  Young Beatle fans will buy or listen to those and just assume they're the canonical, authoritative versions.  They might not know, or care, that there are more authentic versions of Sgt. Pepper, and other albums, out there.



Newer generations are not interested of the music of your time and before, remastered or not. You think they are into Sgt. Pepper? Hah! They are into *Mabel*'s "_God Is A Dancer_". If you want the millenials to discover your favorite old music, the quality of the remaster is least of your problems. Rock music "died" in 2010. We have been living the age of dancepop for the last decade. Commercially that is.

Old used CDs are of course available in second hand shops and places such as Amazon marketplace. A few years ago I purchased some CD albums from the 80's (such as Kate Bush and Peter Gabriel) and used Amazon marketplace to get the original CD releases instead of newer remasters. Those CDs were very cheap.


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## TheSonicTruth (Jan 8, 2020)

GearMe said:


> First off...for their own good, young Beatle fans should be summarily
> redirected to The Stones, The Who, The Grateful Dead, The Allman
> Brothers, The Jimi Hendrix Experience, and The Clapton



Yeah, sure, we know.  You're one of those strange Yankees who likes every classic rock act except for the Beatles. I have personally met a few weirdos who like all classic rock except by Joel or Springsteen.

What are the demographics behind _that_ phenomenon?

By the way, this thread _was _ supposed to be about 16bit vs 24bit from a consumer perspective, right?  Well at least enough of us know that there isn't really a difference, so we can just discuss the music itself, lol!  BTW, I do enjoy material from everyone on your list, and also the Beatles, Joel, and Bruce!


----------



## TheSonicTruth (Jan 8, 2020)

71 dB said:


> 1) Newer generations are not interested of the music of your time and
> before, remastered or not. You think they are into Sgt. Pepper? Hah! They
> are into *Mabel*'s "_God Is A Dancer_". If you want the millenials to discover your
> favorite old music, the quality of the remaster is least of your problems. Rock
> ...




1) I do come across post-Millennials with actual musical taste, occasionally, in my place of work.  They inherited their parents turntable or CD deck and want to know I have any Beatles or Allmans material in the back room.  The only ones who pi$$ me off are the college-age buffoons who buy LPs for the sole purpose of decorating their frickn' DORM  ROOMS.  They want to just hang the discs on the walls from their spindle holes, or better yet, paint them different colors.  I almost feel like telling them "don't even bother buying my vinyl. If you want something for your dorm wall, try an antique payphone or a lithograph of an advertisment for Cunard."  I admonish them: "_Records are meant to be PLAYED and LISTENED TO, not destroyed by Sherwin-Williams_", and the kiddies get all red-faced and embarassed, lol!

2)  How do you think I've amassed my personal collection?  Between Goodwills, Discogs, Library sales, and used yellow tags(when FYE was still around). But now the pickens are gett'n slim...!


----------



## bigshot

Welp! I think I'll watch an episode of Hoarders or Honey Boo Boo now.


----------



## pinnahertz

71 dB said:


> Newer generations are not interested of the music of your time and before, remastered or not. You think they are into Sgt. Pepper? Hah! They are into *Mabel*'s "_God Is A Dancer_". If you want the millenials to discover your favorite old music, the quality of the remaster is least of your problems. Rock music "died" in 2010. We have been living the age of dancepop for the last decade. Commercially that is.


The high school in our town has an FM station.  Their "format" is fundamentally "classic rock".  They've done features on Hendrix, Zeppelin, Janis Joplin and Greatful Dead.  The do play contemporary, indy, etc. too, but it's heavily classic rock.  These are high school kids picking their favorite music.


----------



## TheSonicTruth

pinnahertz said:


> The high school in our town has an FM station.  Their "format" is
> fundamentally "classic rock".  They've done features on Hendrix, Zeppelin,
> Janis Joplin and Greatful Dead.  The do play contemporary, indy, etc. too,
> but it's heavily classic rock.  These are high school kids picking their
> favorite music.




As I've said... There's hope out there


----------



## bigshot (Jan 9, 2020)

I have to be honest... I kind of feel sorry for people who listen to the same music they listened to in high school when they are over 40. It's fine if you want to do that. I won't tell you that you shouldn't, but it's like having the same breakfast, lunch and dinner every day of your life. That makes me sad.


----------



## chef8489

bigshot said:


> I have to be honest... I kind of feel sorry for people who listen to the same music they listened to in high school when they are over 40. It's fine if you want to do that. I won't tell you that you shouldn't, but it's like having the same breakfast, lunch and dinner every day of your life. That makes me sad.


I listen to some of the music I listened to in High School, but I have a pretty eclectic taste in music.


----------



## 71 dB

bigshot said:


> I have to be honest... I kind of feel sorry for people who listen to the same music they listened to in high school when they are over 40. It's fine if you want to do that. I won't tell you that you shouldn't, but it's like having the same breakfast, lunch and dinner every day of your life. That makes me sad.



I do listen to my "high school music" sometimes, but my taste has expanded constantly ever since, sometimes to unexpected directions. For many "high school music" was rock/metal, but for me it was UK Acid House of 1988. I didn't really get into rock until around 2001 (at age of 30) when I got into some soft rock bands and in 2008 I discovered King Crimson.

I have earned the right to like/enjoy stupid silly music because I have done my exploration work and also found tons of very sophisticated art music which I also enjoy. There is nothing wrong with listening to dumb music if it improves your well-being, but it's also worth it to explore with an open mind, because music has so much to offer. So, like you I feel a bit sad for people who think the music they discovered at age 15 is all there is to explore.


----------



## 71 dB

pinnahertz said:


> The high school in our town has an FM station.  Their "format" is fundamentally "classic rock".  They've done features on Hendrix, Zeppelin, Janis Joplin and Greatful Dead.  The do play contemporary, indy, etc. too, but it's heavily classic rock.  These are high school kids picking their favorite music.



Those "kids" running the FM station must be some classic rock enthusiasts who have been "indoctinated" into the genre by their parents and are in no way a representation of the music taste of the millenials in general. If they were, Calvin Harris wouldn't be making millions. Rock "killed" Jazz in the 50's. Jazz continued to live, but in marginal. Rock had a good run of about 50 years, but now it has been pushed into the marginal by dance pop which will be pushed into the marginal by the next "big thing" whatever it is sometimes in the future. This is how music is. Genres are merely different sonic packages for musical ideas. New packages emerge and older packages go out of fashion into the marginal to be appreciated by those who explore beyond the mainstream.


----------



## 71 dB

TheSonicTruth said:


> How do you think I've amassed my personal collection?  Between Goodwills, Discogs, Library sales, and used yellow tags(when FYE was still around). But now the pickens are gett'n slim...!



Maybe you already have almost everything in your personal collection? What is this music you'd want to collect but you don't find anywhere? As a hobby "collecting" something is doomed, because eventually you have collected everything there is to collect or the things you are collecting are too difficult to obtain making the process slow. When I discovered Tangerine Dream in 2008, I collected 30 CDs per year of their music for the first 3 years, but after 90 CDs things became much more difficult. Collecting slowed down, significantly. The stuff I don't have is so expensive I am not willing to pay the prices.


----------



## TheSonicTruth

71 dB said:


> ffer. So, like you I feel a bit sad for people who think the music they
> discovered at age 15 is all there is to explore.



I hope you are not including me in that assessment.  There is a lot of post-2000, 2010s stuff I enjoy.  All I'm saying is, leave the stuff from before 1990 alone.  No 're-envisioning', no 'remastering', etc.


----------



## GearMe

chef8489 said:


> I listen to some of the music I listened to in High School, but I have a pretty eclectic taste in music.



Ditto...Am guessing many folks on these forums have a wide variety of music genres they listen to.  I certainly do and am always interested in new music...if it's engaging and is well done!


----------



## 71 dB

TheSonicTruth said:


> I hope you are not including me in that assessment.  There is a lot of post-2000, 2010s stuff I enjoy.  All I'm saying is, leave the stuff from before 1990 alone.  No 're-envisioning', no 'remastering', etc.



I am not including you because I don't know you well enough to know where you should be included and I never thought you only enjoy older music.

Money talks and even if some people want to protect the past, money rules. So, if there is money to be made of re-envisionistic remasters that's what we will have.
I have to say music before 1990 was not always perfectly mixed at all and skillful remastering can improve things, but "loudness war" is not that!


----------



## SoundAndMotion

TheSonicTruth said:


> Doesn't that just bust your bubble?


No. I prefer the original, but enjoy the CD version often (no turntable in the car!)



TheSonicTruth said:


> ... makes you wanna yank it off your machine, open your window, and hurl that sucker up on your neighbor's roof across the street!!


There's no shame in seeking therapy!


----------



## bigshot (Jan 9, 2020)

71 dB said:


> Those "kids" running the FM station must be some classic rock enthusiasts who have been "indoctinated" into the genre by their parents and are in no way a representation of the music taste of the millenials in general.



It's more likely that their playlists are vetted by a Boomer school administrator.



GearMe said:


> Ditto...Am guessing many folks on these forums have a wide variety of music genres they listen to.



I work with young film makers and one of the questions I ask them is what sort of music inspires them. Inevitably, they answer "all kinds"... so I ask them what their favorite opera is. (blank look) How about Latin? (blank look) Country music? (blank look) Chamber music? (blank look) Early jazz or ragtime? (blank look) Piano concerto? (blank look)... It turns out that "all kinds" means "all kinds of rock music".

As a side note... I heard a Muzak version of The Ramones' "I Wanna Be Sedated" at the supermarket once.


----------



## gregorio (Jan 9, 2020)

old tech said:


> [1] What I was trying to say is that there was a limit to how much compression etc when processing power was limited to analog equipment or 16 bit digital. In other words, the greater processing power could and is (in many cases) abused.
> [2] What I (and Sonic Truth, I think) were getting at is that many or most of these flat transfer CDs subjectively sound better than their later remasters simply because they were a flat transfer without any futzing or attempt to make them louder and more compressed/limited.



1. There were always limits and the history/evolution of popular music genres from the 1960's onwards is largely dependent on abusing them. Arguably the biggest and most obvious such abuse was with the electric guitar in the 1960's, where over-driving guitar amps/cabs not just to the point of distortion but so massively beyond the point of distortion that pretty much the only thing being output was distortion! Compared to that, the amount of distortion from the loudness wars is relatively small, so why is the former not only acceptable but desirable and the latter so objectionable? The simple answer is age/generation! The guitar distortion wasn't acceptable, in fact it was so unacceptable that to my mother Hendrix didn't even qualify as music, it was just a horrible noise! And, a least in part, that is why I liked Hendrix, it was cool and radical, music specifically for my generation and not for older generations but of course now, I'm a member of the "older generations" and some contemporary popular music/genres I like, while to me, some of it barely even qualifies as music and some just sounds like bad music, packed with incompetent mistakes. Probably the old time engineers thought it sounded like an incompetent mistake when they heard guitar feedback back in the '60's. And, guitar distortion is just one of COUNTLESS examples, the same thing happened in the late '70's and early '80's with (analogue) compressors/limiters, over-driven to the point of destruction. In fact the last great analogue compressor (the Distressor, in the mid 1990's) was based on the sound characteristics of several previous vintage compressors and had a setting which emulated the hugely over driven compressors of the late '70's. The setting was called "Nuke", which should tell you all you need to know about what it did to sound quality!!

In other words, your statement is effectively true, the advance in technology (processing power and the algorithms that took advantage of it) could be abused, exactly the same as just about all music technology advances have always been abused since at least the mid '60's. Why should contemporary musicians and engineers not be allowed to do what previous generations of musicians and engineers did? If the response is "because older generations don't like it or think it's a mistake", that's about as counter-productive an argument as I can imagine!

2. I'm sure there were some "flat transfers", especially when CD really took off and they couldn't make the content quick enough but generally there would be some tinkering which might have been more "futzing" or just as likely, some de-futzing and often made louder (more compressed), so it didn't sound too quiet compared to contemporary releases. The loudness war may only have come to audiophiles' attention in the last 15 years or so but has been ongoing for at least 50, in fact it started with Juke boxes in the 1950's.



TheSonicTruth said:


> [1] It's not whether the remastered version sounds better or worse, but rather that it sounds _different.
> [2] You finally take home a CD issue of your absolute favorite album by your favorite artist, put it in your machine, and come to find they done some messed-up schitt in the mastering dept, or, as you mentioned, some weird new mix y'don't remember, makes you wanna yank it off your machine, open your window, and hurl that sucker up on your neighbor's roof across the street!!
> [3] Those constraints, at the time of the release of that 1964 oldie or 1978 gem, resulted in those songs or albums sounding the way they did the first time most of us heard them. And, for 'better' or for worse, that is what, not just a few of us, prefer._



1. Of course, that's the whole point of a remaster, what would be the point of paying a mastering engineer to make a new master that sounded identical to an existing master?

2. I got that even before CD was released. You heard some new song on the radio or TV, went out and bought the cassette or LP and it didn't sound the same. As pinnahertz pointed out, the broadcast chain significantly changed it and not uncommonly, an edit specifically for radio play was made, to fit the required time slot.

3. Define "not just a few of us". Of course some will prefer an original version to a cleaner, usually louder (and in other ways different) master, some others used to an original will prefer the remaster though (or at least not hate it enough not to buy it) and those not used to an original will generally prefer the remaster. It's a sales issue, the labels need to monetise their back catalogue and if that upsets those "some" vehemently against a remaster, then so be it, they are relatively just a few. As bigshot stated though, it really has to be judged on a case by case basis, some remasters are better, even to most of us old timers. This makes those who're vehemently against remasters on principle a tiny "lunatic fringe" and there will always be a few of those, no matter what you do (or don't do).

G


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## TheSonicTruth (Jan 9, 2020)

71 dB said:


> Maybe you already have almost everything in your personal
> collection? What is this music you'd want to collect but you don't
> find anywhere?



When's the last time you found an original CD release of 'TNT' by AC/DC in a Goodwill?   Or 'DARK SIDE OF THE MOON' by Pink Floyd or any of the original Van Halen CD catalog at a library benefit sale?   Nowadays, finding 'RELISH' by Joan Osborne or 'CRACKED REAR VIEW' by Hootie & The Blowfish is a big deal!  Know what I mean?


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## TheSonicTruth

71 dB said:


> I have to say music before 1990 was not always
> perfectly mixed at all



I resemble that comment, lol!

What would you have done differently on, say, 'BORN IN THE USA' or 'ELECTRIC LADYLAND'?

There's no such thing as a perfect music mix, or master - from before or since 1990!


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## 71 dB

bigshot said:


> I work with young film makers and one of the questions I ask them is what sort of music inspires them. Inevitably, they answer "all kinds"... so I ask them what their favorite opera is. (blank look) How about Latin? (blank look) Country music? (blank look) Chamber music? (blank look) Early jazz or ragtime? (blank look) Piano concerto? (blank look)... It turns out that "all kinds" means "all kinds of rock music".



"All kinds" is also an answer when you are unsure of your taste and knowledge of music and don't want to expose it. 

I enjoy some opera music, especially by Rameau and Handel (baroque era) era and Puccini (romantic era). Even when I enjoy a lot of Mozart, I am not into Mozart's operas and I don't like Verdi either. Wagner is ok and so on. Favorite opera? Perhaps _Turandot_ by Puccini, _Giulio Cesare_ by Handel or _Les Indes galantes_ by Rameau. It's impossible to put one work above so many others when they all have so much to offer in their own ways.

I don't really listen to Latin music. Tangos by Piazzolla comes the closest perhaps? 

Country music? Not my thing, but I am a fan of Carly Simon and some of her songs are that style.

Chamber Music? So much of that I enjoy! I could mention Beethoven's late String Quartets of Faure's Piano Quintets. Mozart, Haydn, Dittersdorf, J.S.Bach, C.P.E Bach, Mendelssohn, Saint-Saëns, Brahms, Elgar, Villa-Lobos, Buxtehude…. that list is endless!

Early Jazz/Ragtime? Not my thing.

Piano Concertos? Mozart's later Piano Concertos kick ass! 

Anyway, answering questions like these is difficult so you get the easy answer: "all kinds".


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## bigshot

When I get the blank stare, I offer them some breadcrumbs to get them started in the areas they have no experience in. Creative people are open to new kinds of music, it's just that the media just offers the same thing over and over.


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## 71 dB

TheSonicTruth said:


> When's the last time you found an original CD release of 'TNT' by AC/DC in a Goodwill?   Or 'DARK SIDE OF THE MOON' by Pink Floyd or any of the original Van Halen CD catalog at a library benefit sale?   Nowadays, finding 'RELISH' by Joan Osborne or 'CRACKED REAR VIEW' by Hootie & The Blowfish is a big deal!  Know what I mean?



I'm not into AC/DC, Pink Floyd or Van Halen nor do I live in the US so I wouldn't know. Some CDs are really hard to find, that's just how it is.


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## 71 dB

TheSonicTruth said:


> I resemble that comment, lol!
> 
> What would you have done differently on, say, 'BORN IN THE USA' or 'ELECTRIC LADYLAND'?
> 
> There's no such thing as a perfect music mix, or master - from before or since 1990!



I have never heard those albums so I don't know how well they are produced or what I would do differently.


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## bigshot (Jan 9, 2020)

Born in the USA is a pretty straightforward mix and it sounds great. Electric Ladyland is the exact opposite. Hendrix was experimenting with overdubbing guitar parts and some of it ended up sounding really dense. This was corrected by the original sound engineer, Eddie Kramer who came back out of retirement to supervise a 5.1 mix. It is crystal clear and all the parts are perfectly balanced and can be heard separate from the mass. This is an example of an album that sounds a lot better in the remix than it ever did on LP or CD. The same can be said of Miles Davis' Bit ches Brew, another album that had very complex overdubbing and arrangements that became congested in two channel, but sound clear and focused in 4.0.


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## GearMe (Jan 9, 2020)

bigshot said:


> ...I work with young film makers and one of the questions I ask them is what sort of music inspires them. Inevitably, they answer "all kinds"... so I ask them what their favorite opera is. (blank look) How about Latin? (blank look) Country music? (blank look) Chamber music? (blank look) Early jazz or ragtime? (blank look) Piano concerto? (blank look)... It turns out that "all kinds" means "all kinds of rock music".



As a person that tries to see things in a 'glass half-full' way, I'm gonna go with the 'they're young' thing (yeah...I know OK Boomer) and hope that they continue to grow in their music/life experiences so they can learn to appreciate different genres and artists



bigshot said:


> As a side note... I heard a Muzak version of The Ramones' "I Wanna Be Sedated" at the supermarket once.



Muzak Ramones?   Hmmm...probably better than a Kenny G version of it!    
(Weirdly enough...that song title may be the perfect descriptor for the Kenny G genre of music.  _Yes_, he should have his own genre)

¡Ay, caramba!...speaking of Latin 
WARNING!  You can't un-hear this


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## GearMe

bigshot said:


> Born in the USA is a pretty straightforward mix and it sounds great. Electric Ladyland is the exact opposite. Hendrix was experimenting with overdubbing guitar parts and some of it ended up sounding really dense. This was corrected by the original sound engineer, Eddie Kramer who came back out of retirement to supervise a 5.1 mix. It is crystal clear and all the parts are perfectly balanced and can be heard separate from the mass. This is an example of an album that sounds a lot better in the remix than it ever did on LP or CD. The same can be said of Miles Davis' Bit ches Brew, another album that had very complex overdubbing and arrangements that became congested in two channel, but sound clear and focused in 4.0.



Thanks for the info on the remaster 50th Anniversary for Electric Ladyland...

Looks like a great CD/Blu-ray set...very much appreciate it!


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## TheSonicTruth (Jan 9, 2020)

71 dB said:


> I'm not into AC/DC, Pink Floyd or Van Halen nor
> do I live in the US so I wouldn't know. Some
> CDs are really hard to find, that's just how it is.



What can I say?  I both appreciate listening to, and playing music for, the m'asses!


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## Davesrose

bigshot said:


> I work with young film makers and one of the questions I ask them is what sort of music inspires them. Inevitably, they answer "all kinds"... so I ask them what their favorite opera is. (blank look) How about Latin? (blank look) Country music? (blank look) Chamber music? (blank look) Early jazz or ragtime? (blank look) Piano concerto? (blank look)... It turns out that "all kinds" means "all kinds of rock music".



I never thought that being a music enthusiast was a requisite for being a film maker....when you look at the great film directors of the 20th century, many actually rely on a preferred composer.  It's interesting to see interviews from those composers about which directors give them more direction about what kind of music/theme should be in a scene vs just give the composer free reign.  You see even fewer film makers attempt to be film composers themselves (John Carpenter being a good example of an exception).  Now, more then anything I think movie soundtracks can be an afterthought.  I can hear the same old movements for any battle scene, and a period movie from the 70s or 80s will just have the top billboard hits of that era.  I also see people on Youtube claiming to be "independant film makers".


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## bigshot (Jan 10, 2020)

Davesrose said:


> I never thought that being a music enthusiast was a requisite for being a film maker.....



Film makers supervise every aspect of their films and the soundtrack is every bit as important as the visuals. In my particular field, animation, the timing of the action is planned to the rhythms of the music. Directors have to be as attuned to music as imagery. There are directors who use music as auditory wallpaper, but those aren’t great directors.

For example... Norman McLaren...


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## castleofargh

What's important is that you can listen to some Caruso in 24/96 so you don't miss out on the dynamic, soundstage, and accuracy of whatever they used back then to record. I'm gonna say, flint stenography?
I was also happy to find Nirvana's "Tourette's" on Qobuz at 24/96. I would never be able to properly enjoy that track in mediocre CD quality.
With that said, I have failed to find a hires version of my favorite 2 Unlimited album(<=my high school music). Can't help but suspect a Boomers' deep state operation against "going for the reaa-eal thing".

Ps: @TheSonicTruth , sorry to disappoint you, this seems to be a remaster. Nothing is sacred anymore.


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## TheSonicTruth (Jan 10, 2020)

castleofargh said:


> What's important is that you can listen to some Caruso in 24/96 so you don't miss out on the dynamic, soundstage, and accuracy of whatever they used back then to record. I'm gonna say, flint stenography?
> I was also happy to find Nirvana's "Tourette's" on Qobuz at 24/96. I would never be able to properly enjoy that track in mediocre CD quality.
> With that said, I have failed to find a hires version of my favorite 2 Unlimited album(<=my high school music). Can't help but suspect a Boomers' deep state operation against "going for the reaa-eal thing".
> 
> Ps: @TheSonicTruth , sorry to disappoint you, this seems to be a remaster. Nothing is sacred anymore.




"Flint stenography"?  Your sense of humor is not appreciated. They knew what they were doing back then.  And all that "24/96" & "mediocre CD" sounds like audiophile talk to me. Beneficial only on the production side.    16/44.1 is good enough as a consumer deliverable.


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## Davesrose (Jan 10, 2020)

bigshot said:


> Film makers supervise every aspect of their films and the soundtrack is every bit as important as the visuals. In my particular field, animation, the timing of the action is planned to the rhythms of the music. Directors have to be as attuned to music as imagery. There are directors who use music as auditory wallpaper, but those aren’t great directors.
> 
> For example... Norman McLaren...




I have experience with animation as well: 3D animation.  I haven't found that music is the only basis 3D animators use for timing.  Sometimes it's timed with a narrator, other background elements, or basic physics during a simulation.  There are also many video examples you can find of composers first watching the editing and syncing music to it.  I'm not disagreeing that good directors do have opinions about what music would add most drama to a scene, but I haven's seen evidence that every film maker is a music enthusiast and wants to have a broad knowledge of all music (my point was that there are varying degrees of how many rely on the composer).

There might be some "overall timing" considered with music and animation, but recordings don't happen until after editing and animation has started:



Example of conductor using movie edit for timing:


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## TheSonicTruth (Jan 10, 2020)

GearMe said:


> As a person that tries to see things in a 'glass half-full' way, I'm gonna go with the
> 'they're young' thing (yeah...I know OK Boomer) and hope that they continue to
> grow in their music/life experiences so they can learn to appreciate different
> genres and artists
> ...





Back sixty-ought years ago, when the primary reason for music in stores was mask such retail unpleasantries as the rattling of shopping carts, mechanisms of cash registers, deli equipment, the prattle of fellow shoppers, etc., Muzak was a cheap, royalty-free(to the retailer that is) alternative to paying through the nose monthly to pipe the actual artists' works into the stores.   Technology has since lowered the costs and simplified the logistics of feeding actual songs into stores, but for this old codger it has taken some of the charm, if you will, out of going to Macys, Sears, or Stop & Shop.  I actually prefer Bacharach to Bieber, and Ferante & Teicher to Fergie or Tamia while patronizing my favorite retail establishments.

For example....


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## bigshot (Jan 10, 2020)

You’ll notice I didn’t say “all directors”. I said “good ones”. Up until the TV era, rough timing was done on bar sheets using a metronome. TV doesn’t do that today. It’s one of the main reasons why old hand drawn animation moves in a more interesting way than cranked out TV animation. Features use temp music and click tracks to establish the beat, and musicals record in advance, track read the music and animate to it.


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## Davesrose (Jan 10, 2020)

bigshot said:


> You’ll notice I didn’t say “all directors”. I said “good ones”. Up until the TV era, rough timing was done on bar sheets using a metronome. TV doesn’t do that today. It’s one of the main reasons why old hand drawn animation moves in a more interesting way than cranked out TV animation. Features use temp music and click tracks to establish the beat, and musicals record in advance, track read the music and animate to it.



That's a valid opinion that "good" film makers do want to at least be able to communicate emotions and themes with the composer they choose, just as good ones also have a good relationship with their cinematographer (if they themselves don't have a background on the technical aspects).  I was merely responding to your previous post with a surprise that not all young film makers are music enthusiasts.

Old animations also have a charm of being hand painted and needing to be shot frame by frame.  While the process has changed with 3D animation, understanding movement of a character hasn't.  Many of the first full length 3D animated movies were directed by people who were either cell to cell animators, or had previously been ones.  I think what's nice about good 3D animation is that it keeps the fundamentals of the old animations: IE studying real life animals, over-emphasize movement....there's even automatic scripts for adding squash and stretch

There are examples of bad animation that doesn't take these factors into account: Polar Express is an example.  I saw a presentation from its animators over how unorthodox Robert Zemeckis was.  He didn't have any experience with animation and thought he could just use motion capture and not worry about framing a shot until after animation.  Apparently, most performances were unusable and animators had to go in afterwards.  There was also thousands of dollars wasted on disposable joint tracking points (placed on the actors).  The animators revealed that Hanks wound up being a better motion actor and assumed other parts as well.

Now with digital, there's just more fluidity with what stages are the basis of timing.  I've also just seen that apparently there are software algorithms for the music producers to sync their music to the video (which I'm sure would be consternation for those who like "natural" timing).


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## bigshot (Jan 10, 2020)

A lot of television animation is made without directors at all. Traditionally, animation directors were responsible for supervising and doing the timing and layouts. Now they have 'show runners", who are basically creative executives. No one does timing and layout. The storyboard artist is responsible for doing all the posing and the animatic editor "parallel parks" the timing to find something that "works". But the synchronization is always much more primitive doing it this way. This is how mediocre animation is made.

Yes, the best CGI is the stuff that adheres to the fundamentals of animation... clear staging, rhythmic movement, caricature and exaggeration, etc. And the least effective are the ones that look at animation as a special effect recreating reality. I've produced a little mocap. We had to go back and completely rework it, animating it pretty much from scratch.

In the past, the action and music were planned at the same time. Once the storyboard was complete, the director and music director worked together to plan the timing on musical bar sheets. Once that was finalized, the composer wrote the score and the director did the layouts and cut exposure sheets for the animators. When the animation was completed, the composer would conduct the music to a click track and it would precisely synchronize with the action on the screen, because it all followed the bar sheets. I'm working on a project right now where we are reverse engineering Chuck Jones's timing to document his production process. If you're interested, PM me and I'll show you what we are doing.


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## Davesrose (Jan 10, 2020)

As another animation producer, you're speaking to the choir about how important storyboards and animatics are.  The Polar Express example is proof of how to waste money and get a wooden performance.  But music hasn't been the only metronome for animation.  Voice talent brings in certain intonation.  I'm also actually having a casual Friday and watching The Absent Minded Professor.  There's a scene where he's bouncing flubber: easy to spot that the flubber elements are animated, and there's no music to go by: but there is a rhythmic timing in animation and folly effects.  I've enjoyed being a participant at Siggraph, and seeing workflows from the major Hollywood productions...and I've tried to educate some of my healthcare clients who have no idea about video production.  I can also assure that with major productions like Pixar movies...there's still "dailies" where the director is seeing every wire frame rendering from an animator and providing input about movement and timing.  I'm more specialized as a general medical animator, and sometimes it's taken me a great amount of time to educate a client about the need of a storyboard to not just drive animation, but help with the linear narration as well (let alone the overall timing).  Actually....I don't think TV/digital has been all bad about natural timing.  There has to be a major emphasis on important keyframes.  If I bemoan the youth: when a younger animator approaches me with a portfolio...anecdotally I've tended to see that their modeling and rendering are fine....but timing can be stiff.

Thanks for your heads up about Chuck Jones.  I'm a child from the 80s, so Don Bluth was my influence about good animation (have re-watched Secret of Nihm and think it really holds up as to how beautifully painted it is).  There are more and more folks wanting to get into animation...so studies about the icons of film animation should have a following: are you going to post anywhere?


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## bigshot

I'm the president of a non-profit educational organization for animators... https://animationresources.org My analysis of Chuck Jones's timing technique is going to be published for members in February. We're putting the finishing touches on it right now. It'll be an e-book, video and an explanatory audio podcast. I don't think there's been much written about timing since John Halas's book back in the 60s (which is worth having). Timing is becoming a lost art. In some ways, as technology moves forward, technique gets forgotten and has to be rediscovered. I'm looking to find a way to mesh the rhythmic control of the past with current technology. I've been working on this for about 20 years, but it is very hard to find out information because just about everyone from the golden age is dead and no one documented the more technical aspects of their work.


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## GearMe

TheSonicTruth said:


> ...Macys, Sears, or Stop & Shop



...what are those? 



TheSonicTruth said:


> For example....




Thanks for that!  I thought my Bossa Nova Ramones was bad 

TBH...listening to that made me think that Peter Sellers or David Niven were going to show up any second!


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## Davesrose (Jan 10, 2020)

bigshot said:


> I'm the president of a non-profit educational organization for animators... https://animationresources.org My analysis of Chuck Jones's timing technique is going to be published for members in February. We're putting the finishing touches on it right now. It'll be an e-book, video and an explanatory audio podcast. I don't think there's been much written about timing since John Halas's book back in the 60s (which is worth having). Timing is becoming a lost art. In some ways, as technology moves forward, technique gets forgotten and has to be rediscovered. I'm looking to find a way to mesh the rhythmic control of the past with current technology. I've been working on this for about 20 years, but it is very hard to find out information because just about everyone from the golden age is dead and no one documented the more technical aspects of their work.



Thanks, I'll look at your site when I have the time!  I'm not as pessimistic about animation fundamentals being lost with new technology.  A 3D animated feature is a different art than VFX for live movies.  I saw a good documentary about Brad Bird....which showed him going through the Disney lots recounting his days as a teen working and learning with the "nine old men", and how he influences current animators.  The previews might no longer be line art....but can be animated wireframes: but the critiques are the same.  I think with animation features, people will always value the art and style that gives more drama.

On the flip side, being a stickler about anatomy and growing up with an interest in models and VFX....I have a different expectations with CGI in VFX.  Everyone says how great the VFX were for Lord of the RIngs....which overall they were, but one scene stuck out with me in which Legolas's arm seemed to grow twice as long for a weird move.  Rendering techniques are improving, and interestingly there are some computer algorithms that are getting good about believable human modeling and expression.  Known as "deepfakes" there's some people with home desktops churning out better CGI human actors than decisions made with a particular team.  People have different expectations with a "realistic" rendering in that it has to look photo realistic and have all the subtle movements of a live actor: animated characters can convey more drama with the over exaggeration and talent of the animator.


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## bigshot

Subtle isn't necessarily better. More expressive is better.


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## Arjey

Could someone explain to me why this works this way? So basically, I downloaded a flac 24/192, and a flac 16/44.1.
Then I converted the 24/192 to 16/44.1, and for some reason, it turned out "worse", upper frequencies cut, less information, and it weighs less than the original 16/44.1 that I downloaded.
16/44.1 original:


Vs 16/44.1 converted from 24/192:


It's even worse than an m4a(vbr) converted from the same 24/192 (which is actually almost identical to the original 16/44.1, a bit worse):

And then there is a m4a(vbr) converted from the original 16/44.1 flac, the worst of all:

So why is an m4a converted from a 24/192 flac better than a 16/44.1 flac converted from the same 24/192? Why is it much better than an m4a converted from an original 16/44.1 flac, which is better than a 44.1 flac you get from converting from a 24/192? So why does it turn out like this (from best to worst): original 16/44.1 flac -> m4a from 24/192 -> m4a from original 16/44.1 flac ≈[?] 16/44.1 flac from 24/192?
Does that mean, that if for example, I want to have flacs (16/44.1, the "best" audible quality for listening to music) on my mp3 player/computer, I need to download original (ripped from CDs straight to flac 16/44.1) for that, and I want to have the best m4a for my phone, I need to download 16/44.1 for player AND 24/192 (not necessarily 24/192, but higher than 16/44.1) to convert to m4a for phone? (I can't just download m4a files, because they are usually mastered for iTunes and sound funky)


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## pinnahertz

Arjey said:


> Could someone explain to me why this works this way? So basically, I downloaded a flac 24/192, and a flac 16/44.1.
> Then I converted the 24/192 to 16/44.1, and for some reason, it turned out "worse", upper frequencies cut, less information, and it weighs less than the original 16/44.1 that I downloaded.
> 16/44.1 original:
> 
> ...


But how do they sound in a blind comparison?


Arjey said:


> So why is an m4a converted from a 24/192 flac better than a 16/44.1 flac converted from the same 24/192? Why is it much better than an m4a converted from an original 16/44.1 flac, which is better than a 44.1 flac you get from converting from a 24/192? So why does it turn out like this (from best to worst): original 16/44.1 flac -> m4a from 24/192 -> m4a from original 16/44.1 flac ≈[?] 16/44.1 flac from 24/192?


There are different processes involved, and without knowing what specific software and settings are being used, it can only be assumed that different processing is being applied to each.

For example, resampling from a high rate to a lower one will reduce high frequency content above and around the new Nyquist frequency.  That essentially involves a filtering process, among other things.  FLAC is lossless, though, so if no resampling is required, then no additional filtering is applied.  However, resampling to a lower sample rate is not lossless, and the filter applied may vary depending on the resampling method used.  In some cases it may be possible that resampling and the applied filter may be user adjustable, but good resampling is not a given, there's good, bad, and ugly.

Reencoding to m4a or mp3 applies more filtering, sometimes fixed, sometimes dynamic, as part of the total bit-rate reduction of the codec.  You'll find mostly mp3 coding takes high frequencies above 15kHz off, for example.  But given a high enough bit-rate setting, the results of m4a coding are generally inaudible.  So again, how does it sound?  Lossy codecs end results are subjective, which means judgement doesn't involve strictly measurement.  A good lossy coded, which m4a is, can have no audible impact if appropriate settings are chosen.  Listen with your ears, not your eyes.


Arjey said:


> Does that mean, that if for example, I want to have flacs (16/44.1, the "best" audible quality for listening to music) on my mp3 player/computer, I need to download original (ripped from CDs straight to flac 16/44.1) for that, and I want to have the best m4a for my phone, I need to download 16/44.1 for player AND 24/192 (not necessarily 24/192, but higher than 16/44.1) to convert to m4a for phone? (I can't just download m4a files, because they are usually mastered for iTunes and sound funky)


You are equating "quality" with visible high frequency in a spectrogram. Spectrograms are not hearing, and tend to exaggerate inaudible differences which people erroneously equate to quality.  There may or may not be a correlation.


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## Arjey

pinnahertz said:


> But how do they sound in a blind comparison?
> There are different processes involved, and without knowing what specific software and settings are being used, it can only be assumed that different processing is being applied to each.
> 
> For example, resampling from a high rate to a lower one will reduce high frequency content above and around the new Nyquist frequency.  That essentially involves a filtering process, among other things.  FLAC is lossless, though, so if no resampling is required, then no additional filtering is applied.  However, resampling to a lower sample rate is not lossless, and the filter applied may vary depending on the resampling method used.  In some cases it may be possible that resampling and the applied filter may be user adjustable, but good resampling is not a given, there's good, bad, and ugly.
> ...


I know that more high frequencies doesn't mean better quality, I just don't want to lose the airyness of some songs, so I want to keep as much information as possible. Regarding what I hear, I don't hear any difference between the original 16/44.1 flac and m4a from 24/192, but they do sound different compared to m4a from original 16/44.1 flac and 16/44.1 flac from 24/192.
I guess I'll just keep experimenting. Does anyone know a good free audio converter (I'm interested in converting flac (16 and 24bit) to m4a and flac 16bit). Maybe the problem is in my converter..


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## castleofargh (Jan 19, 2020)

Arjey said:


> Could someone explain to me why this works this way? So basically, I downloaded a flac 24/192, and a flac 16/44.1.
> Then I converted the 24/192 to 16/44.1, and for some reason, it turned out "worse", upper frequencies cut, less information, and it weighs less than the original 16/44.1 that I downloaded.
> 16/44.1 original:Vs 16/44.1 converted from 24/192:
> It's even worse than an m4a(vbr) converted from the same 24/192 (which is actually almost identical to the original 16/44.1, a bit worse):And then there is a m4a(vbr) converted from the original 16/44.1 flac, the worst of all:So why is an m4a converted from a 24/192 flac better than a 16/44.1 flac converted from the same 24/192? Why is it much better than an m4a converted from an original 16/44.1 flac, which is better than a 44.1 flac you get from converting from a 24/192? So why does it turn out like this (from best to worst): original 16/44.1 flac -> m4a from 24/192 -> m4a from original 16/44.1 flac ≈[?] 16/44.1 flac from 24/192?
> Does that mean, that if for example, I want to have flacs (16/44.1, the "best" audible quality for listening to music) on my mp3 player/computer, I need to download original (ripped from CDs straight to flac 16/44.1) for that, and I want to have the best m4a for my phone, I need to download 16/44.1 for player AND 24/192 (not necessarily 24/192, but higher than 16/44.1) to convert to m4a for phone? (I can't just download m4a files, because they are usually mastered for iTunes and sound funky)


First we need to get one important notion cleared out. The strong blue on those images are for signals at -90dB!!!!!!!!!!!!!!!!!!! I don't even go that low when setting the spectrogram range in audicity.
Edit:
Then this specific track or sample of track doesn't seem to show any signal before about -25dB and that's perhaps the weirdest thing to see here. Unless you specifically picked a quiet passage in the track, it doesn't make much sense for any digital track to just waste that headroom.
(edit: this is wrong, I fall for that often and never seem to learn, sorry). 


Now about what you saw going wrong/strange. I would guess that the overall gain is down on the second pic, and most of the mystery is going to be for you to find why.
Somehow the downsampling seems to apply a filter that starts at 18kHz. In the digital domain, there is no real need for that IMO. TBH it would make more sense to me if this was the output of a DAC instead of a converted file.


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## Arjey

castleofargh said:


> First we need to get one important notion cleared out. The strong blue on those images are for signals at -90dB!!!!!!!!!!!!!!!!!!! I don't even go that low when setting the spectrogram range in audicity.
> Then this specific track or sample of track doesn't seem to show any signal before about -25dB and that's perhaps the weirdest thing to see here. Unless you specifically picked a quiet passage in the track, it doesn't make much sense for any digital track to just waste that headroom.
> 
> Now about what you saw going wrong/strange. I would guess that the overall gain is down on the second pic, and most of the mystery is going to be for you to find why.
> Somehow the downsampling seems to apply a filter that starts at 18kHz. In the digital domain, there is no real need for that IMO. TBH it would make more sense to me if this was the output of a DAC instead of a converted file.


I see what your saying about the "blue parts", I know that I can't really hear stuff that are so quiet. But it does change what I hear. I mean, the fact that the driver of my earphone is trying to make that sound does affect the overall sound, if you understand what I mean. If you record a pluck of a guitar, and then cut out all sounds under -70 it will sound very different to the unedited version. So I just want to keep as much info as possible.

The color interpretation is different from Spek (the PC app I use), so take that into account when looking at dB. In Spek, for examle, bright green is ≈-65, here it's -50. The song it's quite. But the app is mobile, so it may be quite a bit wrong interpretating volume. My PC monitor broke, can't use Spek(


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## bigshot (Jan 18, 2020)

Do a blind listening comparison of the files. I bet blind they all sound the same. I've never been able to judge sound quality by charts. I can only spot obvious problems. Better to judge sound by listening.


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## castleofargh

Arjey said:


> I see what your saying about the "blue parts", I know that I can't really hear stuff that are so quiet. But it does change what I hear. I mean, the fact that the driver of my earphone is trying to make that sound does affect the overall sound, if you understand what I mean. If you record a pluck of a guitar, and then cut out all sounds under -70 it will sound very different to the unedited version. So I just want to keep as much info as possible.
> 
> The color interpretation is different from Spek (the PC app I use), so take that into account when looking at dB. In Spek, for examle, bright green is ≈-65, here it's -50. The song it's quite. But the app is mobile, so it may be quite a bit wrong interpretating volume. My PC monitor broke, can't use Spek(


The color code doesn't matter, it's provided on the side of your graphs.
I'm not judging you or telling you to like having stuff low passed at 18kHz for no reason. You obviously should try to get what you wish to get and most people get, which is something low passed harder but also at a frequency closer to samplerate/2. What most converters will do TBH, you just got unlucky with yours or some of the settings in it.

About extra stuff and their impact: 1. If the extra content is at a level and a frequency that is audible to you(not completely masked by louder sounds, and not outside of frequencies you can still perceive), then you hear it and it affects your overall impression of sound. How much depending on how loud and what sound it is.
2. The signal itself is not audible but the headphone is terribly nonlinear with horrible amounts of distortions. Then maybe you'd get to hear a change from the garbage created by the extra yet inaudible signal changing the driver's movements. That I think is what you're describing and it should not have audible consequences as for most headphones an already quiet signal would only generate extra content as distortions at yet another 40 or 50dB below the quiet sound. Reaching inaudible level in most circumstances with even rather average headphones.
3. But in case you're speaking about a concept "à la" analogsurviver and his ultrasonic content being not audible by themselves but being audible as part of a bigger global sound element of sort(I don't think you meant that but I'm just trying to be exhaustive). That is wrong. Wave theory says so, and if things were that "simple", it would be trivial for anybody to pass a blind test between 44.1 and higher sample rates. Or maybe between a track and the same track plus noise at -90dB. Except that such tests are rarely successful and pretty much never with reasonably mastered test track and reasonable listening levels. Meaning that both theory and practical experiments disprove the idea.

In any case, I insist on being with you when you want to keep as much info as possible. I see nothing wrong with that desire. You should try some other converter for your 44.1 flac as something is wrong with the result anyway. I've been using SOX for many years now so I don't know anything else and hope others will have suggestions for something maybe more intuitive. Because SOX is cool and very customizable, but the old school command line thingy isn't super attractive or intuitive at first contact.



bigshot said:


> Do a blind listening comparison of the files. I bet blind they all sound the same. I've never been able to judge sound quality by charts. I can only spot obvious problems. Better to judge sound by listening.


 It's almost impossible to see anything the way the graphs show up in the post, but if you open the first 2 graph in 2 tabs and switch between them, the differences are hard to miss. The 16/44flac version is clearly suspicious and seems to be quieter by a good margin, plus the signal seems low passed near 18kHz. which is strange for a digital conversion to 44.1kHz. So it probably does not sound the same as the original if only because of the loudness difference(or whatever it is I mistook for gain change).


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## bigshot

Yeah, it's easy to tell that you can't tell.


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## 71 dB

castleofargh said:


> First we need to get one important notion cleared out. The strong blue on those images are for signals at -90dB!!!!!!!!!!!!!!!!!!! I don't even go that low when setting the spectrogram range in audicity.
> Then this specific track or sample of track doesn't seem to show any signal before about -25dB and that's perhaps the weirdest thing to see here. Unless you specifically picked a quiet passage in the track, it doesn't make much sense for any digital track to just waste that headroom.



It's a spectrogram so the levels indicate spectral density. The more detailed spectrogram (bigger FFT), the more frequency points and the lower the spectral density, because the signal energy is divided to more frequency points. that's why "yellow" (or about -30 dB) is the loudest we see. The blue is down at around -90 dB, but you need to integrate all those blue points on the frequency line to get the signal level at say 15 kHz to 20 kHz. So if the FFT size is say 1024, the amount of positive frequency points is 512 meaning the frequency range 15-20 kHz has 512 * (20-15) kHz/22.05 kHz = 116 frequency points. If all of these are at spectral density level -90 dB, the signal level at frequency band 15-20 kHz is -90 dB + 10*log10 (116) = -69 dB. That's still very low level considering how insensitive hearing is at these frequencies! The frequency line here is linear (far from how human hearing works) and makes the blue stuff more dramatic it is for human ear. The upper half (11-22 kHz) is just one octave and on logarithmic scale ~1/10 of the picture (if 20 Hz - 22050 Hz plotted)!


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## pinnahertz

Arjey said:


> [/i]I know that I can't really hear stuff that are so quiet. But it does change what I hear.


Sorry, both cannot be true, it's one or the other.


Arjey said:


> I mean, the fact that the driver of my earphone is trying to make that sound does affect the overall sound, if you understand what I mean.


I know what you're trying to say, but I don't think you know what you mean.  Or at least, you have an incorrect concept.


Arjey said:


> If you record a pluck of a guitar, and then cut out all sounds under -70 it will sound very different to the unedited version.


That depends on what you mean by "cut out all sounds under -70".  I'm not even sure what you mean, or how that would be done, or why.


Arjey said:


> So I just want to keep as much info as possible.


Right.  But you're going to m4a, so that actually dumps as much information as possible.  That's the entire goal of lossless codecs, to remove as much information as possible and still have an acceptable result.


Arjey said:


> I
> The color interpretation is different from Spek (the PC app I use), so take that into account when looking at dB. In Spek, for examle, bright green is ≈-65, here it's -50. The song it's quite. But the app is mobile, so it may be quite a bit wrong interpretating volume. My PC monitor broke, can't use Spek(


Really, do yourself a big favor and stop looking at spectrograms.  They do not equate to audibility.  If you study them and see a "huge difference" by your definition, you will strongly bias your perception of the actual result.  As others have noted, they will display information clearly all the way down to the noise floor. Just don't do it.


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## bigshot

The spectral display looks very pretty.


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## gregorio

Arjey said:


> [1] Then I converted the 24/192 to 16/44.1, and for some reason, it turned out "worse", upper frequencies cut, less information, and it weighs less than the original 16/44.1 that I downloaded.
> [1a] I know that more high frequencies doesn't mean better quality,
> [1b] I just don't want to lose the airyness of some songs,
> [1c] so I want to keep as much information as possible.



1. Why is it "worse" to have "_upper frequencies cut, less information, and it weights less_"? I don't know what you mean by "weights less" but "upper freqs cut" and "less information" isn't worse, it's better!!! Because of the following points:
1a. Correct. In fact, more high freqs can often mean worse quality as it's liable to cause distortion (IMD).
1b. "Airyness" exists in roughly the 8kHz - 16kHz range and mainly between about 10kHz-14kHz. Therefore:
1c. Why would you want to keep information significantly above that range? 


Arjey said:


> [1] I mean, the fact that the driver of my earphone is trying to make that sound does affect the overall sound,
> [1a] if you understand what I mean.
> [2] If you record a pluck of a guitar, and then cut out all sounds under -70 it will sound very different to the unedited version.
> [2a] So I just want to keep as much info as possible.
> [3] I can't just download m4a files, because they are usually mastered for iTunes and sound funky.



1. But that's NOT a fact, the driver of your earphone is NOT trying to make "that sound"! For example, let's say you have sounds down at -80dB, the noise floor of the recording is say -60dB and you're listening at a peak level of 70dBSPL, your earphones are not trying to make those sounds down at -80dB, they're just making noise that does NOT affect the overall sound.
1a. Not really. Exactly what the driver of your earphone is "trying to make" is dependant on a number of factors. Additionally, A. "Trying to make" and actually making are not necessarily the same thing (IMD for example) and B. Even if they succeed and the overall sound is affected, that's no guarantee that the difference is in any way audible.

2. That is simply NOT true. If one were to cut all sound below -70dB in the vast majority of reasonable reproduction scenarios it would sound absolutely no different whatsoever, let alone "very" different! Even in some fairly extreme (but still reasonable) reproduction scenarios where the difference *might* be audible, it would still only sound slightly different, not very different.
2a. Again, why? What benefit do you think you gain from keeping info that's inaudible?

3. As the name suggests, "Mastered for iTunes" (MFiT) means mastered specifically for AAC 256vbr. Assuming you download the original AAC 256vbr file, you are getting an exact bit perfect copy of what the mastering engineer released/intended. So, unless you have some serious fault with your equipment, if it "sounds funky" then it's supposed to sound funky and why would you want to change "funky" music into something else? 

It's pretty clear that you don't really understand what you're looking at with a spectrogram and how it relates to either what your earphones are "trying to make" or what is actually audible, as others have effectively stated. Furthermore, as @pinnahertz stated, unless you are doing your "experimenting" using ABX testing (at reasonable playback levels), then all you are accomplishing is the creation of a "strong perception bias" and FALSE results, that completely void your experiments!

G


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## castleofargh (Jan 19, 2020)

71 dB said:


> It's a spectrogram so the levels indicate spectral density. The more detailed spectrogram (bigger FFT), the more frequency points and the lower the spectral density, because the signal energy is divided to more frequency points. that's why "yellow" (or about -30 dB) is the loudest we see. The blue is down at around -90 dB, but you need to integrate all those blue points on the frequency line to get the signal level at say 15 kHz to 20 kHz. So if the FFT size is say 1024, the amount of positive frequency points is 512 meaning the frequency range 15-20 kHz has 512 * (20-15) kHz/22.05 kHz = 116 frequency points. If all of these are at spectral density level -90 dB, the signal level at frequency band 15-20 kHz is -90 dB + 10*log10 (116) = -69 dB. That's still very low level considering how insensitive hearing is at these frequencies! The frequency line here is linear (far from how human hearing works) and makes the blue stuff more dramatic it is for human ear. The upper half (11-22 kHz) is just one octave and on logarithmic scale ~1/10 of the picture (if 20 Hz - 22050 Hz plotted)!


You're right. I forget not to just take every value written at face value. In one form or another I've made that very mistake at least 3 times in the last year. Last time was on a RTA counting the stuff on the graph as noise level, even though the app would have given me the correct value so long as I bothered to click on a given frequency. So even when I got all the work done for me, even after reading the great stuff from Audio Precision dozens of time, and even after knowing that I make that mistake often, I still manage to fall for it from time to time. 
Thank you for pointing it out. @bigshot too, even if it wasn't as detailed an explanation


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## vatch

It must not be forgotten that very few instruments even make much noise at all with any real power at or above 20khz; a 96khz audio frequency is absolutely unnecessary and will mostly be very high frequency noise with no musicality even though you can't hear it anyway.

The below chart, "THE FREQUENCY SPECTRUM, INSTRUMENT RANGES, AND EQ TIPS" lists the frequency ranges of many popular instruments for reference:

http://www.guitarbuilding.org/wp-content/uploads/2014/06/Instrument-Sound-EQ-Chart.pdf


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## pinnahertz

vatch said:


> It must not be forgotten that very few instruments even make much noise at all with any real power at or above 20khz; a 96khz audio frequency is absolutely unnecessary and will mostly be very high frequency noise with no musicality even though you can't hear it anyway.
> 
> The below chart, "THE FREQUENCY SPECTRUM, INSTRUMENT RANGES, AND EQ TIPS" lists the frequency ranges of many popular instruments for reference:
> 
> http://www.guitarbuilding.org/wp-content/uploads/2014/06/Instrument-Sound-EQ-Chart.pdf



As much as I agree with your first statement, the attached chart is very misleading.  It apparently charts only the sounds fundamental, not any harmonic content.  It also implies that above the charted frequency no content exists at all, which is completely incorrect.  If you want to prove this, just make a recording of your own voice and filter it with the most basic low pass filter at the charted cut-off frequency.  

The harmonic content of a sound of any kind gives it its distinctive character.  The point here is that no sounds, harmonics or fundamentals, can be heard above the upper limit of human hearing, with the corollary that human hearing response is not flat with a brick wall cut-off either, but displays a filter roll-off characteristic that starts lower into the "audible" range, and is affected by specific sound pressure and masking principles.   But even with all of that said, 96kHz sampling with 48kHz pass band is not necessary to replicate the original signal.


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## 71 dB

castleofargh said:


> 1 --- You're right.
> 
> 2 --- I forget not to just take every value written at face value. In one form or another I've made that very mistake at least 3 times in the last year.
> 
> ...



1 --- It's not often I get this, but thanks! I appreciate it. My years in the university weren't time wasted after all. 

2 --- It's easy to make these mistakes. It's good to always ask yourself "does this make sense?" Vinyl enthusiasts have tried to demonstrate vinyl to have_ larger_ dynamic range than CD with this mistake: Because of the properties of spectral density and vinyl (RIAA curves etc.) the noise floor spectral density level drops easily well below -100 dB at higher frequencies and then people compare that to the CD dynamic range of about 90 dB or so without realising that for CD the same spectral density of noise floor (dither) drops insanely low and is possibly only higher at the highest frequencies if shaped dither is used (meaning at the lower frequencies the noise floor is even lower!) 

3 --- Errare humanum est. 

4 --- No problem.


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## 71 dB

vatch said:


> It must not be forgotten that very few instruments even make much noise at all with any real power at or above 20khz; a 96khz audio frequency is absolutely unnecessary and will mostly be very high frequency noise with no musicality even though you can't hear it anyway.
> 
> The below chart, "THE FREQUENCY SPECTRUM, INSTRUMENT RANGES, AND EQ TIPS" lists the frequency ranges of many popular instruments for reference:
> 
> http://www.guitarbuilding.org/wp-content/uploads/2014/06/Instrument-Sound-EQ-Chart.pdf



Years ago I read about a listening test were people evaluated how much in their opinion sound quality dropped when 44.1 kHz audio was downsampled to 22.05 kHz cutting everything above 10 kHz away and the conclusion was surprisingly little! Here it's about frequencies people definitely hear (10-16 kHz). That's why it's silly how much some people worry about totally inaudible frequencies above 20 kHz.


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## vatch

Yeah, I agree.  It does simplify it a bit much and there are things going on in the upper range.  The main thing really is that audio reproduction above 20khz and also below 20hz for that matter is unnecessary and highly likely detrimental as there is more computation and possibility of error in addition to increased high frequency electronically induced noise.  In essence, most speakers can't reproduce it anyway.


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## bigshot

I don't think there's much in the way of harmonics above 10kHz that wouldn't be masked by lower frequencies. If you roll off everything above 10kHz, you can hear a slight difference, but it doesn't amount to a hill of beans in the grand scheme of things. But audiophiles are known to focus on the hills of beans and ignore the grand scheme.


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## Vamp898 (Mar 5, 2020)

To finally beat this topic to death.

I compared an 24bit/96KHz Orchestra recording (Song called 39 from the Tokyo Philharmony Orchestra from the Album Miku Symphony 2019) with the same recording in 16bit/44.1KHz

I inverted the 16bit/44.1Khz Song and mixed them together, the result was, of course, the difference.

What was the difference? At the point where the biggest difference was, there was an -57db loud(""loud"") sound at 22-45KHz

I was curious what this sound was so i lowered the frequency and there you go: It was white noise.

So if you choose for CD-Quality and not choose the Hi-Res Audio, you are missing out on -57db loud ultrasonic white noise.

I did the same with an 24bit/44.1KHz Electronic Song (Naomi from LukHash) and the loudest difference was an -86db loud white noise at 20KHz

Hope this makes you happy and this case is closed once and for all.

By the way, the difference from an OGG to an FLAC is way higher. If you do the same i mentioned above to an OGG at Quality 10 (comparable to an 400kbit/s MP3 if that would exist), you can here almost the whole song in the difference. Way quiter than the original recording of course and in way worse quality (sounds like an 64kbit/s MP3) but there is the whole song there.

The biggest difference, with the equalizer tuned to my personal taste, was the whole song but at -36db and bad quality. The difference changes drastically from song to song and, if you use an equalizer, there are parts of the songs where, when you know the difference, it could be heard if you turn the song loud enough. I played the MP3 and then enabled/disabled the difference and at some parts, especially in the low frequencies, there can be a difference. But chances are small anyone will ever recognize it comparing the songs side by side. I could only recognize it enabling the differences, i could not tell a difference when comparing the whole song.

Anyway I'd always go for FLAC because, at least for me, this test showed an big enough difference that i dont want to risk it. At this level of difference, chances for the decoder to ass things up are high imho and my Audio Player Supports 2TB MicroSD Cards and for the 128GB Card i have in there, i payed 13€ so whatever, who cares.

I totally agree with the initial poster of this thread. You will never hear the difference from an 16bit/44.1KHz to any higher quality recording, no matter what you do. Its an ultra sonic white noise, trust me, you'll never hear that and don't want it.

Last but not least, i noticed that the difference is bigger when you download a Song that is available in CD-Quality and Hi-Res Audio. The difference is still to small to be ever heard, but somehow if i downscale the song myself, the difference is smallet than buying a CD Quality Audio in the first place. But this is not always the case, I only found one album with that from what I checked so far.

Because the price difference for an 20€ album is roughly 3-5€, i still buy the hi-res audio songs just because im to afraid someone screwed up the downscaling, but this is the only reason. There is no technical difference for an human being.


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## themysticmuse

It's 03:17 am here in Sweden and I've been reading this thread for the last 3 hours. This is the best thing since sliced bread. Thanks OP, I've learnt a lot.


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## bigshot (Jun 30, 2020)

Vamp898 said:


> I inverted the 16bit/44.1Khz Song and mixed them together, the result was, of course, the difference. What was the difference? At the point where the biggest difference was, there was an -57db loud(""loud"") sound at 22-45KHz I was curious what this sound was so i lowered the frequency and there you go: It was white noise.



You were probably blowing up the air conditioning room tone for the recording venue. CD noise floor is ridiculous overkill. You can't even record in a recording studio with a room tone below -95dB. You'd have to be in Carlsbad Caverns or an anechoic chamber or something. The room tone in your living room is probably higher than 40dB. So to play everything possible on a CD at an audible level, you would need to boost the level above that living room noise floor to 40+95=125dB. You're in the range of causing permanent hearing damage in a very short space of time there. Don't try this at home, kids!


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## theaudiologist1

Man this thread is still going after more than a decade. Wow!

Anyways, none of this matters because 1 bit DSD wipes the floor with PCM.


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## chef8489

theaudiologist1 said:


> Man this thread is still going after more than a decade. Wow!
> 
> Anyways, none of this matters because 1 bit DSD wipes the floor with PCM.


As you are in sound science how about you post scientific evidence to back up your claim. You do know that 99% of dsd was recorded in pcm and converted right. There are very few studios that have the equipment to do everything in dsd.


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## bigshot (Jul 2, 2020)

theaudiologist1 said:


> Anyways, none of this matters because 1 bit DSD wipes the floor with PCM.



Ten years and you still haven't read it!



chef8489 said:


> There are very few studios that have the equipment to do everything in dsd.



You can't edit in DSD, so most studios record in DSD, bump to high rate PCM to edit and mix, then back to DSD. It's all overkill. Donald Fagen's The Nightfly is one of the best sounding albums I've ever heard and it was recorded and mixed in plain vanilla 16/44.1. Then they released it on SACD! haha


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## theaudiologist1

chef8489 said:


> As you are in sound science how about you post scientific evidence to back up your claim. You do know that 99% of dsd was recorded in pcm and converted right. There are very few studios that have the equipment to do everything in dsd.



also the other way around as well. Modern DAC's are all (or mostly) delta-sigma DAC's which are essentially DSD, so you don't even hear "true" PCM either. Both fields convert to another throughout the process
It's true that most releases are recorded in PCM. However, since I listen to a lot of classical, a lot of classical is recorded in Native DSD, and even analog-to-DSD has an advantage provided that it doesn't convert to PCM in the process. Also due to the delta-sigma DAC's, upsampling/oversampling PCM to DSD helps reduce one conversion process.


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## bigshot

As long as you are using an audibly transparent format and you aren't processing it enough to bump into the limits, it doesn't matter what format you use. The reason your SACDs sound good is because the way they are recorded, not the type of file format they use. One of the best classical recordings I've ever heard was recorded in 1952.


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## chef8489

"Despite the marketing hype, there are almost no pure DSD recordings available to consumers. This is partially because up until quite recently there was no way to edit, mix, and master DSD files. So most pure DSD recordings that are commercially available are the rare DSD recordings made from a direct-to-analog recording, or those recorded direct to DSD without any post-production. There are some new studio software packages that can edit, mix, and master in DSD, but these are quite rare in the industry, and mostly used by small boutique recording companies. Most DSD recordings are, in fact, edited, mixed, and mastered in 5-bit PCM (aka Wide-DSD). The marketing hype DSD flow chart you see below rarely exists anywhere but in theory. Yikes…the secret is out. "

"There are several generations and levels of quality in purely digital DSD recordings. The least pure are DSD recordings made from old PCM masters. Many of these PCM masters had low resolution, as well as significantly higher quantization errors and lower linearity than modern PCM recordings. Since you can never get better than the original masters, these DSD recordings sound as bad as or worse than the original low-resolution PCM masters. The purest common DSD recordings come from modern DSD masters that are recorded in Wide-DSD, which is in fact a 5-bit or 8-bit PCM format at ultrahigh DSD sampling rates. Wide-DSD is what most recordings studios are currently using. "

"
most commercially available DSD recordings have to be converted back and forth to a PCM format in order to do post-production editing, mixing, and mastering. In each of these conversions, more quantization noise and/or quantization errors are added to the recording. This leads many to ask: why degrade performance by adding the additional step to convert to DSD when the master is already in PCM?

It is quite unlikely that any or many of recording studios that are currently using Wide-DSD for editing, mixing, and mastering will ever upgrade to software that can edit, mix, and master in true DSD, since DSD is in fact an obsolete format. Even Sony no longer supports DSD. The modern format that recording studios will likely be upgrading to is MQA, a 24-bit 192KHz PCM compression format that requires significantly less bandwidth than normal PCM to stream. That is why HD music streaming services such as  Roon and Tidal are switching over to MQA for their ultra-HD selections. So with the invention of MQA compression, PCM is quickly becoming the preferred HD music format.

Another common marketing myth about DSD vs. PCM is that when blind listening tests were done comparing DSD to PCM, there was a consensus that PCM had a fatiguing quality and DSD had a more analog-like quality. This was proved to be total marketing BS. One way that marketing lie was perpetuated was with hybrid SACDs that have DSD64 and 16-bit 44.1KHz PCM on the same disk. The DSD64 tracks have roughly 33 times the resolution of the 16-bit 44.1KHz tracks so that they could make DSD sound better than PCM in comparisons. The truth is that in recent blind studies they've proved that high-resolution PCM and DSD are statistically indistinguishable from one another. Considering that nearly all DSD recordings were edited, mixed, and mastered in PCM, it is no wonder.

Then there are the differences in the ways DAC chips work. Most modern DAC chips are single-bit or Sigma Delta. Most modern single-bit DAC chips can decode multiple file formats, including PCM, DSD, and Wide-DSD. Of course when they are decoding PCM, a single-bit DAC chip has to first convert it into DSD, the chip's native format. Another reason for the common misconception that DSD performs better than PCM has to do with the poor quality of the real-time PCM to DSD converters built into native DSD single-bit DAC chips.

On the other hand, there are multi-bit R-2R ladder DAC chips. Few companies still manufacture multi-bit DAC chips anymore because they are so much more expensive to manufacture than single-bit DAC chips. Multi-bit DAC chips are optimized for and can only decode PCM formats. Of course there are some DACs that use multi-bit DAC chips with FPGA input stages that convert DSD to PCM, but the multi-bit DAC chips themselves can not decode DSD.

In almost all cases I would recommend playing music files in the native format that your DAC chip decodes. That would be PCM for a multi-bit DAC chip and DSD for a single-bit DAC chip. There are several brands of player software on the market that have real-time PCM to Double-Rate DSD converters. HQ Player is one of the most sophisticated player software packages on the market today. HQ Player can be configured for real-time PCM to DSD conversion as well as real-time DSD upsampling to Double, Quad, Octuple, and even higher rate DSD formats. Using player software that is capable of converting PCM to DSD and upsampling it to at least Quad-Rate DSD is highly recommended."


----------



## bigshot (Jul 2, 2020)

chef8489 said:


> Despite the marketing hype, there are almost no pure DSD recordings available to consumers.



Pentatone does native DSD. I did a controlled listening test on the Pentatone Pavo Jarvi Stravinsky chamber music SACD. I compared the native DSD layer to the Redbook layer. I did the comparison myself, and then a sound mixer friend let me do a test on his reference system. Neither of us could hear any difference at all between the SACD and the redbook. They sounded identical. I've also compared running HDMI out PCM and analogue out native from my Oppo blu-ray player. Couldn't hear any difference there. It's all perfect for human ears.

I think a lot of the reports of DSD sounding better are due to 1) better mastering (I found a number of SACDs that had different mastering on the redbook layer, and one where it was a completely different mix!) and 2) good old fashioned expectation bias.


----------



## chef8489

bigshot said:


> Pentatone does native DSD. I did a controlled listening test on the Pentatone Pavo Jarvi Stravinsky chamber music SACD. I compared the native DSD layer to the Redbook layer. I did the comparison myself, and then a sound mixer friend let me do a test on his reference system. Neither of us could hear any difference at all between the SACD and the redbook. They sounded identical. I've also compared running HDMI out PCM and analogue out native from my Oppo blu-ray player. Couldn't hear any difference there. It's all perfect for human ears.
> 
> I think a lot of the reports of DSD sounding better are due to 1) better mastering (I found a number of SACDs that had different mastering on the redbook layer, and one where it was a completely different mix!) and 2) good old fashioned expectation bias.


Ik there are a couple of studios that can record in dsd but it is very few. Even fewer that can edit in pure dsd. No i dont believe there is a difference as bit perfect is bit perfect. There is so much in marketing and hype out there so people can sell.


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## theaudiologist1 (Jul 2, 2020)

chef8489 said:


> "Despite the marketing hype, there are almost no pure DSD recordings available to consumers. This is partially because up until quite recently there was no way to edit, mix, and master DSD files. So most pure DSD recordings that are commercially available are the rare DSD recordings made from a direct-to-analog recording, or those recorded direct to DSD without any post-production. There are some new studio software packages that can edit, mix, and master in DSD, but these are quite rare in the industry, and mostly used by small boutique recording companies. Most DSD recordings are, in fact, edited, mixed, and mastered in 5-bit PCM (aka Wide-DSD). The marketing hype DSD flow chart you see below rarely exists anywhere but in theory. Yikes…the secret is out. "
> 
> "There are several generations and levels of quality in purely digital DSD recordings. The least pure are DSD recordings made from old PCM masters. Many of these PCM masters had low resolution, as well as significantly higher quantization errors and lower linearity than modern PCM recordings. Since you can never get better than the original masters, these DSD recordings sound as bad as or worse than the original low-resolution PCM masters. The purest common DSD recordings come from modern DSD masters that are recorded in Wide-DSD, which is in fact a 5-bit or 8-bit PCM format at ultrahigh DSD sampling rates. Wide-DSD is what most recordings studios are currently using. "
> 
> ...



I basically said the same thing: that most DSD is recorded in PCM. That, I agree, has no "direct" advantage (PCM upsampled to DSD on the go with some software might push the noise more outside the hearing range so that's a plus) over PCM. But native DSD and even analog-straight-to-DSD without any convertion does sound different and closer to analog than PCM, mainly due to the sampling rate. DSD has no "resolution" (resolution = bit depth) but makes up for it with a much higher sampling rate
.

Personally, DSD sounds more flat and natural (albeit a bit more quiet) than PCM, which has a lot more "oomph" to the sound. This is considering both my Native DSD and PCM files, and also listening to PCM without convertion and with DSD upsampling.

TLDR on mastering, mixing and editing:
Native DSD = great
Analog-to-DSD without PCM convertion = good
PCM-to-DSD = useless and even makes it worse

DSD isn't going anywhere. Just recently it got a resurgence among many audiophiles without the need of SACD's. Newer, higher sampling rate DSD are becoming available (DSD128, DSD256, DSD512 and even DSD1024). If Sony doesn't care about DSD, why do their DAC's and DAP's continue to support them?


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## bigshot (Jul 2, 2020)

theaudiologist1 said:


> But native DSD and even analog-straight-to-DSD without any convertion does sound different and closer to analog than PCM



You are in Sound Science now. You can say whatever you want in the rest of Head-Fi, but here I get to ask you for a controlled listening test that backs up what you say. I’ve done one that shows that native DSD sounds EXACTLY like PCM 16/44.1. I say you are basing your opinion on expectation bias and placebo effect. Do you have evidence to back up your statement, because there’s plenty to show you are wrong.


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## Sonic Defender (Jul 2, 2020)

Removed, redundant


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## Sonic Defender (Jul 2, 2020)

theaudiologist1 said:


> I basically said the same thing: that most DSD is recorded in PCM. That, I agree, has no "direct" advantage (PCM upsampled to DSD on the go with some software might push the noise more outside the hearing range so that's a plus) over PCM. But native DSD and even analog-straight-to-DSD without any convertion does sound different and closer to analog than PCM, mainly due to the sampling rate. DSD has no "resolution" (resolution = bit depth) but makes up for it with a much higher sampling rate
> .
> 
> Personally, DSD sounds more flat and natural (albeit a bit more quiet) than PCM, which has a lot more "oomph" to the sound. This is considering both my Native DSD and PCM files, and also listening to PCM without convertion and with DSD upsampling.
> ...


No, no, no, ridiculous please try to understand why these claims are akin to Trump calling investigative journalism fake news.


----------



## gregorio

theaudiologist1 said:


> Anyways, none of this matters because 1 bit DSD wipes the floor with PCM.



Unfortunately you've got that exactly backwards, it should read: PCM "_wipes the floor_" with 1 bit DSD! 

Sure, 1 bit DSD is *marketed* as higher fidelity but *the science* proves that it's actually lower fidelity and of course, this is the Sound Science subforum, not just another marketing subforum!  Therefore, here's the science: "_Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications_" - Stanley P. Lipshitz and John Vanderkooy:
"_Single-stage, 1-bit sigma-delta converters are in principle imperfectible. We prove this fact. ...  In contrast, multi-bit sigma-delta converters, which output linear PCM code, are in principle infinitely perfectible. ...  The audio industry is misguided if it adopts 1-bit sigma-delta conversion as the basis for any high-quality processing, archiving, or distribution format to replace multi-bit, linear PCM._"

However, "_wipes the floor_" is effectively a relative and purely technical issue: Because the serious flaw/issue with 1 bit DSD is inaudible in practical useA (music recording).

And lastly, as @chef8489 correctly stated, the vast majority of SACD/1 bit DSD recordings are actually PCM converted to 1 bit DSD.



chef8489 said:


> [1] As you are in sound science how about you post scientific evidence to back up your claim.
> [2] You do know that 99% of dsd was recorded in pcm and converted right.
> [2a] There are very few studios that have the equipment to do everything in dsd.



1. Totally agree!

2. As far as I'm aware, the majority or vast majority of DSD recordings were actually recorded in DSD. However, the raw DSD recordings are then converted to PCM for mixing, processing and mastering. So, the only truly 1 bit DSD releases are a relatively tiny handful of "direct to disk" recordings which have NOT been digitally mixed, processed or mastered.
2a. There are NO STUDIOS AT ALL that have the equipment to do everything in 1 bit DSD, because it's a technical impossibility to mix/process 1 bit DSD and therefore the "_equipment to do everything_" in 1 bit DSD simply doesn't exist, hence why it HAS to be converted to PCM for mixing/processing/mastering. Having said this, I believe that a couple of years or so ago a commercial system was released which allowed "everything in DSD" but not 1 bit DSD. As I recall, 1 bit DSD is converted to 8 bit DSD, processed and then converted back to 1 bit DSD. For this reason, as your statement just said "dsd" (not specifically 1 bit DSD), then technically it's correct. However, it could only apply to recordings made in the last couple/few years, still requires a conversion in the case of 1 bit DSD (to 8 bit DSD and back again) and AFAIK, this system is not commonly used and the majority of today's 1 bit DSD releases still involve converting to PCM.

BTW, I'm not really disputing what you stated @chef8489, just being a bit more precise.

G


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## chef8489

gregorio said:


> Unfortunately you've got that exactly backwards, it should read: PCM "_wipes the floor_" with 1 bit DSD!
> 
> Sure, 1 bit DSD is *marketed* as higher fidelity but *the science* proves that it's actually lower fidelity and of course, this is the Sound Science subforum, not just another marketing subforum!  Therefore, here's the science: "_Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications_" - Stanley P. Lipshitz and John Vanderkooy:
> "_Single-stage, 1-bit sigma-delta converters are in principle imperfectible. We prove this fact. ...  In contrast, multi-bit sigma-delta converters, which output linear PCM code, are in principle infinitely perfectible. ...  The audio industry is misguided if it adopts 1-bit sigma-delta conversion as the basis for any high-quality processing, archiving, or distribution format to replace multi-bit, linear PCM._"
> ...


No worries I didn't read it as disputing. Just clarifying it.


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## bigshot

I guess he's gone now. Not a very good job of trolling and an even worse job of defending his position if he was actually serious.


----------



## theaudiologist1

gregorio said:


> Unfortunately you've got that exactly backwards, it should read: PCM "_wipes the floor_" with 1 bit DSD!
> 
> Sure, 1 bit DSD is *marketed* as higher fidelity but *the science* proves that it's actually lower fidelity and of course, this is the Sound Science subforum, not just another marketing subforum!  Therefore, here's the science: "_Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications_" - Stanley P. Lipshitz and John Vanderkooy:
> "_Single-stage, 1-bit sigma-delta converters are in principle imperfectible. We prove this fact. ...  In contrast, multi-bit sigma-delta converters, which output linear PCM code, are in principle infinitely perfectible. ...  The audio industry is misguided if it adopts 1-bit sigma-delta conversion as the basis for any high-quality processing, archiving, or distribution format to replace multi-bit, linear PCM._"
> ...




I already DID say, and was aware, that 90% of DSD recordings were recorded in PCM or analog (~100% if you don't count classical) Read the the post I posted. I simply said that the ones that ARE native without ANY convertion DO sound different from PCM.


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## chef8489

theaudiologist1 said:


> I already DID say, and was aware, that 90% of DSD recordings were recorded in PCM or analog (~100% if you don't count classical) Read the the post I posted. I simply said that the ones that ARE native without ANY convertion DO sound different from PCM.


If you have 2 recording setups at the same time one dsd and one pcm i doubt you could tell the difference. Let's say a classical recording that doesn't have to be edited.


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## KeithPhantom

theaudiologist1 said:


> Man this thread is still going after more than a decade. Wow!
> 
> Anyways, none of this matters because 1 bit DSD wipes the floor with PCM.


It stomps PCM with a higher noise floor (that needs to be noise-shaped) and requiring more data to represent the same information. Also, it cannot be processed in its native format (requiring DXD and this is PCM) and has little support compared to PCM. At least interpolating multi-level Delta Sigma took all of the disadvantages of DSD-like DACs (DSD being a storage medium for single-level DS data) to make the state-of-the-art in DACs nowadays.


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## KeithPhantom

theaudiologist1 said:


> Personally, DSD sounds more flat and natural (albeit a bit more quiet) than PCM, which has a lot more "oomph" to the sound. This is considering both my Native DSD and PCM files, and also listening to PCM without convertion and with DSD upsampling.


DSD does not have a sound. If possible, could you present us with a DSD and PCM copies from the same master that present significant differences? To be factual, evidence for your claims must be presented.


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## KeithPhantom

theaudiologist1 said:


> I already DID say, and was aware, that 90% of DSD recordings were recorded in PCM or analog (~100% if you don't count classical) Read the the post I posted. I simply said that the ones that ARE native without ANY convertion DO sound different from PCM.


If that is true, even pure tones by themselves or with their overtones (harmonics) will sound different (or will have changes in the waveform). I bet you that if we null the same recording from PCM and DSD, we will get silence (it won't be total because the noise floor is different, but the noise floor is inaudible anyway).


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## theaudiologist1 (Jul 3, 2020)

KeithPhantom said:


> It stomps PCM with a higher noise floor (that needs to be noise-shaped) and requiring more data to represent the same information. Also, it cannot be processed in its native format (requiring DXD and this is PCM) and has little support compared to PCM. At least interpolating multi-level Delta Sigma took all of the disadvantages of DSD-like DACs (DSD being a storage medium for single-level DS data) to make the state-of-the-art in DACs nowadays.


the super high noise floor is only a DSD64 problem if I'm not wrong. DSD128+ mostly solved the issue. PCM also has to be converted to DSD as well in many DAC's. The support is increasing in many modern products. Even budget DAC's support DSD and I'm not sure but I heard some phones like LG's have native DSD (not sure if true or not). Ironically I see DSD more than I see PCM rates above 24/192, and there's recordings recorded in DSD256 and DSD512 on NativeDSD more than in DXD.


Honestly, I somewhat take back my first post. I took the brutal blackpill and realized DSD hardly even exists, and although I think it sounds better than 16/44.1, it just barely ties with 24/96. And tbh I'm a "listen in the format it was released" guy: I listen to SACD's in DSD64 and my vinyl rips are 20-bit/384kHz (vinyl doesn't benefit from a high bitdepth since it only has a max dynamic range of ~60dB aka 10bits but most;y benefits from a higher sampling rate), and the files I download are in the format they were recorded in. The DSD sounding more "flat" and less "punchy" was my experience and probably just the way my DAC interprets the formats.


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## KeithPhantom

theaudiologist1 said:


> Honestly, I somewhat take back my first post. I took the brutal blackpill and realized DSD hardly even exists, and although I think it sounds better than 16/44.1,


I ripped and converted all of my SACDs to 44.1/16, and I cannot distinguish between the two, so I don't think there is an audible difference.


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## KeithPhantom

theaudiologist1 said:


> higher sampling rate


Why is this? Increasing the dynamic range by increasing the sampling rate is extremely inefficient. You only get 3 dB (I think) for every time you quadruple the sampling rate. Also, the interpolation in many DACs is only present to make the implementation of the low-pass filter easier. More points do not yield a better curve, you only need S/2 as stated by Nyquist.


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## theaudiologist1 (Jul 4, 2020)

KeithPhantom said:


> Why is this? Increasing the dynamic range by increasing the sampling rate is extremely inefficient. You only get 3 dB (I think) for every time you quadruple the sampling rate. Also, the interpolation in many DACs is only present to make the implementation of the low-pass filter easier. More points do not yield a better curve, you only need S/2 as stated by Nyquist.



A 20bit file is more than enough for the DR of vinyl. I simply have a higher sampling rate so that it's closer to a vinyl sound wave and a more natural sound (which technically has a sampling rate of infinity). That's actually one reason I liked DSD if it wasn't for the 1bit resolution.

Regarding interpolation, should I enable "invert polarity" in Audirvana? What is it for?


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## KeithPhantom (Jul 4, 2020)

theaudiologist1 said:


> Regarding interpolation, should I enable "invert polarity" in Audirvana? What is it for?


Polarity is only the relative positions of positive and negative samples, so inverting it would result in having positive values becoming negative and the opposite is true. It has nothing to do with interpolation (which is commonly known as oversampling). To keep the audio signal "as it is", leave that unchecked, that setting is only used to test the polarity of audio (for what I know).


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## KeithPhantom

theaudiologist1 said:


> which technically has a sampling rate of infinity


Also, vinyl is not digital, so it does not have a sampling rate. And no, you don't need infinite samples to perfectly reproduce the band-limited signal contained in it. Mathematically speaking (and vinyl), instruments can produce infinite harmonics, but many of the higher ones will be drowned down to the noise floor and masked by it, this will bandlimit the signal to a finite sum of harmonics thus a perfect representation is possible by the Nyquist theorem.


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## gregorio

theaudiologist1 said:


> [1] I basically said the same thing: that most DSD is recorded in PCM.
> [1a] That, I agree, has no "direct" advantage (PCM upsampled to DSD on the go with some software might push the noise more outside the hearing range so that's a plus) over PCM.
> [2] But native DSD and even analog-straight-to-DSD without any convertion does
> [2a] sound different and
> ...



Unfortunately, you are compounding your error!

1. I'm not sure that it is. As far as I'm aware, most DSD is recorded in DSD and then converted to PCM for mixing/processing. Do you have any reliable evidence to the contrary?
1a. No, resampling PCM to DSD does not "_push the noise more outside the hearing range_". Using an extremely high sampling rate spreads the dither noise of THAT sampling process over a wider frequency band but does NOT affect the dither/noise of any previous sampling process.

2. Sorry but I can't make sense of your statement, neither the individual parts of it, nor as a whole. What do you mean by "native DSD *and even* analog-straight-to-DSD"? There is *no* "native DSD", there is ONLY analogue to DSD, with the rare exception of purely digital synthesised sound, which is PCM!
2a. Again, this is the Sound Science subforum! One cannot just keep repeating the same disputed assertion without reliable evidence, so please provide some!
2b. Again, as my previous post illustrates, dsd is NOT "_closer to analogue than PCM_", the opposite, it is further from the analogue input signal! And, it's not due to the sampling rate, it's due to the bit depth.

3. Sorry but that makes no logical sense. If "_resolution = bit depth_" and "_DSD has no resolution_" then dsd MUST have a bit depth of zero bits ... which is obviously false! SACD has a bit depth of 1 bit and as "_resolution = bit depth_" then SACD has a resolution many times lower than CD (16 bits) but:
3a. Makes up for it with much more aggressively noise-shaped dither, NOT "_a much higher sample rate_". A much higher sample rate is required because it provides a broad band of ultrasonic frequencies in which all that additional dither noise can be (inaudibly) located. However, SACD does not "_make up for it_" entirely: As Lipshitz/Vanderkooy proved, not enough dither can be applied to 1 bit DSD without an overload state existing, so insufficient dither has to be applied, resulting in MORE distortion than can be achieved with 16bit! (Although this higher distortion is inaudible with music recordings).



theaudiologist1 said:


> [4] Personally, DSD sounds more flat and natural (albeit a bit more quiet) than PCM, which has a lot more "oomph" to the sound.
> [5] TLDR on mastering, mixing and editing:
> [5a] Native DSD = great
> [5b] Analog-to-DSD without PCM convertion = good
> [5c] PCM-to-DSD = useless and even makes it worse



4. This isn't the "Personal Impressions" forum, it's the Sound Science forum and therefore you must provide reliable evidence! And, as you are actually contradicting the established science (that DSD is NOT flatter, more natural or has less "oomph" than PCM), then not only are you going to need reliable evidence but extraordinary reliable evidence! From your description, you appear to be comparing different masters, *not* DSD vs PCM!

5. This is incorrect, it should read:

5a. Native DSD = Doesn't exist! There can be no mastering or mixing in 1 bit DSD and only "butt edits" are possible. Using 8 bit DSD to master, mix and edit is possible (though only very recently) but should theoretically/technically be inferior to converting to PCM (although most likely inaudibly inferior).

5b. Analog-to-DSD without PCM convertion = worst option!  I presume you are referring either to the so called "direct to disk" recordings, which of course doesn't allow for mastering or editing and aren't "direct to disk" because it first has to go through an analogue mixing desk, which of course adds significant analogue noise/distortion or, a traditional analogue recording converted to DSD after mastering, mixing and editing, in which case you've also got the same additional analogue mixing desk noise and distortion PLUS, even more additional noise/distortion from several generations of tape!

5c. PCM-to-DSD = useless and even makes it worse. I agree! After mastering, mixing and editing in PCM (which is the best option), converting the resultant master to DSD damages audio fidelity. However, as this damage is inaudible, then it's fair to just call it "useless". BUT, it's only "useless" as far as audio fidelity is concerned, as far as marketing is concerned it's very valuable because some/many audiophiles will pay considerably more for these DSD recordings (and the equipment to reproduce them), in the false belief they're actually higher fidelity!

G


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## old tech (Jul 4, 2020)

theaudiologist1 said:


> A 20bit file is more than enough for the DR of vinyl. I simply have a higher sampling rate so that it's closer to a vinyl sound wave and a more natural sound (which technically has a sampling rate of infinity). That's actually one reason I liked DSD if it wasn't for the 1bit resolution.
> 
> Regarding interpolation, should I enable "invert polarity" in Audirvana? What is it for?


Actually you are totally wrong.  A 12 bit file is more than ample to capture every nuance of a vinyl record, including all the hiss and distortions.  You obviously know nothing about sample rates and what they do, let alone what a sound wave is.


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## miksu8 (Jul 4, 2020)

Doesn't DSD have softer attacks because of the representation?


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## castleofargh

The root of perceived differences with digital audio(CD and up), in increasing order of likelihood, IMO:
-digital format.
-full moon.
-gremlins.
-the specific DAC used(not counting level difference between DACs or audio paths in a DAC).
...
...
-cables.
...
...
...
-misuse of the equipment/settings(when it's even possible)
...
...
-volume level difference.
-different mastering.
-listener's bias.


----------



## gregorio

theaudiologist1 said:


> [1] I already DID say, and was aware, that 90% of DSD recordings were recorded in PCM or analog (~100% if you don't count classical) Read the the post I posted.
> [2] I simply said that the ones that ARE native without ANY convertion DO sound different from PCM.



1. I did "read the post you posted" and I'm requesting evidence to support your claim because as far as I'm aware, the majority of DSD recordings were actually recorded in DSD.

2. The science: The mathematical theory, the objective measurements AND the controlled listening tests, *ALL* conclusively demonstrate the opposite, there is no audible sound difference. So, "_simply saying_" they "_DO sound different_", cannot be viewed here in the Sound Science forum as anything other than delusional nonsense, unless you have enough reliable evidence to invalidate decades of science.



chef8489 said:


> If you have 2 recording setups at the same time one dsd and one pcm i doubt you could tell the difference. Let's say a classical recording that doesn't have to be edited.



Another clarification, sorry! 
If we used a recording setup comprising of just one or two mics and recorded them in PCM and DSD, without editing, mixing or mastering, no one could tell the difference in a listening test. This is a contrived test though, that as far as I'm aware NEVER exists in commercial classical recordings. Commercial classical recordings typically use a minimum of 3-4 mics and sometimes more 50, so they have to be mixed, plus they're virtually always mastered and almost always edited. So, virtually all 1 bit DSD recordings are converted, still no audible difference though. The only exceptions I know of are a few recordings that have been live mixed in the analogue domain and haven't be edited or mastered.



KeithPhantom said:


> I ripped and converted all of my SACDs to 44.1/16, and I cannot distinguish between the two, so I don't think there is an audible difference.



Many people have effectively done the same experiment and (with controlled testing) have not been able to distinguish between SACD and 44/16. The most comprehensive and well known is probably the Meyer & Moran study published in 2007 (http://drewdaniels.com/audible.pdf).



theaudiologist1 said:


> I simply have a higher sampling rate so that it's closer to a vinyl sound wave and a more natural sound (which technically has a sampling rate of infinity). That's actually one reason I liked DSD if it wasn't for the 1bit resolution.



Ah, the old SACD marketing lie rearing it's ugly head again!

1. A natural sound does NOT technically have a sample rate of infinity, it technically does not have a sample rate at all! 

2. The VERY LAST thing we would want is something that's closer to a sound wave produced from vinyl! The whole point of inventing digital audio recording in the first place was to AVOID the analogue noise and distortions of vinyl and all other analogue recording media. To use an analogy: Why would anyone buy say a modern high tech Ferrari and then modify it "_so that it's closer to a_" 1970's Ford Fiesta?

3. A higher sampling rate does NOT result in a more accurate (or natural) sound wave. Shannon proved over 70 years ago that provided you have at least 2 samples per "sound wave" (analogue audio wave), then the "audio wave" could be reproduced perfectly. Obviously, you can't get better than "perfectly", so whether you've got say 3 samples per sound wave or 3,000,000 doesn't make the slightest difference, 3,000,000 samples per sound wave cannot be more perfect than perfect, it can only be equally or less perfect.

Again, this is the Sound Science forum, not the Marketing BS forum!



miksu8 said:


> Doesn't DSD have softer attacks because of the representation?



Not unless it's broken!

G


----------



## theaudiologist1

KeithPhantom said:


> Polarity is only the relative positions of positive and negative samples, so inverting it would result in having positive values becoming negative and the opposite is true. It has nothing to do with interpolation (which is commonly known as oversampling). To keep the audio signal "as it is", leave that unchecked, that setting is only used to test the polarity of audio (for what I know).


 Ok thanks. I was confused because the words sounded similiar.


----------



## KeithPhantom

miksu8 said:


> Doesn't DSD have softer attacks because of the representation?


No, DSD does not have any characteristics that may affect the signal being recorded other than noise, which is pretty inaudible (even for DSD64 {-120 dBFS [20 bits]}). DSD does not sound different than PCM.


----------



## miksu8

KeithPhantom said:


> No, DSD does not have any characteristics that may affect the signal being recorded other than noise, which is pretty inaudible (even for DSD64 {-120 dBFS [20 bits]}). DSD does not sound different than PCM.



Okay so much higher sample rate allows exactly same signal as PCM, except for the possible noise issue.


----------



## KeithPhantom

miksu8 said:


> except for the possible noise issue.


Noise even for undithered Redbook is pretty much inaudible. Noise is a non-issue since the CD became mainstream. PCM and DSD are designed to do the same thing, and they both do it correctly without any major deviations (only being noise a really minor difference at a technical level, but I already explained that).


----------



## theaudiologist1 (Jul 5, 2020)

Thanks guys. I also want to know something else similar to this topic: Is FPGA better than chips and provide a lower latency and better sound, or it it, once again, placebo? Asking in regard of Hugo 2 vs Pro iDSD


----------



## Slaphead

theaudiologist1 said:


> Thanks guys. I also want to know something else similar to this topic: Is FPGA better than chips and provide a lower latency and better sound, or it it, once again, placebo? Asking in regard of Hugo 2 vs Pro iDSD



FPGAs are used by boutique DAC manufacturers because their production volumes are so low that ASICs are financially unviable. A well designed ASIC will outperform a programmed FPGA for any given job - if not in performance then in power consumption.

In the case of Chord I've noticed just enough latency to make production work a touch awkward at times, but it really is just a bit. If you're just listening then there's no problem at all.

FPGAs are not some secret HiFI sauce despite how hi end DAC producers market them - they're just a cheaper way for them to "make" their own DAC chips. That said that doesn't mean that FPGAs are bad - in the right hands they can sound damned good.


----------



## chef8489

gregorio said:


> .Another clarification, sorry!
> If we used a recording setup comprising of just one or two mics and recorded them in PCM and DSD, without editing, mixing or mastering, no one could tell the difference in a listening test. This is a contrived test though, that as far as I'm aware NEVER exists in commercial classical recordings. Commercial classical recordings typically use a minimum of 3-4 mics and sometimes more 50, so they have to be mixed, plus they're virtually always mastered and almost always edited. So, virtually all 1 bit DSD recordings are converted, still no audible difference though. The only exceptions I know of are a few recordings that have been live mixed in the analogue domain and haven't be edited or mastered.


Ik this doesn't happen. I was making a point that if we did the setup, it would be no difference in the sound.


----------



## pinnahertz

The results of Mark Waldrep's  *"HD Audio Challenge"*  are live.


----------



## gregorio

miksu8 said:


> Okay so much higher sample rate allows exactly same signal as PCM, except for the possible noise issue.



Sort of! DSD does have a great deal more noise than say 16/44 PCM. However, given the far larger audio bandwidth of DSD/SACD, all that noise can be spread over a far wider band. You can easily see this in a spectrogram, the dither noise typically reaches it's peak around 25kHz and continues throughout the rest of the spectrum. This amount of ultra-sonic noise can potentially cause an IMD (inter-modulation distortion) issue with downstream equipment, amps and/or headphones/speakers, which is why Sony implemented a 50kHz analogue filter in it's SACD players and REQUIRED an analogue filter at 30kHz - 50kHz in it's licence agreement for third party manufacturers. This fact obviously makes a nonsense of audiophile claim that SACD contained important/useful information up to 100kHz.

In practice, as far as audibility is concerned, SACD (1bit DSD) and CD are pretty much identical when it comes to noise. In the critical hearing band 1bit DSD provides roughly 120dB of dynamic range and 16/44 (that also has noise-shaped dither applied), in the critical hearing band provides roughly .... 120dB of dynamic range!!



theaudiologist1 said:


> [1] Is FPGA better than chips and provide a lower latency and better sound, or it it, once again, placebo?
> [2] Asking in regard of Hugo 2 vs Pro iDSD



1. How does lower latency provide better sound? Latency just refers to the delay between pressing play and the audio playing, or in recording situations it refers the time difference between the analogue input signal (say from a mic pre-amp) and the analogue output signal after passing through the AD/DA (digital conversion) loop. Typically, this delay/time difference is just a few milli-secs and to put that into perspective, the blink of an eye is typically around 300ms! As far as "better sound" is concerned, it's all relatively simple math, it doesn't matter what type of chip you use (ASIC or FPGA), they can all do the math required for audibly transparent/perfect reconstruction (conversion back to analogue).

2. Chord tout it's FPGA in terms of it's speed/ability to run the more demanding math of their bespoke reconstruction filter, that has more "taps" than conventional reconstruction filters. All very impressive as far as audiophile marketing is concerned but conventional reconstruction filters are already audibly perfect, so any audible difference cannot be anything other than placebo!



pinnahertz said:


> The results of Mark Waldrep's *"HD Audio Challenge"* are live.



I always had a lot of respect for Mark Waldrep, both in terms of the quality of his recordings and his generally "no nonsense", factual approach to marketing. There was one exception though, I did disagree with him about his statements of the audible benefits of high-res audio. As he states, " _I was convinced that high-resolution recording — real HD-Audio — would be perceptible._" and "_I was among the strongest advocates for this new and exciting “upgrade” to audio reproduction._" However, a few years ago he (reliably/controlled) tested HD vs CD and kudos to him, he actually publicly admitted he couldn't distinguish them.

Now he's gone a step further, tested hundreds of others and states: "_The hundreds of people that have participated in the second round of the HD-Audio Survey, have confirmed the results of the previous project. It is no longer possible to claim that “hi-res audio” is an important next step in the evolution of audio. HD-Audio is completely unnecessary for the reproduction of hi-fidelity._" !

G


----------



## pinnahertz

gregorio said:


> Now he's gone a step further...."_It is no longer possible to claim that “hi-res audio” is an important next step in the evolution of audio. HD-Audio is completely unnecessary for the reproduction of hi-fidelity._" !
> 
> G


What you want to bet...this will have very little impact on anything.  As much as I agree with his findings, this is like trying to stamp out a false religious belief.  People believe because they want to, facts be damned, and this one is reinforced by "higher and more is better", even if it isn't really.


----------



## jlawler

I used to love the sound of CDs and chuckle to myself at people that listened to vinyl.  One day I had the chance to sit and listen to a quality 24bit recording.  As I listened, my mind became open to the idea that perhaps there was something better than redbook.  There was no positive bias towards analog / 24bit for me (in fact quite the opposite) but my position shifted.  I have certainly heard recordings that are indistinguishable 16bit / 24bit.  I have also heard drastic differences in quality of the 16bit / 24bit sound.  Perhaps that is due to the care which goes into the preparation.  I do know that highly compressed music causes me listening fatigue. That may be a factor for others as well.

What I do not understand why some wish to deprive others of the joy of listening to something that sounds better to them.  If one listens and doesn't hear something better - great, move on.  You saved some money for the future.  No one is forcing anyone to listen to 24bit.  I listen to what I like because I enjoy it.  Others may find a "premium product" increases their satisfaction level.  Perhaps there are certain people who have better hearing capabilities than the majority of the population which are more sensitive to 16bit / 24bit differences.  I just don't understand why this has to be ideological polarizing. I don't think it would hurt me to let others be wrong in this case.  My advice is let the people enjoy the music how they choose.  

Peace


----------



## bigshot

jlawler said:


> I have also heard drastic differences in quality of the 16bit / 24bit sound.



I'm interested... Which specific recordings and what process did you use to compare them?


----------



## chef8489

jlawler said:


> I used to love the sound of CDs and chuckle to myself at people that listened to vinyl.  One day I had the chance to sit and listen to a quality 24bit recording.  As I listened, my mind became open to the idea that perhaps there was something better than redbook.  There was no positive bias towards analog / 24bit for me (in fact quite the opposite) but my position shifted.  I have certainly heard recordings that are indistinguishable 16bit / 24bit.  I have also heard drastic differences in quality of the 16bit / 24bit sound.  Perhaps that is due to the care which goes into the preparation.  I do know that highly compressed music causes me listening fatigue. That may be a factor for others as well.
> 
> What I do not understand why some wish to deprive others of the joy of listening to something that sounds better to them.  If one listens and doesn't hear something better - great, move on.  You saved some money for the future.  No one is forcing anyone to listen to 24bit.  I listen to what I like because I enjoy it.  Others may find a "premium product" increases their satisfaction level.  Perhaps there are certain people who have better hearing capabilities than the majority of the population which are more sensitive to 16bit / 24bit differences.  I just don't understand why this has to be ideological polarizing. I don't think it would hurt me to let others be wrong in this case.  My advice is let the people enjoy the music how they choose.
> 
> Peace


Well thats fine on the other head-fi sub forums, but this is sound science. Sound science is just that the science of the audio.


----------



## KeithPhantom

jlawler said:


> My advice is let the people enjoy the music how they choose


The point of knowing and discussing science in an audio forum is to get a better grasp of the inner workings of our equipment. The point is not only to enjoy, is also to learn and improve.


----------



## bigshot

If I find I like the taste of rat poison in my morning coffee, I hope someone deprives me of the joy by giving me a clue!


----------



## jlawler

bigshot said:


> I'm interested... Which specific recordings and what process did you use to compare them?


There are many examples listed in this thread 


chef8489 said:


> Well that's fine on the other head-fi sub forums, but this is sound science. Sound science is just that the science of the audio.


Science gave us the flat earth and many fallacies in history.  It is not perfect and only explains a very small fraction of the knowledge of the universe.  Of course it is important, but it is also limited.  It's also important to understand why there are outliers rather than dismissing them.


----------



## jlawler (Jul 7, 2020)

bigshot said:


> If I find I like the taste of rat poison in my morning coffee, I hope someone deprives me of the joy by giving me a clue!


true, but 24bit will not kill anyone (at least not volume matched)


----------



## KeithPhantom (Jul 7, 2020)

jlawler said:


> There are many examples listed in this thread
> 
> Science gave us the flat earth and many fallacies in history.  It is not perfect and only explains a very small fraction of the knowledge of the universe.  Of course it is important, but it is also limited.  It's also important to understand why there are outliers rather than dismissing them.


Error, science changes with evidence and proposing that science is always wrong because past mistakes or misunderstandings is fallacious. Also, *you* have to provide evidence that the status quo is wrong.

EDIT: I forgot that outliers are +-1.5 times the IQR (actually, add the 1st and subtract the 3ed quartile)  and that previous assumptions are based in a lack of experiments that were formalized by science when means were obtained.

Another definition of outlier is 3 times the standard deviation in a normal and t distribution).


----------



## bigshot (Jul 7, 2020)

jlawler said:


> There are many examples listed in this thread



Do you mean like this test a few posts before you? It is the subject of the posts you were replying to!

https://www.realhd-audio.com/?p=6993

If you bothered to read it, you'd know that the well known engineer Mark Waldrep who writes a regular column in Real HD-Audio conducted a rigorous listening test and concluded:  "*Hi-Res Audio or HD-Audio provides no perceptible fidelity improvement over a standard-resolution CD or file."

DID YOU EVEN READ THIS THREAD?*

Honestly, I started out to be nice to you, but if you're going to spout nonsense and dismiss me with a wave of your hand, I'm not going to bother.


----------



## gregorio

jlawler said:


> [1]  As I listened, my mind became open to the idea that perhaps there was something better than redbook. There was no positive bias towards analog / 24bit for me ...
> [2] I have certainly heard recordings that are indistinguishable 16bit / 24bit. I have also heard drastic differences in quality of the 16bit / 24bit sound.
> [2a] Perhaps that is due to the care which goes into the preparation.
> [3] I do know that highly compressed music causes me listening fatigue. That may be a factor for others as well.
> [4] What I do not understand why some wish to deprive others of the joy of listening to something that sounds better to them.



1. I'm sorry but that doesn't make sense, because "_your mind becoming open to the idea that there was something [EG. 24bit] better than redbook_" IS pretty much the definition of a positive bias!

2. As explained in the original post, it is NOT possible that you've heard even tiny "_differences in the quality of 16bit/24bit sound_", let alone "_drastic_" differences, because there is NO difference in sound quality, the sound quality is identical! The ONLY difference between 16 and 24bit is the noise floor, which in both cases is inaudible at any reasonable listening level.
2a. No, "_the care which goes into the preparation_" must be the same because there is only one recording/mix. We do NOT make one recording at 16bit, another at 24bit, and then create two separate mixes and masters from them.  There is just one 24bit recording which is mixed, mastered and recorded to both 24bit and 16bit. However, there can be different edits of the mix and different masters (a "radio edit" was not uncommon for example) and at some later date there may also be a "re-mix". In these cases though, more "care"/work has to go into the preparation, not less, and there can be "drastic differences" but of course that's due to a different master, edit or remix, NOT the difference between 24bit and 16bit!

3. But converting from the master's bit depth (which is typically 64bit) to either 24bit or 16bit does not introduce any compression. It is possible to add more compression to either the 16bit or 24bit versions if desired/required, but then of course there's effectively two different masters.

4. As effectively mentioned by others, this is NOT the "Joy of listening" subforum, it's the Sound Science subforum, where we discuss the actual facts/science of sound/audio, REGARDLESS of how or whether that might affect any individual's "joy"!



jlawler said:


> [1] Science gave us the flat earth and many fallacies in history.
> [2] It is not perfect and only explains a very small fraction of the knowledge of the universe.
> [2a] Of course it is important, but it is also limited.
> [3] It's also important to understand why there are outliers rather than dismissing them.



1. You seem to have that backwards! Science gave us the spherical Earth well over 2,000 years ago, it was myth/superstition/propaganda that gave us the flat earth and modern science was developed specifically to separate myth, superstition, propaganda and fallacies from the actual facts. How is it possible NOT to know or understand this?

2. This is obviously NOT true, science explains ALL (100%) of our knowledge of the universe! However, ...
2a. Of course our (human) knowledge is limited, we obviously don't yet have all the potential knowledge, which is why we still have research scientists and theoreticians. However, science not having all the knowledge obviously doesn't mean that it doesn't have any. The reason we teach science in school, starting even at an early age, is to understand what science is: For example, what is a "hypothesis" and what is a "theory". Leading to an understanding of where science is "limited" and may not yet be perfect and where it is. For example, "1 + 1 = 2" is probably about the oldest science known, what subsequent or other knowledge demonstrates or even hints that it's "not perfect" and who in their right mind would dispute it? Due to a great deal of false marketing, this is where so many audiophiles fail. Many appear to believe that the science of digital audio is some new cutting edge hypothesis/theory, that is not fully understood and/or has a good probability of being incorrect/imperfect. This is simply untrue, it's a nearly century old theory that was proven over 70 years ago and pretty much nothing in human history has been as thoroughly researched or demonstrated in practice. If it were incorrect/imperfect, there would be no digital devices and no "digital age"!

3. No, it is not! This is the Sound Science forum and therefore we obviously adhere to the basic tenets of science, which is a logical, ordered process. The FIRST step of the process would be to ascertain if there really are any outliers, using scientific/controlled testing. OBVIOUSLY, only then, if there are actually any outliers, would it be important to understand why there are. However, given the obvious conditions of reasonable normalisation (IE. All commercial music and sound recordings) and reasonable listening volumes, there has NEVER been any outliers in the tens of thousands of controlled tests performed over the course of nearly three decades. Therefore, it is obviously not important to understand why some people claim to be an outlier and entirely correct that they be dismissed, unless they have a significant amount of reliable evidence to support their claim!

G


----------



## pinnahertz

bigshot said:


> Honestly, I started out to be nice to you, but if you're going to spout nonsense and dismiss me with a wave of your hand, I'm not going to bother.


Don't bother.  He just mentioned "flat earth" as a result of science when it was, in fact, a cultural belief that had nothing to do with science.  Just like his audio beliefs.  

Again, don't bother.  Just not worth it.  And my previous post has just been proven true.


----------



## castleofargh

jlawler said:


> What I do not understand why some wish to deprive others of the joy of listening to something that sounds better to them.


It's something you made up, so if someone can understand it, it's you.



jlawler said:


> Science gave us the flat earth and many fallacies in history. It is not perfect and only explains a very small fraction of the knowledge of the universe. Of course it is important, but it is also limited. It's also important to understand why there are outliers rather than dismissing them.


Your example is about popular beliefs, not science.

There is indeed no need for things to turn into false dilemmas. So how about you stop trying to create some?


----------



## bigshot

Joined a little over a week ago. Nickname of a championship wrestler, 5 posts, all in Sound Science in cables and 24 bit... I think what we are seeing here is a troll who got spanked and sent on his way in the past, so he created a nice new nick and started again fresh. This guy isn't very good at it though. He could only get to five posts before being roundly dismissed.


----------



## 71 dB

jlawler said:


> 1. I used to love the sound of CDs and chuckle to myself at people that listened to vinyl.  One day I had the chance to sit and listen to a quality 24bit recording.  As I listened, my mind became open to the idea that perhaps there was something better than redbook.
> 
> 2. There was no positive bias towards analog / 24bit for me (in fact quite the opposite) but my position shifted.  *I have certainly heard recordings that are indistinguishable 16bit / 24bit.*
> 
> ...


1. The obvious question here is: How does a downmixed 44.1/16 version sound compared to the 24 bit version? To avoid expectations bias you can ask your friend to play them so that you don't know which one is which. Do you hear a difference or not?

2. This should tell you when the 16 bit version is downmixed from the 24 bit they are indistinguishable.

3. It is a known fact sometimes there are difference in the master between 16 bit and 24 bit versions. In these cases the 24 bit version IS better, but not because it's higher resolution, but because it is from a better master. Listening to the 24 bit version for this reason is totally ok, but it's good to know _why_ it's better so that you don't reject 16 bit stuff needlessly. Sellers of hi-res audio benefit from the faulty beliefs of audiophools. They can charge more for hi-res versions or make people "upgrade" their favorite albums yet again. You said you used to love the sound of CDs. Ask yourself why this warm relationship has been compromized?

4. I am not telling you to not listen to 24 bit audio. It's not my business to tell you what to do. This is just a discussion about what the facts are and to be aware of them. If you tell others you listen to 24bit versions because they are from better masters then great! Thumps up! Way to go! Well done! If you tell us you listen to 24 bit versions, because it's _perceptually_ better format than 16 bit then we do have an issue with it, because science tells otherwise.


----------



## KeithPhantom

71 dB said:


> The obvious question here is: How does a downmixed 44.1/16 version sound compared to the 24 bit version? To avoid expectations bias you can ask your friend to play them so that you don't know which one is which. Do you hear a difference or not?


I just decimated all my 24 and 32 bit downloads to 44.1/16(I kept the original files in an external hard drive just in case) and did an ABX, I found no difference (by the way I'm only 21 and I can still hear 20 kHz from my left ear). .


----------



## bigshot (Jul 9, 2020)

I think most of us have done that. It isn't hard to do. I honestly don't understand why people keep talking nonsense when a simple test would put them on the right track.

By the way, here is a photo of Jerry Lawler for those who aren't familiar with his wrestling career...


----------



## 71 dB

Of course I know legendary Jerry "The King" Lawler and I am a Finn!



KeithPhantom said:


> I just decimated all my 24 and 32 bit downloads to 44.1/16(I kept the original files in an external hard drive just in case) and *did an ABX, I found no difference* (by the way I'm only 21 and I can still hear 20 kHz from my left ear). .



Exactly!


----------



## jlawler

71 dB said:


> 1. The obvious question here is: How does a downmixed 44.1/16 version sound compared to the 24 bit version? To avoid expectations bias you can ask your friend to play them so that you don't know which one is which. Do you hear a difference or not?
> 
> 2. This should tell you when the 16 bit version is downmixed from the 24 bit they are indistinguishable.
> 
> ...


I do get what you are saying here.  

There are a couple of other benefits of 24bit that I didn't delve into.  When you conduct processing on music it generally sounds less artificial & develops less artifacts when using 24bit sources.  Some recordings improve drastically to my ears when processed.  I also enjoy listening to some recordings in multichannel which requires processing on sources where true multichannel mixes are not available.  Also, when I compare 24/192 to 16/44, the overall presentation of the sound to my ears is much less harsh.  For example, cymbals, trumpets and saxophones are much more lifelike & pleasing to my ears.  I can abx a difference in these rates with foobar2000.


----------



## sonitus mirus

jlawler said:


> I do get what you are saying here.
> 
> There are a couple of other benefits of 24bit that I didn't delve into.  When you conduct processing on music it generally sounds less artificial & develops less artifacts when using 24bit sources.  Some recordings improve drastically to my ears when processed.  I also enjoy listening to some recordings in multichannel which requires processing on sources where true multichannel mixes are not available.  Also, when I compare 24/192 to 16/44, the overall presentation of the sound to my ears is much less harsh.  For example, cymbals, trumpets and saxophones are much more lifelike & pleasing to my ears.  I can abx a difference in these rates with foobar2000.


Foobar2000 ABX tool has a validation component with an sha1 checksum for each file and a final digital signature that can be validated.  If you find a section of a song where you can easily identify a difference, cut this sample down to 30 seconds so it can be shared with the group without risk of copyright infringement.  Test again with these 30-second samples and then send the full log of the test results with the 2 files used.

If you are statistically able to identify a difference between the files, and you provide the file samples, we can analyze the files and explain what you did wrong when converting them for testing, identify that these are completely different masters, or perhaps find that the files provided were not the actual files used in your ABX test.


----------



## KeithPhantom

jlawler said:


> Also, when I compare 24/192 to 16/44, the overall presenta


Provide the files and the log of your ABX since I did the same and didn't find any difference p=0.194.


----------



## miksu8 (Jul 11, 2020)

gregorio said:


> Sort of! DSD does have a great deal more noise than say 16/44 PCM. However, given the far larger audio bandwidth of DSD/SACD, all that noise can be spread over a far wider band. You can easily see this in a spectrogram, the dither noise typically reaches it's peak around 25kHz and continues throughout the rest of the spectrum. This amount of ultra-sonic noise can potentially cause an IMD (inter-modulation distortion) issue with downstream equipment, amps and/or headphones/speakers, which is why Sony implemented a 50kHz analogue filter in it's SACD players and REQUIRED an analogue filter at 30kHz - 50kHz in it's licence agreement for third party manufacturers. This fact obviously makes a nonsense of audiophile claim that SACD contained important/useful information up to 100kHz.
> 
> In practice, as far as audibility is concerned, SACD (1bit DSD) and CD are pretty much identical when it comes to noise. In the critical hearing band 1bit DSD provides roughly 120dB of dynamic range and 16/44 (that also has noise-shaped dither applied), in the critical hearing band provides roughly .... 120dB of dynamic range!!



Btw Rob Watts, who I think also knows something about this, says there is also an issue with transients with DSD.


----------



## 71 dB (Jul 10, 2020)

jlawler said:


> I do get what you are saying here.
> 
> There are a couple of other benefits of 24bit that I didn't delve into.  When you conduct processing on music it generally sounds less artificial & develops less artifacts when using 24bit sources.  Some recordings improve drastically to my ears when processed.  I also enjoy listening to some recordings in multichannel which requires processing on sources where true multichannel mixes are not available.  Also, when I compare 24/192 to 16/44, the overall presentation of the sound to my ears is much less harsh.  For example, cymbals, trumpets and saxophones are much more lifelike & pleasing to my ears.  I can abx a difference in these rates with foobar2000.



You are correct about 24 bit being "safer" bet compared to 16 bit if additional processing is done, althou as long as the processing isn't drastic 16 bit should be ok. When I say 44.1/16 is enough for consumer audio it assumes no additional processing or only mild processing (say cutting bass by 3 dB) is done.

Your DAC probably has different sound with 192 kHz and 44.1 kHz. You could try upsampling 44.1 kHz to 192 kHz and test if the sound changes. DACs are imperfect devices and always introduce something to the sound because of jitter, reconstruction filters and non-linearities. How audible it is is another issue.


----------



## bigshot

jlawler said:


> I can abx a difference in these rates with foobar2000.



I don't believe you. Why did you wait until now to say you had done controlled tests? You're just making stuff up.


----------



## KeithPhantom

jlawler said:


> I can abx a difference in these rates with foobar2000


Well, prove it by providing a log and your methodology. I don't think you can hear either quantization noise or the noise floor.


----------



## bigshot

He's googling other threads to see what he should say.


----------



## miksu8 (Jul 11, 2020)

Maybe a dumb question, but is conversion from 24 bit to 16 bit lossless when sample rate is the same? Scientific explanation would be nice.


----------



## bigshot

No, 24 bit has more zeros and ones, but it doesn’t matter because it’s been clearly proven that human ears can’t detect a difference. See the article in my dig called 16 bit is all you need.


----------



## miksu8 (Jul 11, 2020)

bigshot said:


> No, 24 bit has more zeros and ones, but it doesn’t matter because it’s been clearly proven that human ears can’t detect a difference. See the article in my dig called 16 bit is all you need.



If it's the last link in your sig it says not found.


----------



## castleofargh

71 dB said:


> You are correct about 24 bit being "safer" bet compared to 16 bit if additional processing is done, althou as long as the processing isn't drastic 16 bit should be ok. When I say 44.1/16 is enough for consumer audio it assumes no additional processing or only mild processing (say cutting bass by 3 dB) is done.
> 
> Your DAC probably has different sound with 192 kHz and 44.1 kHz. You could try upsampling 44.1 kHz to 192 kHz and test if the sound changes. DACs are imperfect devices and always introduce something to the sound because of jitter, reconstruction filters and non-linearities. How audible it is is another issue.


Most of those processing concerns can be addressed without starting with 24bit files(or high sample rate for that matter), and usually are without us having to lift a finger. The DSPs that still run at 16/44 do it as a deliberate choice(or were made by a guy in 5 minutes and probably have better alternatives). Even something as simple as volume control is handled at 32 or 64bit nowadays, and anything that can benefit from resampling will just do it. Then stuff are dithered (or not) and resampled when sent out to the DAC at whatever bit depth and sample rate we have set for it. Which usually should be 24 or even 32 if the DAC and driver happens to handle it.
All this will still happen with a 16/44 file. And a 24/96 probably still isn't the bit depth and sample rate used by heavy VSTs or whatever DSP or even the DAC itself that will probably end up with a handfull of bits and crazy high sample rate.

Obviously working with higher resolution makes sense for various reasons in production, but indeed for consumer audio we can have most of the benefits of higher everything while using 16/44 tracks.


----------



## castleofargh

miksu8 said:


> Maybe a dumb question, but is conversion from 24 bit to 16 bit lossless when sample rate is the same? Scientific explanation would be nice.


The encoding format is lossless, the conversion to a lower resolution isn't. So you can put 16/44 data into Wav, Flac or any lossless format at the same resolution and you will be able to retrieve all that data as it was. You can decode it, re-encode it, and still you should get the same data. That's what the lossless denomination of the format describes. 
But if you go from 24bit data to 16bit data, you've literally changed the resolution and removed some bits at the end of the code for each sample. That's not a lossless operation.


----------



## gregorio

jlawler said:


> There are a couple of other benefits of 24bit that I didn't delve into.
> [1] When you conduct processing on music it generally sounds less artificial & develops less artifacts when using 24bit sources.
> [2] Some recordings improve drastically to my ears when processed.
> [3] Also, when I compare 24/192 to 16/44, the overall presentation of the sound to my ears is much less harsh. For example, cymbals, trumpets and saxophones are much more lifelike & pleasing to my ears.
> [3a] I can abx a difference in these rates with foobar2000.



1. Processing virtually always occurs at 32bit or 64bit, regardless of whether the source files are 16bit or 24bit. So, the artefacts are at exactly the same level regardless of a 16bit or 24bit source!

2. Sure, but of course that's the freq response of your HPs (or speakers + room), the freq response of your ears and your personal preferences, which has nothing to do with the source file's bit depth!

3. Firstly, there is no mechanism/process which could cause that effect and Secondly, it is easily verified (with a null test for example) that the "_sound to your ears_" is identical with 16bit or 24bit. The only difference is ultrasonic content (above about 22kHz) and very low level dither noise, BOTH of which are inaudible (to your or anyone else's ears) at reasonable listening levels! The only rational explanation is therefore that you are "hearing" some difference that occurs AFTER the "_sound to your ears_", EG. Some bias in your perception.
3a. There are two conditions under which it can be possible to ABX a difference between 24/192 and 16/44:
A. Using a 24/192 recording with significant frequency content at 22kHz or higher that is causing IMD in your amp or speakers (within the audible spectrum) and comparing it with a 16/44 conversion, that obviously can't have any content above 22kHz and is therefore not causing any audible IMD. And ...
B. Using a very low level, non noise-shaped (TPDF) dithered test signal (or finding a very low level segment in a music recording) and amplifying it massively, so that the dither is audible.
In case "A", although one could discern an audible difference, the result is actually backwards. The 24/192 version is reproduced with unwanted distortion, effectively at a LOWER fidelity than the 16/44 conversion! In case "B" we have created an artificial scenario that CANNOT exist when we're listening to music recordings, in practice it would massively overload a system, blow your drivers and/or ear drums, as explained in the OP. Furthermore, the difference with this artificial scenario is the audibility of TPDF dither noise, NOT a "_much less harsh presentation_".
If you really can "_ABX a difference in these rates_" at reasonable levels, then you MUST provide the evidence, because you are contradicting a significant body of established scientific evidence!



miksu8 said:


> Btw Rob Watts, who I think also knows something about this, says there is also an issue with transients with DSD.



Rob Watts does apparently "_know something about this_". Unfortunately though he misrepresents/lies about it!! From your linked post:



Rob Watts said:


> [1] .... This is the reason why -6dB is the max for DSD; but the innate noise floor modulation problem occurs well below -6dB.
> [2] .. Dave's noise shaper will perfectly resolve a -301 dB signal - and this is essential for the perception of sound stage depth.
> [3]  My pulse array (5 bits, 2048FS) does not show this behavior on simulation - a -60dB step change has no consistent delay compared to 0dB step change.
> [3a] This is due to the very high speed of operation,
> ...



1. Obviously that's nonsense. If -6dB is the max for DSD, then how can a noise problem (or anything else) occur beyond the max? If -6dB really were the max, every SACD would be unlistenable because the very highest peaks could only be 6dB above the noise floor and the majority of the recording would be below the noise floor! The actual noise floor (in the audible spectrum) of SACD is NOT -6dB, it's about -120dB, which is easily verifiable.

2. If the last one wasn't bad enough, this assertion is way beyond nonsense, it's utterly ridiculous! Consider that the theoretical limit of 24bit is -144dB but it's only in theory, in practice we can't achieve anywhere near that level because if 0dB is say the sound level of a truck driving past from about 10ft away, then -144dB would roughly be the sound level produced by two hydrogen atoms colliding! Of course, the sound of two hydrogen atoms colliding is way, way below the ability of speakers/drivers to reproduce and of the human ear drum to detect. So, talking about a real world signal at -144dB is very silly indeed BUT here's Rob Watts talking about a "-301 dB signal", which is roughly 100 million times lower than -144dB! So what's 100 million times more than "very silly indeed"?? The best I can come up with is "utterly ridiculous beyond imagination" which apparently, in Rob Watts marketing BS language, translates to "_essential for the perception of sound stage_"! It makes Monty Python seem entirely reasonable ... you've got to laugh!!!

3. And neither does an "off the shelf" $2 DAC chip with a standard linear phase filter!
3a. True but obviously you don't need more speed than a stock DAC chip.
3b. That's interesting, many/most pro audio ADC/DACs use 6 bit resolution at similarly high sample rates, I wonder why Rob Watts "cheaps out" with only 5 bits?
3c. Nonsense, of course you can dither a DSD system, if you couldn't all SACDs would be swamped with noise and un-listenable. It's true that you can't dither a 1bit DSD system adequately enough to linearise all quantisation distortion but you can reduce it to below audibility.

Sorry, but it's one of the oldest audiophile marketing tricks in the book: Take some issues that are inaudible, purely theoretical or miniscule to the point of laughable, FALSELY describe them as "_massive and unique problems_", explain how your new/super-duper DAC solves them and is therefore "massively" better than other DACs and worth it's MASSIVELY over-inflated price!!!

G


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## KeithPhantom

castleofargh said:


> But if you go from 24bit data to 16bit data, you've literally changed the resolution and removed some bits at the end of the code for each sample. That's not a lossless operation.


And increased the noise floor by adding more quantization errors and decimating the theoretical dynamic range of the music.


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## danadam (Jul 11, 2020)

miksu8 said:


> If it's the last link in your sig it says not found.


Here's the correct link: https://xiphmont.dreamwidth.org/57937.html

@bigshot You should update your sig 

EDIT: Ups, it just leads to the bigshot's link  
Wayback Machine then: https://web.archive.org/web/20200426202431/https://people.xiph.org/~xiphmont/demo/neil-young.html


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## bigshot

danadam said:


> @bigshot You should update your sig



Fixed! Thanks!


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## 71 dB

KeithPhantom said:


> And increased the noise floor by adding more quantization errors and decimating the theoretical dynamic range of the music.



If you simply_ truncate_ the 24 bit sample points into 16 bit samples then this happens. If you add 16 bit dither noise to the 24 bit file before truncation, you mitigate problems: quantization errors get randomized into noise which doesn't correlate with the signal and also you prevent increased distortion. In that sense the original signal still has it's original dynamic range, but is masked partly by the added dither noise. Reverb tails for example decay into the noise the same fashion analog sound works so that signal levels below the dither noise level can be heard (if the listening level was insane) whenever the dither doesn't completely mask it. A 16 bit audio file produced from 24 bit file can _theoretically_ have a "ananog-like noisy" dynamic range of 120 dB depending on what kind of dither is used. It's good to remember bits do not quarantee dynamic range. If I digitize old noisy C-cassettes into 24 bit files I certainly don't get huge dynamic range. A 24 bit file might have only 18 bits worth of dynamic range* for example so that turning it into "ananog-like noisy" 16 bit file which has a "noisy" 20 dB dynamic range doesn't mean much. In consumer audio about 80 dB of dynamic range is all we need, about 13 bits worth of dynamic range. In this sense even 16 bit is overkill by a few bits, but that's just a nice safety margin.

* Try and record acoustic instruments at 24 bit dynamic range and come back to tell us how well that went.


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## gregorio

castleofargh said:


> [1] But if you go from 24bit data to 16bit data, you've literally changed the resolution and removed some bits at the end of the code for each sample.
> [2] That's not a lossless operation.



1. Careful here. What do we mean by "resolution"? Effectively, we haven't changed the resolution, the resolution of the output signal is the same, what has changed is the amount of noise which accompanies the output signal but at 16bit, that noise is inaudible.

2. It IS an audibly lossless operation though.  We could even argue that in practice we have not "lost" any resolution because at least the last 10 LSBs of a 24bit recording are just noise, of which we loose 8 and add 1 (16bit) LSB of .... noise. So, we've _"literally changed the resolution_" purely in terms of exchanging inaudible noise for the same amount of inaudible noise! Is that really lossy? 

Therefore:



KeithPhantom said:


> [1] And increased the noise floor by adding more quantization errors and
> [2] decimating the theoretical dynamic range of the music.



1. Not really. There are no quantisation errors, what we have instead is dither noise, which *would* have "_increased the noise floor_" ONLY IF the noise floor of the 24 bit recording were below about -120dB. Do you know of any commercial 24bit recordings with a noise floor lower than -120dB? The lowest noise floor commercial recording I've heard of, has a noise floor about 300 times higher than that and the vast majority 1,000+ times higher. Bare in mind, it's standard practice to use noise-shaped dither when converting from >16bit to 16bit, which puts the digital noise floor of the 16bit version at about -120dB.

2. That depends on what you mean by "_theoretical dynamic range of music_". Your assertion could be true (in theory) for only one very specific form of music: Music comprised SOLELY of signals digitally synthesised (at >20bit), but even then it is only "_theoretical_" because AFAIK, there are no music recordings in practice that actually employ a dynamic range greater than 120dB. All other forms of music, even including almost all electronic music, is constrained by either analogue inputs, analogue modelling, acoustic recording conditions or all of the above, EACH of which have a noise floor above or way above -120dB. So even theoretically, the dynamic range of music is NOT "decimated", except in the one *potential* case just mentioned. Incidentally, almost all electronic music, even purely "generated in the box"  electronic music (without any recorded instruments or vocals), relies on at least some "digital synths" which aren't really synthesizers, they're reliant on sample playback and therefore constrained by the noise floor of the samples. 

G


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## KeithPhantom

71 dB said:


> If you simply_ truncate_ the 24 bit sample points into 16 bit samples then this happens. If you add 16 bit dither noise to the 24 bit file before truncation, you mitigate problems: quantization errors get randomized into noise which doesn't correlate with the signal and also you prevent increased distortion. In that sense the original signal still has it's original dynamic range, but is masked partly by the added dither noise. Reverb tails for example decay into the noise the same fashion analog sound works so that signal levels below the dither noise level can be heard (if the listening level was insane) whenever the dither doesn't completely mask it. A 16 bit audio file produced from 24 bit file can _theoretically_ have a "ananog-like noisy" dynamic range of 120 dB depending on what kind of dither is used. It's good to remember bits do not quarantee dynamic range. If I digitize old noisy C-cassettes into 24 bit files I certainly don't get huge dynamic range. A 24 bit file might have only 18 bits worth of dynamic range* for example so that turning it into "ananog-like noisy" 16 bit file which has a "noisy" 20 dB dynamic range doesn't mean much. In consumer audio about 80 dB of dynamic range is all we need, about 13 bits worth of dynamic range. In this sense even 16 bit is overkill by a few bits, but that's just a nice safety margin.
> 
> * Try and record acoustic instruments at 24 bit dynamic range and come back to tell us how well that went.


True, I already had knowledge of all of this, but I just tried to get the message across. In actual music, the last LSBs are actually filled with noise and not many productions take advantage of the full dynamic range of their medium. Also, when decimating even with dither, you cannot keep the same theoretical dynamic range of the higher-count bit format because of math, but that's inaudible anyways in the 24 vs 16 bit comparison.


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## bigshot

Most of the stuff we discuss around here is inaudible!


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## 71 dB

bigshot said:


> Most of the stuff we discuss around here is inaudible!


MAYBE WE SHOULD USE MORE CAPSLOCK!!


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## bigshot

COMING IN LOUD AND CLEAR NOW!


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## KeithPhantom

gregorio said:


> Not really. There are no quantisation errors, what we have instead is dither noise, which *would* have "_increased the noise floor_" ONLY IF the noise floor of the 24 bit recording were below about -120dB.


Thanks for this information, didn't know this little fact. Well, actually I could infer this, but I didn't. 




gregorio said:


> That depends on what you mean by "_theoretical dynamic range of music_". Your assertion could be true (in theory) for only one very specific form of music: Music comprised SOLELY of signals digitally synthesised (at >20bit), but even then it is only "_theoretical_" because AFAIK, there are no music recordings in practice that actually employ a dynamic range greater than 120dB.


Also true, but what I meant was more a perfect scenario, in real life almost all recordings don't get close to -120 dB, and this a limit even most (not state-of-the-art) electronics struggle to get consistently in terms of SFDR and other measurements. 16 bits with dither should be enough for transparency.


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## bigshot

You'd have to record a jackhammer in Carlsbad Caverns to get a noise floor of -120dB. Most recording studios have a room tone from air conditioning and traffic noise in the mid 20s I would guess.


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## gregorio

KeithPhantom said:


> [1] Thanks for this information, didn't know this little fact. Well, actually I could infer this, but I didn't.
> [2] Also true, but what I meant was more a perfect scenario, in real life almost all recordings don't get close to -120 dB, and this a limit even most (not state-of-the-art) electronics struggle to get consistently in terms of SFDR and other measurements. 16 bits with dither should be enough for transparency.



1. It's covered to an extent in the OP. Dither is and always has been a requirement of digital audio recording. In the very earliest days (1960's and much of the 1970's), the technology didn't exist to perform dither in the digital domain, so analogue white noise was injected. Digital dithering was far more controllable/efficient and eliminates (linearises) ALL quantisation error. At the start of the 1990's, when bit depths greater than 16 became available in the pro audio world, noise-shaped dither was invented specifically for the situation of a higher than 16bit recording (or mix) that had to be converted to 16bit for consumer distribution.

2. Again though, it depends on what you mean by "_perfect scenario_". Do you mean a scenario that is perfect within the "laws of physics", EG. Is never achieved in the real world but potentially/theoretically could be. Or, do you mean a hypothetical scenario which ignores the "laws of physics",  EG. Could never exist in the real world?
For example, it might be theoretically possible to construct a recording venue with a 0dB SPL noise floor and, it is theoretically possible for say a large symphony orchestra to produce peak levels at 120dB SPL (if one records in or very near the orchestra). So theoretically wouldn't we have a dynamic range of 120dB (a noise floor at -120dB)? No, we wouldn't! Hypothetically we might but not in theory, because a large symphony orchestra requires 90+ musicians, all of whom are breathing, moving and therefore raising the noise floor considerably. And even in theory, a large symphony orchestra with no musicians or 90+ dead musicians is going to struggle to produce a 120dB peak level! 

We've also got a problem with the theoretical max dynamic range of mics and in combination with a mic pre-amp, which at the very least is going to add thermal noise. We have a similar problem at the other end of the chain, with reproduction. Even if we have a recording with a noise floor of -120dB and we reproduce it with a DAC that has a noise floor of -120dB, we now have a noise floor of 117dB. However, we obviously can't listen to the output of an DAC, we first have to amplify the analogue output of the DAC and then convert it to acoustic sound waves. So at the very least we've got the added thermal noise of the amp and speakers/HPs, not to mention the listening environment noise floor. In theory, the best we could achieve is probably somewhere around 100-110dB dynamic range but the best commercial studios actually manage is about 90dB (at huge cost!). Lastly, we have to consider that commercial music/sound recordings are, by definition, entertainment. They are designed to be "comfortable" with moderately priced equipment, they are NOT designed to require $1m worth of construction/reproduction equipment and go so far beyond "comfortable" that we're push the limits of human hearing safety. This limits dynamic range to around 40dB-50dB or in the case of a niche market, to about 60dB or so.

G


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## ScareDe2

I would say 16 bits 48khz works best for listening to music and 24 bits 41khz works best for watching movies.


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## KeithPhantom

ScareDe2 said:


> I would say 16 bits 48khz works best for listening to music and 24 bits 41khz works best for watching movies.


My question is why do you think that.


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## bigshot (Jul 14, 2020)

You've got that backwards. 16/44.1 is for CDs. 16/48 is for broadcast and DVDs. 24 is for mixing and mastering. They all are audibly transparent.


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## ScareDe2 (Jul 14, 2020)

On youtube I went from 16 bits 44 Khz to 48 Khz (in Window sound) and I preferred the latter. Then I found this video :



As for the 24 bits 44 Khz it is my window setting of choice for live recorded music that was not transfered to CD, DVD Bluray !


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## bigshot

YouTube uses compressed audio. This is the lowest form of trolling.


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## KeithPhantom

ScareDe2 said:


> On youtube I went from 16 bits 41 Khz to 48 Khz (in Window sound) and I preferred the latter.


Interpolation won't improve sound at all in this case. This is most likely expectation bias at its finest.


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## Davesrose

KeithPhantom said:


> Interpolation won't improve sound at all in this case. This is most likely expectation bias at its finest.



Time and time again, this poster has shown bias for "analog" distortion and other woo bias.  My take: it's futile to engage with him.



ScareDe2 said:


> The analog conversion works in synergy with the rest of the chain and helps simulate natural motion pictures.





ScareDe2 said:


> Anyway, before they close more topics and start deleting I will leave it at that for now. You guys are lucky modos are hunting down my threads because I had more science in the making.
> GG ✌



I don't know if I'm missing something, but when I engaged about current video only being digital, I didn't get one scientific point.


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## bigshot

He's a troll and not a very good one. A good troll doesn't make everyone give up on him. It's time for him to ditch this sock account and go create another one and come back and try again. I don't think he is able to last more than a few days under any name though.

I was telling Castle the other day, this group is incredibly useful. There are a lot of people here that know their stuff. If someone wants help with something, there's always knowledgable suggestions to be had. For people to come here and piss all over us like that is disrespectful. I don't feel like I'm under any obligation to extend respect to people who don't do the same for me. If you don't like what we say, just go away. There are a bunch of forums on Head-Fi where nonsense is welcomed. If you come here, you should be here because you want to hear what we have to say. Not to argue absurdly ill-informed positions with us. Just stay away from Sound Science.


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## ScareDe2

My current theory is that you want to receive at the same bitrate than the import work. For video, it seems that most video editors will import in 44Khz (not 41khz my mistake) hence why window sound seems better at 24 bits 44Khz for this purpose.

For music, I think uploaders import at 16 bits 48 Khz, but it is just a guess.

At the end of the day, use what works best for your ears. One thing is sure though, this industry has complicated things and should be blamed for that.


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## bigshot

You are talking nonsense. And you don't listen when someone nicely offers you the facts. You should take your theories elsewhere where people might appreciate them.


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## ScareDe2

My posts likes ratio is better than yours. My nonsenses are more appreciated than your facts.


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## bigshot (Jul 14, 2020)

There should be a button to click for nonsense trolling. Ignored.


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## KeithPhantom

ScareDe2 said:


> My posts likes ratio is better than yours. My nonsenses are more appreciated than your facts.


Facts do not care about like ratios you get in a random forum. Facts are testable and are established under experiments and peer review.


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## KeithPhantom

KeithPhantom said:


> Facts do not care about like ratios you get in a random forum. Facts are testable and are established under experiments and peer review.


I would ask for @castleofargh to actually come and if possible communicate to our fellow friend that scientific discussion is what it is discussed in this sub-forum, and being willingly ignorant miscarries this purpose.


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## bigshot

The only thing he tests is all of our patience.


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## Davesrose

bigshot said:


> There should be a button to click for nonsense trolling. Ignored.



Well anytime anyone who's not aware and tries to engage him: we do have a quote that's from the horse's mouth:



ScareDe2 said:


> My posts likes ratio is better than yours. *My nonsenses* are more appreciated than your facts.


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## ScareDe2

It is just better to leave the audio in the format it is originally in. It was reported that Window resampling was measured and result was not good.


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## bigshot (Jul 15, 2020)

Well... How about them Dodgers?

I'm curious about the Mark Waldrop HD audio study... http://www.realhd-audio.com/?p=6993 Is there an article that details how it was conducted and the specific results, or does he plan to publish it in a peer reviewed journal first?


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## Davesrose

bigshot said:


> Well... How about them Dodgers?



You know I'm a member of a photography forum that has a "title fairy".  So instead of "500+ Head-Fier" ScareDe2's title could read "My nonsenses are more appreciated than your facts"


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## KeithPhantom

bigshot said:


> I'm curious about the Mark Waldrop HD audio study... http://www.realhd-audio.com/?p=6993 Is there an article that details how it was conducted and the specific results, or does he plan to publish it in a peer reviewed journal first?


I read the description and you had to complete a survey to participate in it, also you would agree to not use software that reveals the sampling rate and bit depth. So, he is actually giving excessive trust to a group of complete strangers (the study's population) to get some results. I think it isn't scientific at all since there are no control groups and the chain of control is compromised.


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## ScareDe2

If the original work is published in hi-res format, then you should use hi-res playback. Simple.


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## bigshot

Well, it came out the same anyway I guess. I know with my own lossy vs lossless test that people try to game the system by peeking at waveforms. I've inserted a monkey-wrench for them in my test procedure. None of them have been able to overcome it. One guy on another forum tried and I was forced to embarrass him. (After I'd gotten him to make a bunch of incriminating statements, of course.) He never returned to the forum.


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## old tech

KeithPhantom said:


> I would ask for @castleofargh to actually come and if possible communicate to our fellow friend that scientific discussion is what it is discussed in this sub-forum, and being willingly ignorant miscarries this purpose.


I've already raised the issue with the mods.  He obviously is a troll, I mean no-one could be that thick, surely?


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## bigshot

I spoke to Castle too. Either he is a troll, or as you say... and I'm not allowed to say it or I'll be the one in trouble.


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## gregorio

ScareDe2 said:


> I would say 16 bits 48khz works best for listening to music and 24 bits 41khz works best for watching movies.



Firstly, there is no 24bit 41kHz format, I presume you mean 24bit 48kHz and if so, on what platform are you watching movies that actually provides 24/48 sound?



ScareDe2 said:


> [1] On youtube I went from 16 bits 44 Khz to 48 Khz (in Window sound) and I preferred the latter.
> [2] Then I found this video :
> [2a] As for the 24 bits 44 Khz it is my window setting of choice for live recorded music that was not transfered to CD, DVD Bluray !



1. It's NOT possible to "_prefer the latter_" because they are exactly the same! Youtube uses AAC lossy compression, converting that into 16/44 or 16/48 gives identical results. If you had a completely full 2 litre water container and you transferred all that water into a 5 litre water container, how much water would you have? Now repeat the experiment but transfer all the water from the 2 litre container into a 6 litre container, how much water do you have now and how would it be possible to prefer the amount of water in the 6 litre container over the exact same amount of water in the 5 litre container?

2. Generally pretty accurate, although he'd get the same result if he used 16/44 instead of 16/48.
2a. Why, when you've just cited a video showing there's no audible difference using 16 bit?


ScareDe2 said:


> [1] It is just better to leave the audio in the format it is originally in.
> [2] It was reported that Window resampling was measured and result was not good.



1. How can it be "better" if it's impossible?

2. *IF* this assertion were true, then why are you resampling your audio in Windows?



ScareDe2 said:


> [1] My current theory is that you want to receive at the same bitrate than the import work.
> [2] For video, it seems that most video editors will import in 44Khz (not 41khz my mistake)
> [2a] hence why window sound seems better at 24 bits 44Khz for this purpose.
> [3] For music, I think uploaders import at 16 bits 48 Khz, but it is just a guess.
> ...



1. This isn't the "ScareDe2 Current Theory" subforum, it's the Sound Science subforum. And, why do you want to "_receive at the same bit rate than the import work_"?

2. No, your mistake again! The worldwide commercial standard for audio import by video editors is 24 bits 48kHz, although there are a very few cases where a higher sample rate is employed. And, the audio import format used in the video industry is almost never the final export/distribution format. For example, the audio import format for HDTV is 6 channels of 24/48 but HDTV is never broadcast in that format, typically it's broadcast as Dolby Digital but some countries use multi-channel MP3 or AAC format.
2a. Hence why 24/44 would never be better, it would be the same!

3. Again, this is the Sound Science subforum, not the "Just a guess" subforum. And unfortunately, it's a very bad guess because "music uploaders import" a wide variety of audio formats, from lossy formats such as MP3 to 24/192 or even higher on rare occasions but you've guessed just about the one format (16/48) that is pretty much NEVER used!

4. Agreed, and that's WHY we use double blind/ABX testing! So we can actually determine "_what works best for your ears_" as opposed to simple A/B sighted testing which does NOT determine what works best for your ears, it determines what works best for your perception biases!

5. Oh dear, the "one thing" you're sure of, is also wrong! Firstly, it's not especially complicated compared to many other industries, it's fairly easy for the average person to learn/understand in layman's terms, if they wish to. And Secondly, these "things" have been created by the industry for good reasons: To provide artistic options in the creation of music/sound products, to provide the best storage of audio information in terms of quality and ease of use and to provide the best fidelity for distribution to consumers. And of course, as consumer technology has changed/improved over the decades, so too has audio information storage and consumer distribution formats. What other option is there and how is all this not obvious? I agree with you though in terms of the distribution formats available to consumers, some of which exist purely for audiophile marketing purposes rather than for any audible benefit, which of course is what this thread is about in the first place!



ScareDe2 said:


> If the original work is published in hi-res format, then you should use hi-res playback. Simple.



Do you mean "Simple" in terms of a "simple" self-contradiction? You stated that 24/44 was your "setting of choice" AND posted a video proving there is no audible difference using/converting to 16bit. And, you can't playback anything more hi-res than 16bit audio anyway, did you not read the OP before posting?

G


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## castleofargh

ScareDe2 said:


> I would say 16 bits 48khz works best for listening to music and 24 bits 41khz works best for watching movies.





ScareDe2 said:


> It is just better to leave the audio in the format it is originally in. It was reported that Window resampling was measured and result was not good.


Ah yes, of course. Most movies have 44.1kHz audio tracks while most of the music albums are encoded at 48kHz. Makes sense. 🙃 Guys you were wrong, you don't need moderation. You need to call Fringe division. He's giving away more and more evidence that he's from the other side.



Seriously @ScareDe2, what are you doing? Even if it rarely looks like it, this section should be reliant on fact based knowledge. Not on "I feel like I'm right so I'm gonna make claims on a subject I know almost nothing about". The rules of this forum tell you to join this section when discussing biases, placebo and the best ways to limit their impact on our understanding of audio stuff. It doesn't tell to join as a case study. We have more examples of cognitive bias than we care to see already, thank you very much. 
For an objective approach(which this section is more or less dedicated to), the role of personal opinions from sighted impressions is usually to be targeted and eliminated from our testing methods and discussions about objective facts. All your empty statements so far have been deeply rooted in your personal feelings of some anecdotal experience done sighted. Once again, it's the wrong section to do that. It simply is.
This section, for so many legitimate reasons demonstrated many times, is very skeptical of sighted impressions and will not accept them as evidence of something objective.

You know how mad some people get in the rest of the forum when someone keeps bringing up blind testing to contest what the other guy is saying? You know how that person bringing blind tests ends up with his posts moved here or most likely deleted and him locked out of the thread if he doesn't stop? Well it's the same situation but reversed. Making empty claims(mostly false ones so far!) and posting sighted impressions disguised as objective facts, that disturbs and annoys the members of this section. As you can very well see from the previous posts.


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## ScareDe2

44.1Khz and 48Khz samplerate are used for both audio and video and it is up to you to find what works best depending on the web site and content you watch/listen.

The video is just a complement to the information discussed in this topic.

One of my other theory is that we actually hear above the human audible limit. It is just that the brain/ear stops processing the pitch, but the higher frequency can still be detected (perhaps even way higher than 20Khz), and so can the noise created by window resampling.


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## bigshot

You make stuff up and then declare it to be true. You’re a random falsehood generator. Go away.


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## KeithPhantom

ScareDe2 said:


> One of my other theory is that we actually hear above the human audible limit. It is just that the brain/ear stops processing the pitch, but the higher frequency can still be detected (perhaps even way higher than 20Khz), and so can the noise created by window resampling.


We don't even have the biology required for that, also you would have to show evidence and not just create conjectures about what you think, show facts.


----------



## taffy2207

His 'theories' would give an Aspirin a headache.


----------



## jlawler

I must admit that given some of the reception my post got I do doubt you will accept my evidence, but I went ahead and did it anyways.  I created the 16/44 from the 24bit version using sox -V3 -v 0.99 -G -b 16 rate -v -L 44100.

https://gofile.io/d/7BUuGQ

foo_abx 2.0.6d report
foobar2000 v1.5.1
2020-07-16 17:49:45

File A: 09 Don't Bring Me Down.flac
SHA1: a19b2743680e319eaee3c8bcc08867e475848723
Gain adjustment: -3.06 dB
File B: 1644 Don't Bring Me Down.flac
SHA1: 26ce5826ba2caa06c68042789bb8ba1ac5a7a179
Gain adjustment: -2.97 dB

Output:
WASAPI (event) : DENON-AVR (2- NVIDIA High Definition Audio), 24-bit
Crossfading: NO

17:49:45 : Test started.
17:50:12 : 01/01
17:50:41 : 02/02
17:50:54 : 03/03
17:51:08 : 04/04
17:51:23 : 05/05
17:51:44 : 06/06
17:52:00 : 07/07
17:52:52 : 08/08
17:52:52 : Test finished.

 ---------- 
Total: 8/8
p-value: 0.0039 (0.39%)

 -- signature -- 
9b3537e98ce7e7ad2dbdbbf2008e3829bab2f373


----------



## jlawler

As I said, not every source is discernibly different (16 vs 24), so I do not doubt that you could possibly find one in which I cannot differentiate, but I am sure I could redo the test with a proper 16/44 conversion from the source I sent & I could provide more sources where the differences are evident if you believe there is something wrong with that source.


----------



## bigshot (Jul 16, 2020)

The first problem is your files. You compared 16 to 24. That introduces a slight bump on many computers when it switches to 24. If one file is 24 and the other is 24 to 16 and back up to 24 again, you won't have that problem. You can send your 24 bit source and we will send you back test files made from it.

The second problem is that you chose an analogue recording to test with. Analogue tape isn't capable of noise floors lower than 16 bit. So there's no way you could tell the difference anyway. You might want to find a native 24 file to work with rather than a legacy title.

I'm happy to help you do a good test if you are sincere. But it's a waste of everyone's time if you just want to try to fool us.


----------



## ScareDe2

KeithPhantom said:


> We don't even have the biology required for that, also you would have to show evidence and not just create conjectures about what you think, show facts.



With bone conduction, a deaf person can detect the 25 Khz frequency. Normal hearing persons can detect ultrasonic frequencies around 50 Khz and I have read young healthy subjects can hear as far as the 150 Khz frequency. I am surprised this have not been discussed, or it has and I did not see it anywhere on headfi. There is a lot to read about this topic and I just give this random link here:

http://www.tinnitusjournal.com/arti...dy-implications-for-highfrequency-therapy.pdf

Bone conduction transmission occurs constantly as sound waves vibrate bone - Wikipedia


----------



## KeithPhantom

ScareDe2 said:


> With bone conduction,


Didn't know, thanks for this.


ScareDe2 said:


> Normal hearing persons can detect ultrasonic frequencies around 50 Khz


This is what I don't think is possible, I am only 21 and can hear 20 kHz from my left ear and this is pretty hard. 150 kHz is what I don't think is possible, you would have to show evidence fo this. Also remember, I am saying "I do not think", I am not saying it could be otherwise, just show the evidence.


----------



## Davesrose (Jul 16, 2020)

ScareDe2 said:


> With bone conduction, a deaf person can detect the 25 Khz frequency.



No, a person with profound deafness can't detect anything (inner ear and/or nerves completely compromised).  Your linked study is hardly a definitive sample (5 normal hearing people and one "deaf" person...and only focusing on 25khz and 62.5 samples).



KeithPhantom said:


> This is what I don't think is possible, I am only 21 and can hear 20 kHz from my left ear and this is pretty hard. 150 kHz is what I don't think is possible, you would have to show evidence fo this. Also remember, I am saying "I do not think", I am not saying it could be otherwise, just show the evidence.



The only main studies I'm aware of that have shown ultra-high frequency hearing are divers (where the medium is no longer air). https://www.navy.mil/submit/display.asp?story_id=60632


----------



## ScareDe2

I was not there to witness God writing the tablets of stones but you have enough evidences to find the Truth by yourself now, like a full grown up lady.

That being said, we are surrounded by noise interferences and it can be a good idea to filter that noise. I like capacitors, ferrite choke, same samplerate as the published work or at least what works best for my listening situation, and a bit of luck.


----------



## KeithPhantom

Davesrose said:


> No, a person with profound deafness can't detect anything (inner ear and/or nerves completely compromised).  Your linked study is hardly a definitive sample (5 normal hearing people and one "deaf" person...and only focusing on 25khz and 62.5 samples).
> 
> 
> 
> The only main studies I'm aware of that have shown ultra-high frequency hearing are divers (where the medium is no longer air). https://www.navy.mil/submit/display.asp?story_id=60632


Also thanks for the information.


----------



## castleofargh

ScareDe2 said:


> With bone conduction, a deaf person can detect the 25 Khz frequency. Normal hearing persons can detect ultrasonic frequencies around 50 Khz and I have read young healthy subjects can hear as far as the 150 Khz frequency. I am surprised this have not been discussed, or it has and I did not see it anywhere on headfi. There is a lot to read about this topic and I just give this random link here:
> 
> http://www.tinnitusjournal.com/arti...dy-implications-for-highfrequency-therapy.pdf
> 
> Bone conduction transmission occurs constantly as sound waves vibrate bone - Wikipedia


https://www.head-fi.org/search/1549633/?q=bone+conduction&t=post&c[nodes][0]=133&o=date
Not sure if that link works for others? just in case, it involves searching for "bone conduction" in the Sound Science section

But beside reading the paper, there isn't much to talk about. A headphone probably won't have that effect because we use comfy pads, not jackhammers. And because as I understand it, the more a headphone itself shakes, the less the diaphragm is moving as it should=> more distortions. so we might need bad headphones for something like that to work well.
 Most headphones won't extend or remain loud enough at those frequencies. 
Typical music content probably doesn't have the needed ultrasonic energy even at unreasonably loud listening levels. If/when it does, it might just end up audibly masked by the rest of the music(perhaps with the exception of poorly filtered DSD, but then that's all noise-shaped crap and nobody wishes to perceive that). 

When the right conditions are met for ultrasounds to be heard from bone conduction( AKA not using typical headphones), listeners seem to be hearing something similar to a 12 to 14kHz tone that doesn't correlate with the actual frequency of the ultrasound. So it might be something closer to your windows resonating from your woofer, than to music content. and of course nothing in the research is suggesting improved soundstage, improved details or whatever random stuff claimed without any sort of evidence by audiophiles.

the important part here, is that it's not the empty claim of some random guy on the web saying "trust me I heard it" under sighted conditions. it's been properly demonstrated that bone conduction of ultrasounds can lead to hearing something. there is no indication, so far, that the studies were flawed, so we believe that it's a fact. that's how it works.


----------



## ScareDe2 (Jul 17, 2020)

You can detect high frequency noise at negative decibel value and it will be converted to a noise closer to where the music plays.


----------



## pfzar (Jul 17, 2020)

WHAT?

Man, this has been interesting to follow. Certainly a lot of theories going on.  I think there a lot of scientists rolling over in their graves.


----------



## danadam

jlawler said:


> I must admit that given some of the reception my post got I do doubt you will accept my evidence, but I went ahead and did it anyways.  I created the 16/44 from the 24bit version using sox -V3 -v 0.99 -G -b 16 rate -v -L 44100.




```
]$ sox "09 Don't Bring Me Down.flac" -n stats
             Overall     Left      Right
...
Pk lev dB      -3.08     -3.08     -3.30
RMS lev dB    -18.72    -18.46    -18.99
RMS Pk dB     -15.07    -15.07    -15.51
RMS Tr dB     -99.03    -98.14    -99.03
```


```
]$ sox "1644 Don't Bring Me Down.flac" -n stats
             Overall     Left      Right
...
Pk lev dB      -3.33     -3.44     -3.33
RMS lev dB    -18.82    -18.56    -19.08
RMS Pk dB     -15.17    -15.17    -15.60
RMS Tr dB     -94.56    -94.09    -94.56
```
Here's the null of your files:

Try the conversion without "-v 0.99 -G". Then you'll get something that's a better match:

```
]$ sox "1644 proper.flac" -n stats
             Overall     Left      Right
...
Pk lev dB      -3.24     -3.36     -3.24
RMS lev dB    -18.73    -18.48    -19.00
RMS Pk dB     -15.08    -15.08    -15.51
RMS Tr dB     -94.47    -94.04    -94.47
```
And the resulting null:


And as others said, for ABX test convert it back to 24/96 and compare that with the original.


----------



## sonitus mirus

danadam said:


> And as others said, for ABX test convert it back to 24/96 and compare that with the original.



Fully agree with this part as has been mentioned by @bigshot as well.  

This will best remove the behavior of the device that is playing back the different files.  It is possible the Denon AVR is upconverting the 16/44 file to 16/48 internally, which could be creating a noticeable difference in the audio or perhaps some unassuming cue from the hardware.   I know that the Pono player would play 16/44 files using a completely different type of filter that would roll off very early compared to the filter used with 24/96 files.  While I might not be able to hear any difference with 1-2dB around 12kHz, some people surely can and the comparison is not apples to apples with regards to potential audible differences with the file types.

http://archimago.blogspot.com/2015/08/measurements-ponoplayer-another-mans.html

Thanks @jlawler for participating. You have my respect for actually taking the time and effort to do the test and post the results.


----------



## KeithPhantom

ScareDe2 said:


> You can detect high frequency noise at negative decibel value and it will be converted to a noise closer to where the music plays.


What does that even mean? Negative decibel value relative to what? Decibels are a measure of ratios in logarithmic scales, they need a reference to make sense.


----------



## bigshot

I think he means that ultrasonic noise can create harmonic distortion down in the audible range if the playback equipment isn’t designed to work with super audible frequencies. He’s parroting something without understanding it.


----------



## KeithPhantom (Jul 17, 2020)

bigshot said:


> I think he means that ultrasonic noise can create harmonic distortion down in the audible range if the playback equipment isn’t designed to work with super audible frequencies. He’s parroting something without understanding it.


He means intermodulation between aliased and passband frequencies when a simple reconstruction filter isn't used or not used correctly. It can happen, but you are hearing only the intermodulation that is in main audio band (20 Hz-20kHz) as distortion, not the supersonic frequencies, and that DAC has to be pretty much broken for this to be even noticeable.


----------



## bigshot (Jul 17, 2020)

Audibility doesn’t matter to him I think. I’m not used to wrapping my head around fatal flaws that are imperceptible.


----------



## ScareDe2

bigshot said:


> Audibility doesn’t matter to him I think. I’m not used to wrapping my head around fatal flaws that are imperceptible.


----------



## KeithPhantom

ScareDe2 said:


>



If you are using Paul as evidence, you're talking about vested interests that are openly expressed by himself. Also, tell him he should accept the public debate Ethan Winer challenged him.


----------



## ScareDe2

I thought the debate was closed. If we use good over-ear headphones or good speakers with high definition audio (expensive or not) we detect the high frequency noises even if the signal is weak. That noise is being processed by the brain and relocated inside the frequency range of the music. I would say it is rather felt/perceived than heard. With noise filter it sounds more gentle on the ear and more enjoyable. By the way, I do not agree with everything Paul says I just think his videos are fantastic and deserve mentions.


----------



## KeithPhantom

ScareDe2 said:


> we detect the high frequency noises even if the signal is weak.


What do you mean by "high frequency"? What is a "weak" signal in your example?



ScareDe2 said:


> That noise is being processed by the brain and relocated inside the frequency range of the music.


There is a century of research showing how the upper limit of human hearing is 20 kHz, if you want to go against it, you better present evidence and your methodology to be scrutinized by the public. 



ScareDe2 said:


> With noise filter it sounds more gentle on the ear and more enjoyable.


What do you mean by filter? Are you using interpolation so you can use a less abrupt reconstruction filter? Is it linear-phase or minimum phase, or other type of filter? How do you know it sounds "gentler"?

Finally, why do you worry about differences that are usually -100 dBFS and can be masked by the distortion of your transducers? How do you separate transducer and electronic distortion by only "hearing"? Are ears more sensitive than AP analyzers?


----------



## ScareDe2

I have already answered everything.

As for the transducer distortion, it's close to the intent of the producer who himself worked with transducer distortion. The window resampling noise peak around 20Khz is a distortion you want to get rid of.


----------



## KeithPhantom

ScareDe2 said:


> As for the transducer distortion, it's close to the intent of the producer who himself worked with transducer distortion.


What do you mean by this? So if the distortion is high in a specific transducer, the "producer" wanted to engineer something subpar?



ScareDe2 said:


> The window resampling noise peak around 20Khz is a distortion you want to get rid of.


Can you show evidence that this is a problem? I can ask @amirm who actually was the director in charge of the Windows Audio stack and see if this is even true? Even if it is true, at what level is the distortion located?


----------



## KeithPhantom

ScareDe2 said:


> I have already answered everything.


You only addressed one of my points. What about the other questions?


----------



## bigshot (Jul 18, 2020)

PS Audio is full of crap. It's all woo woo snake oil sales pitch. If that's your source, you shouldn't even be in Sound Science. Inaudible is inaudible. If he wants to claim that he can hear things that a century of science has proven can't be heard, he should submit himself to a fair test. If he ducks it and turns around and tries to sell people something based on that bologna, then I get to call him on it. That guy is full of it. And there's a word for people who continue to listen to him after the facts have been explained to them. But I'll get in trouble if I say it. See if you can guess what the word is. You can lead a horse to water, but you can't make him think.


----------



## KeithPhantom

I don't know why people care so much of stuff that is in inaudible territory, stuff that's even below than the background noise of their place. Also, some think their ears are more sensitive than a microphone or an Audio Precision analyzer and think audio has a magical property that hasn't been discovered. Audio has pretty much been figured out, the equipment we have nowadays is capable of exceeding any human spec and the tools we use to design and measure them too.


----------



## bigshot (Jul 18, 2020)

The reason that snake oil salesmen like Paul keep pointing at things outside the range of human hearing is because that is the easiest place to plant expectation bias. If you can actually hear something, you can hear it. That might conflict with what he is telling you you should expect it to sound like. But if he points at gamma rays or sunspots or plasma fields or supersonic frequencies with gigantic numbers representing them, the sucker has no frame of reference about what any of that sounds like, and there is no actual sound there to break the spell. That leaves the field wide open for bias to paint rosy, candy colored paintings all over the empty aether. If you want to sell something to someone that isn't really what you're telling them it is, you'll have a lot less blowback if you sell them something that they can't touch, hear, feel or see. Woo woo voodoo magic is the best place to plant the seeds of expectation bias. It's the manure where delusions grow and blossom. Suckers aren't interested in optimizing the sound they can actually hear. They are interested in pursuing magical thinking because it makes them feel like they know more than "experts". Classic Dunning Kruger.


----------



## miksu8

gregorio said:


> 1. Processing virtually always occurs at 32bit or 64bit, regardless of whether the source files are 16bit or 24bit. So, the artefacts are at exactly the same level regardless of a 16bit or 24bit source!
> 
> 2. Sure, but of course that's the freq response of your HPs (or speakers + room), the freq response of your ears and your personal preferences, which has nothing to do with the source file's bit depth!
> 
> ...



I agree that there is some advertisement jargon in Rob's posts. About dithering - how is it possible to dither 1 bit DSD? Is it somehow converted to more bits so that dithering becomes possible?


----------



## gregorio

miksu8 said:


> [1] About dithering - how is it possible to dither 1 bit DSD?
> [2] Is it somehow converted to more bits so that dithering becomes possible?



1. Dithering is a probability function applied to a batch of samples, noise-shaped dithering in the case of 1bit DSD. So it can be applied at any arbitrary amount/level. Typically you'd need 1 - 2bits of dither to fully linearise all quantisation error, which obviously isn't possible if you've only got 1 bit to start with. So, with 1-bit DSD, it's applied at a level significantly less than 1bit (the equivalent of around 0.17 bits rings a bell but I could be mistaken), which therefore does NOT fully linearise all quantisation error. In other words, it is NOT possible to properly dither 1 bit DSD, there will still be some quantisation error distortion. In controlled tests though, this distortion is not audible.

2. This is what nearly all DACs do, they're multi-bit (typically about 6bit) delta/sigma converters. However, you can't convert 1bit to multi-bit, apply dither and then convert back to 1 bit DSD, because the conversion back to 1 bit would obviously introduce 1 bit quantisation error again and we'd be back where we statred.

If you're interested in the details, I'd recommend Lipshitz & Vanderkooy, who've published several papers on the subject of dithering 1-bit DSD. Also, Gerzon & Craven, who published the mathematical proof/rules of noise-shaping at the end of the 1980's.

G


----------



## GearMe

ScareDe2 said:


> I have already answered everything...




So here's the challenge you face...

This forum will often throw the 'prove-it', 'show me the data', etc. argument at anyone voicing an opinion that is considered different than the accepted views held in the Sound Science forum.  Fair enough...the forum's title surely indicates the group's focus here. 
(FWIW, I tend towards a Sound Science 'mindset' for amps/dacs/codecs/etc given a background in engineering/business/data analytics,) 

Additionally, you will make absolutely no headway with this forum if you talk about 'inaudible' frequencies having an impact on your listening experience...unless you have the data from a properly executed test...subjected to peer review (i.e. AES, etc.).  Or, if it's your data from a test you've run, prepare to have your methodologies heavily scrutinized.  And...if they don't meet the group's standards, then either back to the drawing board for you or 'thanks for playing' which could be frustrating to a newcomer.  

So, given this rinse/repeat scenario, I've often found myself wondering what the benefit/purpose of this small group really is for the Head-Fi community. 

_Is it for helping people along the journey of gaining a better understanding of the Science behind the Sounds we hear through our systems?_ 

Which...would be a noble effort requiring high levels of patience, positivity and mentoring skills from the group!   
Like this 





sonitus mirus said:


> Thanks @jlawler for participating. You have my respect for actually taking the time and effort to do the test and post the results.




Of course, that would be awesome!  The end result of this approach would, most likely, be an overall increase in the knowledge-base of Head-Fi and also the size of this group as its reputation for being helpful SMEs spread throughout the rest of Head-Fi.

_Or is it something entirely different? _
Only you can judge how this interchange is/has been beneficial to you.  If you find the dialogue helpful...that's great.  If not...don't expect much to change -- the 'data' says otherwise.


----------



## Davesrose

miksu8 said:


> I agree that there is some advertisement jargon in Rob's posts. About dithering - how is it possible to dither 1 bit DSD? Is it somehow converted to more bits so that dithering becomes possible?



Noise shaping and 1-bit converters

"Most A/D converters made since 2000 use multi-bit or multi-level delta sigma modulators that yield more than 1 bit output so that proper dither can be added in the feedback loop."


----------



## bigshot (Jul 18, 2020)

GearMe said:


> _Is it for helping people along the journey of gaining a better understanding of the Science behind the Sounds we hear through our systems?_



The problem is not with the group. It's with individuals who come here with the express purpose of trolling us. Look at the patterns... The trolls generally arrive in pairs. They draw in people to spend a lot of time explaining things to them. Then they totally ignore what is shared with them and go on repeating their false claims. They shift from one subject to another, jumping from one thread to another to plant seeds of chaos, then sit back and fan the flames. One of the pair will take the lead until that one's grip on the individual posting style he's adopted begins to falter, then he'll disappear and the other one takes over like a tag team. Their arguments always are the same... we can *perceive* ultrasonic frequencies through bone conduction/brain waves/whatever. We can hear things that can't be measured. ABX isn't the best way to judge sound. Science doesn't know everything. You aren't scientific enough... It's like a broken record. Ask them to cite an authority and it's always the same snake oil salesman in the same youtube video. I think it's pretty clear that these aren't all individual people. One person, or a small handful of people, are coming back over and over again under different aliases just to jack with us.

Everyone agrees that it's nice to be nice. If someone comes to us with the sincere interest in doing the research, and expending the energy to understand, they get a LOT of help from this group. Gregorio's posts are encyclopedias of information if someone wanted to take the things he talks about and do a little googling to figure them out. I see names and avatars here of people who started out arguing with us and realized their error and now have joined in agreeing with us. They rarely admit it outright, but it's pretty easy to see what they learned here. I have one win under my belt helping a doubter conduct his own listening test and figuring things out for himself. If we are approached properly, someone could gain a lot from this group.

But there's no respect gained without respect offered. I'm not going to waste my time tossing pearls to the swine. None of us have to be here. We post because we want to help. That is a given. But the person needs to want to be helped first. If they don't and they're just here to mess with us, the gloves come off. Nothing wrong with that. Eventually it isn't fun for them to jack with us any more and they go away. We should think about how we can help accelerate that process.


----------



## ScareDe2

Inaudibility is fake news. There are many undergoing researches about ultrasounds and their impact on humans. Your claims only rely on audiometry using air conduction. Science is not settled about an audible limit. Science is not settled about anything in fact not even what is a women or a men. It uses the 20-20khz conventionally because this is what is commonly observed in practice.


----------



## bigshot (Jul 18, 2020)

You make this stuff up. You used to not know much and just went on your feelings. Now you're Mr Science and you know all about "cutting edge research". Your mask is slipping. Time to jump to another user ID.


----------



## Davesrose

Sorry, science itself has settled on what is a male vs female vs asexual species.  Now recently we have been having an awareness of psychological sex identification (but anatomy is pretty conclusive).  I can't name call...but 20hz-20khz is the accepted auditory range for air conduction (a few studies indicate maybe down a few more hz).  The tests that have indicated ultra high frequency hearing are special circumstances (most readibly, divers being able to hear 100khz+).  When it comes to high frequencies, instruments don't have fundamental notes going to 20khz, and LFE goes from 40 or 37hz and starts rolling down.  There's also a wide variety with individual anatomy: ear shaping is different with folks and with age we have reduced frequency range hearing.  I am fortunate enough to have pretty good hearing for my age (in my 40s and hear up to 17khz....and I know that's my max and I don't have super abilities past that).


----------



## bigshot




----------



## Davesrose (Jul 18, 2020)

Bigshot, you're giving too much credit!  That chimp is enough a chemist to have the conditions to not have exploded the lab.


----------



## KeithPhantom

For now, it seems he got something correct, there is a gradual increase of intermodulation distortion in the interpolation implementation of Windows Audio Stack due to poor filtering (potentially audible since IMD is around -40 dBFS). Archimago measured the Windows Audio Stack and Linux (PulseAudio), Linux showing a pretty good result. Here the link: https://archimago.blogspot.com/2015/11/measurements-windows-10-audio-stack.html


----------



## bigshot (Jul 18, 2020)

Get a Mac.

See the video link in my sig for a demonstration of what -40dB actually sounds like. For the purposes of playing music in the home, that is inaudible.


----------



## KeithPhantom

bigshot said:


> Get a Mac.


Linux and ALSA have me covered by now, but I'm thinking about switching.


----------



## Davesrose (Jul 18, 2020)

KeithPhantom said:


> For now, it seems he got something correct, there is a gradual increase of intermodulation distortion in the interpolation implementation of Windows Audio Stack due to poor filtering (potentially audible since IMD is around -40 dBFS). Archimago measured the Windows Audio Stack and Linux (PulseAudio), Linux showing a pretty good result. Here the link: https://archimago.blogspot.com/2015/11/measurements-windows-10-audio-stack.html



I'm not really sure how relevant this is for modern implementations?  The article is referring to Direct Sound....something that has been depreciated.  Windows has been relying on WASAPI.  I have a HTPC that I use for 4K HDR and it can bitstream audio formats to Dolby Atmos/DTS:X.  They are all apps that do adhere to current device standards.  Edit: RE Mac...actually, I do have Macs for development, but have been PC user for 3D animation.  One thing I have found is that PC tends to be better about HDMI output for Dolby standards (because of the new standards I've just mentioned).


----------



## KeithPhantom

Davesrose said:


> I'm not really sure how relevant this is for modern implementations?  The article is referring to Direct Sound....something that has been depreciated.  Windows has been relying on WASAPI.  I have a HTPC that I use for 4K HDR and it can bitstream audio formats to Dolby Atmos/DTS:X.  They are all apps that do adhere to current device standards.


WASAPI should be a direct connection with decoder (DAC), so I wouldn't expect to have trouble in there. But the fact that this is present in a modern OS due to possible "hardware constraints" (I bet my old Pentium 4 can do interpolation and filtering correctly without skipping a beat) isn't good.


----------



## Davesrose (Jul 18, 2020)

KeithPhantom said:


> WASAPI should be a direct connection with decoder (DAC), so I wouldn't expect to have trouble in there. But the fact that this is present in a modern OS due to possible "hardware constraints" (I bet my old Pentium 4 can do interpolation and filtering correctly without skipping a beat) isn't good.




Still, look at the date of your link: 2015.  It is outdated itself.  I'm agnostic when it comes to PC vs Mac, but when it comes to multimedia, I think Windows 2018 Creators update probably was the best all round setup for video and audio (IE Dolby Atmos, native HDR, etc).  Me thinks your link about audio processing for 2 channel upsampling has changed as well.


----------



## KeithPhantom

Davesrose said:


> Still, look at the date of your link: 2015.  It is outdated itself.  I'm agnostic when it comes to PC vs Mac, but when it comes to multimedia, I think Windows 2018 Creators update probably was the best all round setup for video and audio (IE Dolby Atmos, native HDR, etc).  Me thinks your link about audio processing for 2 channel has changed as well.


True, still, I don't trust Windows for privacy stuff. Also, I like how Linux feels and how open it is, bit I agree it has being improving these years.


----------



## bigshot

Let's just award him the Nobel Prize anyway.


----------



## Davesrose (Jul 18, 2020)

KeithPhantom said:


> True, still, I don't trust Windows for privacy stuff. Also, I like how Linux feels and how open it is, bit I agree it has being improving these years.



There's quite a few resources about what to disable for not having any tracking.  My biggest complaint with Windows is why they have to keep changing their simplified control panel...to being even more unusable.  At least there's a link to the classic control panel.    Since getting into software development, I've been starting to default with Unix CLI and use Git for PC and Terminal for Mac.  Most interesting thing I have found with my relatively new MacBook Pro: it has a T2 chip: which prevents 3rd party OS from being installed (so Linux kernel crashes....there's an unofficial hack that *may* work).  You're left with Mac versions and keeping to Windows in Bootcamp (so I do have a Windows 10 Pro instance for having HDR and Atmos output with HDMI if I want to present movies on the go).


----------



## ScareDe2

Make Sound Noise-Free Again.


----------



## bigshot

ScareDe2 said:


> Make Sound Noise-Free Again.



Make this forum and the threads in it noise free.


----------



## gregorio

ScareDe2 said:


> [1] Inaudibility is fake news.
> [2] There are many undergoing researches about ultrasounds and their impact on humans.
> [3] Your claims only rely on audiometry using air conduction.
> [4] Science is not settled about an audible limit.
> ...



1. Everything is audible then .... Great, humankind doesn't need to spend billions on LIGOs, particle accelerators, radio telescopes, etc., we just need to get an audiophile to listen, for little more than the price of a Starbucks! 

2. There's some ongoing research in the field of bone conduction and ultrasound. There's pretty much none in the field of the audibility of air pressure ultrasound because it's already been done, countless times.

3. Huh? Yes of course it is, this website is called "Head-Fi", not "Bone-Fi"!  However, the proliferation of utterly ridiculous assertions/posts on this website indicates that maybe it should be called "Bonehead-Fi"! Consumer audio reproduction headphones/speakers output air pressure (sound) waves, didn't you know that?

4. This statement is a lie, it's been settled for many decades.

5. This statement is another lie. For example, do you really think that science "_is not settled about_" 1 + 1 = 2 ? Science is obviously "settled" about a great number of things.

6. No it's not! Science uses 20Hz - 20kHz because that's the maximum EVER observed in practice, the maximum observed in practice for adults actually averages more like 20Hz - 16kHz. It can be observed that some humans can detect air pressure (sound) waves up to around 25kHz or so, but ONLY with specifically manufactured artificial signals under specific laboratory/test conditions. HOWEVER, such signals + conditions do NOT exist "in practice" (either in the real world or in commercial music/sound recordings) and therefore, the assertion that "human hearing has a max range of 20Hz to 20kHz in practice" is NOT invalidated!

G


----------



## bfreedma

So our troll is now using the slogan of an anti-science politician to support his view of audio science...  IMO, it's time for our Mod to step in and put an end to this fiasco.

And time to stop responding to such weak and obvious attempts at disruption and attention seeking from someone who is clearly here just for that.  Unfortunately, even collectively, you can't fix willful ignorance.


----------



## magicscreen

It is very unfortunate the CD format is only 16/44. It is very limited.
Of course there is a big brain washing that 16/44 is enough.
They have to sell CDs.
Poor misguided children who have to listen digital format instead of analogue.


----------



## castleofargh

ScareDe2 said:


> Inaudibility is fake news. There are many undergoing researches about ultrasounds and their impact on humans. Your claims only rely on audiometry using air conduction. Science is not settled about an audible limit. Science is not settled about anything in fact not even what is a women or a men. It uses the 20-20khz conventionally because this is what is commonly observed in practice.


I need to put a stop to the off topic at least. Enough sample rate in the bit depth topic.
Anybody willing to continue that conversation, please do it there instead:
https://www.head-fi.org/threads/perception-of-ultrasounds.937809/#post-15749219 



@bigshot, I've seen @ScareDe2 on some french forums several years ago(thought the nickname was some witty frenglish). I think he's Canadian, maybe? And is probably just himself. If he isn't, you're still making statements/allusions about it without proof, in Sound Science... Give me some freaking evidence or don't say stuff like that.
Give me the money! No that's something else. 

Also, the biggest reason why it's hard to put 100% of the fault on @ScareDe2, is that you're playing the other side while making just as absolute statements and wild oversimplifications as he does. In that regard, neither of you is on the side of science, while both seem to argue that you are. You can clearly see that when he contests some of the stuff you said, it's mainly about your statements of all DAC sounding the same, ultrasounds being inaudible to humans, and all the stuff where it's so easy to find an exception to disprove your claim.

I know why you do it, I've argued against it too many times already, which is really the only reason why I mostly don't react anymore and maybe give the impression that I agree with you. But let's be clear that I do not like your oversimplifications. I don't think they should be the go to answers of a science oriented forum.
 You say it's to avoid confusion for newbies so they can go focus on the important stuff, and that's a valid point for sure. But when it becomes the very reason why people disagree with you, oversimplifying has officially become the problem.


----------



## jlawler

GearMe said:


> So here's the challenge you face...
> 
> _Is it for helping people along the journey of gaining a better understanding of the Science behind the Sounds we hear through our systems?_
> 
> Which...would be a noble effort requiring high levels of patience, positivity and mentoring skills from the group!


I am obviously new here, but in my experience so far, I can say, sadly, _mostly _no...   I have had to endure baseless accusations of impropriety (even though I gave an extreme amount of detail about what I did - not trying to hide anything).  

Also, I have to ignore baseless assertions that I am not myself but instead someone else masquerading as another (which could be easily confirmed or denied by asking a mod to check by looking at access to my account - if the accuser had such a good faith suspicion).  I'm sure this influeneces the opinion of others regarding me in a negative way and this is completely unfair as well as unwelcoming (which I had done nothing to deserve).



> Of course, that would be awesome!  The end result of this approach would, most likely, be an overall increase in the knowledge-base of Head-Fi and also the size of this group as its reputation for being helpful SMEs spread throughout the rest of Head-Fi.
> 
> _Or is it something entirely different? _
> Only you can judge how this interchange is/has been beneficial to you.  If you find the dialogue helpful...that's great.  If not...don't expect much to change -- the 'data' says otherwise.


Why should I have to enter a gauntlet in order to have a conversation?  I don't think it encourages dialogue or community.


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## KeithPhantom (Jul 19, 2020)

bigshot said:


> See the video link in my sig for a demonstration of what -40dB actually sounds like. For the purposes of playing music in the home, that is inaudible.


I didn't want to express my disagreement with this, but as a person at the other side of a device that allows us to communicate, I would like to express that I do not agree with you on this statement. -40 dBFS (assuming 0 dBFS to be a reasonable 80 to 90 dB SPL) is noticeable with music and test tones (watch Ethan Winer's video where he does this experiment). A DAC with distortion in this range may or may not be different than other DACs depending on the frequencies involved, the intermodulation/harmonic distortion, and the noise level. This is supported by the collective effort of ASR that measured and did listening test with an occasional ABX, where it was found that many horribly measuring DACs (SFDR around 40 dB) either didn't change the sound or just did it slightly. There are outliers in every situation (just read the paper where 3 out of 32 subjects could detect in lab-perfect conditions a signal of 28 kHz, but I consider this an outlier, not definite evidence of ultrasound detection capabilities by humans), and these are expected out of every situation, even from many established opinions in science. Science will change if evidence is presented, the methodology is scrutinized, and results are replicated, science isn't static.

Sorry @castleofargh I wanted to express my opinion, I'm not talking about ultrasonics or inaudibility, I'm now restricting myself to talk about bit depth in this specific thread.


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## bigshot (Jul 19, 2020)

castleofargh said:


> I think he's Canadian, maybe? And is probably just himself. If he isn't, you're still making statements/allusions about it without proof, in Sound Science.



The forum displays IP addresses to mods. That is the way to test it. I always kept an eye on IP addresses when I was modding forums back in the day. It's very useful.

Perhaps it's two knuckleheads coordinating their efforts over PM. It's the same thing. If he isn't a troll, he's the other thing. No real difference between those two things in practice.



KeithPhantom said:


> -40 dBFS (assuming 0 dBFS to be a reasonable 80 to 90 dB SPL) is noticeable with music and test tones (watch Ethan Winer's video where he does this experiment).



Ethan Winer uses the absolute worst case scenario... a horrible buzzing sound under very quiet cello music alternating on and off. It doesn't get any worse than that. And -40 is the last point where the sound is really at all audible under the music. With pop music, it wouldn't be audible. With distortion that followed the modulations of the music, it wouldn't be audible. If it was continuous your ears would quickly hear past it.

I agree with you that under some worst case circumstances -40dB is audible. But your living room probably has a noise floor that high. If you take your portable rig out, street noise will be much higher. You deal with noise greater than that all the time and don't even think about it. People listen to LPs with noise in that range and claim it sounds better than CDs. There is such a thing as not perfect but good enough to do the job.

Audio people always want to draw hard and fast lines to define what is "acceptable" and what isn't. Then they sit down and think of unlikely extreme cases to give them an excuse to push that line a little further. Then an even less likely one to push it a little further... pretty soon we are in Amir territory. It's important to keep a handle on *relative importance* when we talk about thresholds.

-40dB is about the last point that it even matters at all any more- and then only in worst case situations. And the since decibel scale is logarithmic the difference between -40dB and -96dB is massive. No reason at all to worry about CDs. And noise and distortion levels in modern cheap amps is just as low if not more so. What's the problem here? There are much bigger fish to fry if you want great sound that this stuff. That's my point.


----------



## jlawler

sox -V3 -b 16 rate -v -L 44100
sox -V3 -b 24 rate -v -L 44100

https://gofile.io/d/XzTems

foo_abx 2.0.6d report
foobar2000 v1.5.1
2020-07-19 20:56:20

File A: 09 Don't Bring Me Down.flac
SHA1: a19b2743680e319eaee3c8bcc08867e475848723
Gain adjustment: -3.06 dB
File B: fake24 Don't Bring Me Down.flac
SHA1: ca846c3afe2c4447b1355e542f2666ba7d120c72
Gain adjustment: -3.06 dB

Output:
WASAPI (event) : DENON-AVR (2- NVIDIA High Definition Audio), 24-bit
Crossfading: NO

20:56:20 : Test started.
20:56:40 : 01/01
20:57:03 : 02/02
20:57:24 : 03/03
20:57:48 : 04/04
20:58:23 : 05/05
20:58:45 : 06/06
20:58:58 : 07/07
20:59:16 : 08/08
20:59:16 : Test finished.

 ---------- 
Total: 8/8
p-value: 0.0039 (0.39%)

 -- signature -- 
ee60e9789ab7aa2df76c1e6f78cc6ff5b203c7f8


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## dazzerfong (Jul 19, 2020)

jlawler said:


> sox -V3 -b 16 rate -v -L 44100
> sox -V3 -b 24 rate -v -L 44100
> 
> https://gofile.io/d/XzTems
> ...



Just had a look at the files - one's in 96000 Hz, the other is 44100 Hz.

If you're testing for bit depth only, keep the sample rates the same. Otherwise, and not surprisingly, different sample rates are interpolated and sample slightly by DACs. In extreme cases, like with this random old DAC I have lying around, the music is 'sizzly' with 96000 Hz source as it downsamples to 44100 Hz poorly.

If I was cynical I'd say that your AVR says what the sampling rate is on its readout, but I'm willing to take the benefit of doubt here. If you're interested, do the following:

- Start with the same source file
- Only touch the bit rate - leave the sampling rate the same. Bump one down to 16-bit, then open that 16-bit and save it as a 24-bit file.
- Save both as WAV files. That way, the files are equal in size and there's no compression whatsoever (lossless or otherwise)

Have fun!


----------



## bigshot (Jul 19, 2020)

I think you need someone to independently verify your results. I would like to see you do this with a test conducted by an independent person. I'm sure you can get volunteers here in this group if you let us know where you are located.

I don't use this ABX software. How would you go about gaming it? Like this? https://hydrogenaud.io/index.php?topic=119399.0



dazzerfong said:


> If I was cynical I'd say that your AVR says what the sampling rate is on its readout



My AVR has a handy screen that shows exactly what it is playing right there on the TV in front of you. Wouldn't even take 3 minutes to burn through a test like this that way.


----------



## dazzerfong (Jul 20, 2020)

bigshot said:


> I think you need someone to independently verify your results. I would like to see you do this with a test conducted by an independent person. I'm sure you can get volunteers here in this group if you let us know where you are located.
> 
> I don't use this ABX software. How would you go about gaming it? Like this? https://hydrogenaud.io/index.php?topic=119399.0
> 
> ...



His method makes no sense - if you change the volume, you by nature changed the file. The file hash will be completely different.

If he was inclined to cheat it, he'd have to reverse-engineer the hash algorithm of foobar's ABX plugin. Doing a file swapperoni would be obvious through the hash. Until proven otherwise, not going to accuse him of that.


----------



## bigshot (Jul 20, 2020)

I'm Mac so I don't know how Foobar works. But I see people saying that the purpose is to find out for yourself, not proving anything to others. This guy is bound and determined to prove he can tell a difference in less than 20 seconds. I know how close that really is. The readout of the AVR then.


----------



## gregorio

magicscreen said:


> [1] It is very unfortunate the CD format is only 16/44. It is very limited.
> [2] Of course there is a big brain washing that 16/44 is enough.
> [3] They have to sell CDs.
> [4] Poor misguided children who have to listen digital format instead of analogue.



1. It is indeed very limited but that is NOT "unfortunate", it's the opposite!

2. Indeed, there was a lot of marketing that 16/44 was enough. Then there was a lot of marketing/brain washing that it wasn't enough and Hi-res was required. So, how can we tell which is brain-washing/marketing BS and which is the actual facts/truth? Hint: The answer is the name of this subforum!

3. True but they also have to sell hi-res playback equipment and content.

4. You have this backwards, even school children are taught that we listen to acoustic sound waves, not digital or analogue audio. And, didn't you know that "digital format" is converted from/to analogue?



jlawler said:


> [1] Why should I have to enter a gauntlet in order to have a conversation?
> [2] I don't think it encourages dialogue or community.



1. You don't have to "_enter a gauntlet_" in order to have a conversation. However, if you are going to make assertions of fact that are contrary to the established science/facts, that's an entirely different matter and OF COURSE you have to "_enter a gauntlet_"! If you didn't have to "_enter a gauntlet_" then human knowledge would be based SOLELY on whatever anyone wanted to claim, which of course is why science exists in the first place. It has evolved into a strict, onerous "gauntlet" (called the "scientific method") which represents humankind's most accurate and reliable method of separating fact from fiction, that discourages even the most sophisticated charlatans.

2. Absolutely, that's the point! More precisely, it's to discourage the dialogue of false information being presented as fact and obviously, if this in turn discourages a community that relies on false information presented as fact, then that's their problem.



jlawler said:


> sox -V3 -b 16 rate -v -L 44100
> sox -V3 -b 24 rate -v -L 44100
> https://gofile.io/d/XzTems



Firstly, the two files you posted appear to be of slightly different length and only null by about -20dB! I don't have access to my studio and usual tools but I took your 96kHz original, converted it myself to 16bit (at home on my old laptop and freeware editor) and it nulled beyond the limit of Audacity's meters (-60dB). 

Secondly, it's trivially easy to discern 16bit from 24bit! Just find a particularly quiet part of the song, usually the last one or two seconds where it fades to digital silence, whack the volume up by about 30dB or so and it's easy to discern a difference, even with internal laptop speakers. Of course though, +30dB does NOT comply with the condition of "reasonable listening levels". If you tried to listen to the whole song at +30dB either most of it would be nothing but total distortion or you'd damage your drivers or ears (as explained in the OP)!

I'm not accusing you of deliberately falsifying evidence, maybe you just made an inadvertent technical error with the files and/or inadvertently made the listening methodology error just described?

G


----------



## dazzerfong (Jul 20, 2020)

bigshot said:


> I'm Mac so I don't know how Foobar works. But I see people saying that the purpose is to find out for yourself, not proving anything to others. This guy is bound and determined to prove he can tell a difference in less than 20 seconds. I know how close that really is. The readout of the AVR then.



Seems like you searched 'ABX cheat foobar', saw the title of a thread that suits your beliefs, and went with it.

EDIT: called it!






See here's the thing I don't get. You demand proof for anything, someone offers it, instead of scrutinising you immediately throw it out and accuse someone of cheating without any discourse. With the ivory tower you've made, what's the point in engaging in discourse?


----------



## bigshot (Jul 20, 2020)

The reason I doubt is because you are saying that you can do in 20 seconds what no one else has been able to do under controlled conditions. If you want to follow the scientific method, the next step for you is to prove you can discern a difference when someone else is overseeing the test. If you aren’t cheating, I would suggest that you contact someone at the AES that can get your results published for peer review. You don’t inspire confidence in me, and the way to get me to accept your results is to do a double blind controlled listening test. Other people may be nice and give you the benefit of the doubt, but I won’t believe it just because someone with ten posts and a pseudonym on an Internet forum says it. But proving it to me in this group isn’t important. You should prove it in peer reviewed circles.

It was pretty easy to find that with a search engine wasn’t it. When did you first find that page?


----------



## jlawler

gregorio said:


> 1. You don't have to "_enter a gauntlet_" in order to have a conversation. However, if you are going to make assertions of fact that are contrary to the established science/facts, that's an entirely different matter and OF COURSE you have to "_enter a gauntlet_"! If you didn't have to "_enter a gauntlet_" then human knowledge would be based SOLELY on whatever anyone wanted to claim, which of course is why science exists in the first place. It has evolved into a strict, onerous "gauntlet" (called the "scientific method") which represents humankind's most accurate and reliable method of separating fact from fiction, that discourages even the most sophisticated charlatans.



The guantlet has been baseless accusations aimed at debasing me & discrediting me as a person - that is not part of the "scientific method."



> 2. Absolutely, that's the point! More precisely, it's to discourage the dialogue of false information being presented as fact and obviously, if this in turn discourages a community that relies on false information presented as fact, then that's their problem.


I have not presented any false information.



> Firstly, the two files you posted appear to be of slightly different length and only null by about -20dB! I don't have access to my studio and usual tools but I took your 96kHz original, converted it myself to 16bit (at home on my old laptop and freeware editor) and it nulled beyond the limit of Audacity's meters (-60dB).


I used standard tools (sox) in the manner recommended & disclosed my config parameters in case they needed any adjustment.  I made no modifications beyond that.  If you want to provide the 16bit version you made I can retest.



> Secondly, it's trivially easy to discern 16bit from 24bit! Just find a particularly quiet part of the song, usually the last one or two seconds where it fades to digital silence, whack the volume up by about 30dB or so and it's easy to discern a difference, even with internal laptop speakers. Of course though, +30dB does NOT comply with the condition of "reasonable listening levels". If you tried to listen to the whole song at +30dB either most of it would be nothing but total distortion or you'd damage your drivers or ears (as explained in the OP)!
> 
> I'm not accusing you of deliberately falsifying evidence, maybe you just made an inadvertent technical error with the files and/or inadvertently made the listening methodology error just described?
> 
> G


I did not change the volume nor did I use any tools to measure / display the bit rate or sample rate.  I just used my ears and the difference is very obvious to them.


----------



## KeithPhantom

gregorio said:


> Secondly, it's trivially easy to discern 16bit from 24bit! Just find a particularly quiet part of the song, usually the last one or two seconds where it fades to digital silence, whack the volume up by about 30dB or so and it's easy to discern a difference, even with internal laptop speakers. Of course though, +30dB does NOT comply with the condition of "reasonable listening levels". If you tried to listen to the whole song at +30dB either most of it would be nothing but total distortion or you'd damage your drivers or ears (as explained in the OP)!


So this is the reason I can hear the noise floor of many recordings, I know it's this because all songs in an album have a particular hiss, and if I start listening to another one, the hiss is different. It has being annoying to say the least. I listen at around 80 to 90 dB SPL with headphones. I've checked ground loops and faulty equipment, but even with my phone used as a source/DAC/AMP I can do it. I'm maxing out the digital volumes and using my amplifier as volume control. It also happens with music from Spotify and in multiple OS. I have transcoded my library twice (FLAC-ALAC-FLAC+downsampling and dither for files >44.1/16), it could be this.


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## bigshot (Jul 20, 2020)

KeithPhantom said:


> So this is the reason I can hear the noise floor of many recordings, I know it's this because all songs in an album have a particular hiss, and if I start listening to another one, the hiss is different.



That is most likely tape hiss or air conditioner noise in the recording venue. Gregorio is talking about boosting the volume +30dB over normal listening level. That means boosting the volume to the point that it is uncomfortably loud. People have been known to gain ride their tests and focus on the ring outs of the song to goose the volume enough to hear a difference. Foobar is not intended to be proof to anyone except the person taking the test. If someone wants to cheat, they can cheat. But ultimately, they're just lying to themselves. If you have to cheat to prove your point, why not just switch to a point that is defensible?

I've done hundreds of 24 vs 16 listening tests. I do one every time I finish a mix to make sure the master file is the same as the mix I am approving. This is not generally an issue that is questioned except by audiophiles who either have never taken the time to test for themselves, or are so deeply invested in their argument they feel the need to cheat to make their point.

If someone comes along and has superpowers and says "it's easy to tell the difference", I'm not going to just automatically believe them because they say it. People say all kinds of things that aren't true on the internet all the time. If their results are that much of an anomaly, I'd suggest that they get their results independently verified. They can feel free to get all huffy and offended and say "How dare you question my honor!" I really couldn't give a flying flip about that. We're all just avatars and typing. The internet isn't the field of honor. If you want respect, you have to earn it... and that takes more than just 10 or 15 posts trolling an internet forum.

If it's that easy to perform an extraordinary feat, an independent person with experience in the field should be conducting the test and observing the results. If it's true, it's a lot more important than just an argument in an internet forum. It should be shared with the scientific community and peer reviewed. Of course, if it's a lie, it's not worth anyone's time at all...

I never offer my own tests as proof to anyone else but me. My goal is to improve my own sound system and find out for myself. I'm not a research scientist. I encourage everyone to do controlled tests *for themselves*. It doesn't matter what other people think. People who are too invested in proving things to other people raise my suspicions to be honest. It smells like someone who is out to prove their point at all costs. If he gets mad and huffy, maybe he'll go away. He isn't interested in listening to anything anyone else says, so that would be the best of all options.

However, if this guy can actually hear 24 bit as distinctly different than 16, it doesn't change the huge body of research on this topic that has already been done. All it means is that for him there is a difference. That doesn't mean that it makes a lick of difference for anyone else in the world. If he wants to prove 24 is necessary, he needs to prove that lots of other people can tell the difference too.

And he needs independent verification.


----------



## bigshot (Jul 20, 2020)

All of this is silly anyway, because he's claiming to hear greater than -96dB on a recording made in 1976 on 24 track analogue tape that didn't have a noise floor that low. It's kind of obvious what's going on here...

Brrrrrruuuuuuucccceeeee...


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## jlawler (Jul 20, 2020)

What is silly is that some who "claim" to adhere to science are really more akin to religious zealots when presented with evidence that may call into question their beliefs.

It is not your right to smear me & as long as the mods allow it I will keep defending myself.  Actually, I will just ignore you from now on as you obviously have nothing positive to offer.


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## bigshot (Jul 20, 2020)

I have a better idea. If you really do have ears that are capable of hearing things that are inaudible for the rest of humans on Earth, you're wasting your time here. Go make an appointment with an audiologist and tell him that you can hear -100dB under music that peaks at 0dB. Ask him to verify it or refer you to someone who can. He will know what to do.

I see through you. And I'd bet I've seen through you before... probably in this very thread.


----------



## castleofargh (Jul 21, 2020)

jlawler said:


> What is silly is that some who "claim" to adhere to science are really more akin to religious zealots when presented with evidence that may call into question their beliefs.
> 
> It is not your right to smear me & as long as the mods allow it I will keep defending myself.  Actually, I will just ignore you from now on as you obviously have nothing positive to offer.


kind of modo stuff but mostly my personal feelings on it:
I tend to let people talk. This section, for various reasons, is going to have heated arguments, that's just a fact. I mean when a discussion becomes too heated about technical stuff, psychoacoustics, or testing protocols in the rest of the forum, it's not rare for admins to move the posts here without a word and we have to deal with it. It's like the science debate club where almost nobody knows how to have a debate, or do science. people have clearly very different concepts of what can constitute a proof, and also very different ways to handle their personal beliefs being questioned(the religious zealots of audio are obviously a thing). If I was to stop any and all level of what could be interpreted as a personal attack to comply with the rules, we'd better close this section of the forum.
That's where the report button becomes most relevant. When one thinks he's on HBO and someone else is going crazy thinking he's watching CBS with a defective "beep" machine, he can report the post and explain in what way that post shouldn't be left on the forum. It's a way to bring those posts to our(modos) attention and even if we decide to do nothing, in the long run we get a trail of what someone has been doing, so a justified report is never a waste of time. Just like IMO, telling people when they go too far is not a waste of time. It's the community trying to apply self discipline. Even me, I'm just a noname member who's been given an extra "delete" button. Those who expect me to handle everything without being told to, will get consistently disappointed ^_^.


Normal non modo post:
It's been mentioned briefly, but most people here will have more respect for someone who does bother to run some sort of controlled experiment. Yes we can easily cheat and it would be risky to just take ABX logs at face value no matter what they show. No, it's probably not going to change humankind's understanding of the world. But it's also a fact that most people will never bother trying. So the moment you did, you got promoted to the very unwillingly exclusive club of "we actually tried". so few people do, that IMO it is worthy of praise and increased respect.

Now of the 2 logs you gave so far, the first one had some conversion issue, the second still doesn't seem to specifically test bit depth.
And it's important to repeat what @gregorio said, that it's trivial to make 16 vs 24bit audibly noticeable. All we need is a track with a very quiet signal so we can increase the gain without paying for it, until the quantization noise becomes noticeable. so if that's the type of question we're trying to answer, well it's already been done and there is a consensus that we can pass.
So it's assumed that you're not going to do that, but will instead use a playback volume that's close to what you would usually use to listen to music. Otherwise we're answering a very different question when passing the ABX. Audibility here is just a matter of threshold, for the test to be relevant to the audiophile hobby, it needs to be done under typical listening conditions.
If even like that, you still can easily tell the difference, then it would be interesting to also try the 24bit bounced to 16 then back to 24. and to test that against the original. that test should result in guessing stats, so if actual 16 vs 24bit gives audible difference, it means that the source or the DAC are handling those files differently and are the cause for the noticeable difference instead of 16bit truncation. Which would also be interesting and a very important distinction.

So I hope you get that most people who keep saying that your test is not enough or is wrong, are in fact bringing up those distinctions and suggesting more tests based on the result you get each time, as a perfectly normal scientific approach would say to do. They're not all just trying to shut you down.


----------



## castleofargh

@bigshot stop it with the baseless accusations and obvious double standards. You cannot ask every guy with a belief to run a blind test or we won't listen to him, and then when the result of the blind test doesn't say what you want, declare that the guy is cheating and that nothing he's going to do will count as evidence.
Morally it's blatant dishonesty. And for the standards of this forum, you should mention the possibility of cheat as one of several hypotheses, not act like it's a fact when you cannot support your claim with evidence. So far, both logs he submitted were with files that don't really fit the requirements for 16 vs 24bit tests, meaning that there could be a bunch of reasons why he's passing even if he's not cheating. Not considering those possibilities is ignorance/cherry picking. I don't think we're supposed to do that. Certainly we shouldn't when talking in the name of science.


----------



## ScareDe2 (Jul 21, 2020)

castleofargh said:


> we'd better close this section of the forum.



Although I would like to take a moment to criticize this moderation as being one of the worst I ever saw, I think this section will not develop any rails to get up on, so you might as well just close it, and y'know, call it a day.


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## Hifiearspeakers

ScareDe2 said:


> Although I would like to take a moment to criticize this moderation as being one of the worst I ever saw, I think this section will not develop any rails to get up on, so you might as well just close it, and y'know, call it a day.



The moderator has been more than fair here.


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## dazzerfong (Jul 21, 2020)

ScareDe2 said:


> Although I would like to take a moment to criticize this moderation as being one of the worst I ever saw, I think this section will not develop any rails to get up on, so you might as well just close it, and y'know, call it a day.



Rich coming from you. You're like an arsonist that complains that a firefighter did a bad job of putting out your fire.



jlawler said:


> I did not change the volume nor did I use any tools to measure / display the bit rate or sample rate.  I just used my ears and the difference is very obvious to them.





jlawler said:


> What is silly is that some who "claim" to adhere to science are really more akin to religious zealots when presented with evidence that may call into question their beliefs.
> 
> It is not your right to smear me & as long as the mods allow it I will keep defending myself.  Actually, I will just ignore you from now on as you obviously have nothing positive to offer.



Unfortunately you didn't address the elephant in the room: your files aren't properly made. To the skeptical outsider, it seems as if you're cheating. I simply believe that you're running into a bout of inexperience.

If you want, I can provide some properly-processed files for you. If you don't trust me, simply ensure the frequency and bit rate of the end files are the same.


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## sonitus mirus

jlawler said:


> What is silly is that some who "claim" to adhere to science are really more akin to religious zealots when presented with evidence that may call into question their beliefs.
> 
> It is not your right to smear me & as long as the mods allow it I will keep defending myself.  Actually, I will just ignore you from now on as you obviously have nothing positive to offer.


You are making an extraordinary claim that goes against well-established science.  For your observations to hold any credence, you would need to demonstrate exceptional evidence.  

The device under test has still not been ruled out as a factor in the most basic of methods.  It was mentioned above that one file was 44.1kHz and the other 96kHz.   One way to help eliminate the possible issue with your device creating some tell by switching between the sampling rates would be to convert to 16/44 and then back again to 24/96.  This way, both test files will be in a 24/96 FLAC shell, but the converted file would only have 16/44 data.  There would be a lower chance that something is identifiable when switching back and forth between the test files.

Then, if your files check out and you still identify differences indicated in your test logs, unfortunately the most likely explanation is still not that there is an audible different between 16 and 24 bit audio files in normal listening conditions.  We would need more information about how your were conducting the test and the equipment used.  Maybe this would be enough information to explain your results.

Also, having a neutral party proctor the test would be another method to help eliminate some additional questions about the authenticity of the test.

What you have provided is not significant evidence with regards to science, and does little to strengthen any case that those that seek to learn from science are similar to a religious zealot.  This position only emboldens many of us to consider your motivations for participation in this form to primarily be to troll.


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## magicscreen

It is a known story that Philips wanted CD audio to be 14-bit while Sony wanted 16-bit.

Then would be a scientific topic here: 14 bits are enough, we do not need 16 bits.


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## KeithPhantom

magicscreen said:


> It is a known story that Philips wanted CD audio to be 14-bit while Sony wanted 16-bit.
> 
> Then would be a scientific topic here: 14 bits are enough, we do not need 16 bits.


It wasn't because it was enough (perceptually there's a slim to no difference for most listeners), it was because Philips only had a 14-bit DAC.


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## gregorio

jlawler said:


> [1] The guantlet has been baseless accusations aimed at debasing me & discrediting me as a person -
> [1a] that is not part of the "scientific method."
> [2] I have not presented any false information.
> [3] I used standard tools (sox) in the manner recommended & disclosed my config parameters in case they needed any adjustment.  I made no modifications beyond that.  If you want to provide the 16bit version you made I can retest.
> ...



1. I personally have NOT debased or discredited you directly as a person, although I have made an accusation aimed at discrediting/debasing the information you've presented. However, it is NOT a baseless accusation, it is based on an objective measurement, a null test which demonstrated a null of only (very roughly) 20dB.
1a. *Absolutely it is, in fact it's arguably the most important part of the scientific method!*! The scientific method absolutely requires the very close scrutiny of ALL aspects of a test/experiment by others: The test material,  the test methodology, the analysis of the resultant data and the asserted conclusions from those results. In fact, the scientific method REQUIRES at least two steps in this process: Firstly, peer review and then replication of the results by others repeating the test/experiment. Faults/Errors are frequently found during this process and depending on their seriousness, the end result can easily be that the scientist/s are personally discredited!

2. Clearly you have presented false information. To start with, as already mentioned, there is some error in the down converted file you've used for comparison. Although again, I'm NOT accusing you of deliberately presenting false information.

3. But you haven't "_used standard tools in the manner recommended_"! You have presented a commercial music mix and the commercial standard for creating a 16bit version from a higher bit depth original/master is NOT sox, nor the use of standard (TPDF) dither. The "recommended standard" is the use of noise-shaped dither and there are/have been a number of tools used as "standard" over the last 25 years or so, for example, in order of time period: Sony Super Bit Mapping, Apogee UV22HR, Pow-R, iZotope and various others but not sox. However, the developer community does appear to use sox quite commonly, presumably because they are thoroughly comfortable using a command line interface. My guess (and it is just a guess because like most audio engineers I've never used sox), is that you've made some error in the parameters you've defined, because I've be very surprised if the developer community had missed some bug or deficiency in sox.

4. There are various reasons how/why a difference 16 vs 24 bit can be detected. I've already mentioned an "unreasonable" listening level but there are other possibilities, for example: You are also downsampling and it's possible that the original contains enough >22kHz content to cause audible IMD in your reproduction chain, while of course the 16/44 version would not contain any >22kHz content. There are other possibilities, such as incorrect settings selected in Foobar's the ABX plugin. The problem is that you are claiming "very obvious" audible differences and "very obvious" audible differences require relatively large objective differences. However, digital audio theory does NOT predict such large differences, this is confirmed by actual objective measurements and further confirmed by countless thousands of controlled listening tests over a period of 25 years which demonstrates no audible difference at all and certainly NOT a "very obvious" audible difference. This extensive body of interlocking supported evidence represents well established, accepted science, which your claim contradict! And, whenever such a claim has been made here, it has ALWAYS turned out to be an error in testing methodology and/or materials, sometimes an inadvertent error and sometimes a deliberate fraud.

5. There two rather obvious problems/falsehoods with this assertion: Firstly, you are NOT "calling into question our beliefs", you are contradicting a very well established, accepted body of evidence/science. And secondly, it's a basic axiom of science that "_An extraordinary claim requires extraordinary evidence_". However, you have not presented extraordinary evidence. You have attempted to provide evidence that is more reliable than the sighted tests typically presented by audiophiles, however, there are serious questions regarding your two comparison files and your methodology, so you have not yet reached the bar required even for reliable evidence, let alone extraordinary evidence! Until you do, extreme scepticism is the ONLY logical response from those who "adhere to science", which is pretty much the opposite of "silly"!!



KeithPhantom said:


> [1] So this is the reason I can hear the noise floor of many recordings, I know it's this because all songs in an album have a particular hiss, and if I start listening to another one, the hiss is different.
> [1a] It has being annoying to say the least.
> [2] I listen at around 80 to 90 dB SPL with headphones.



1. But that's observational evidence that you are NOT hearing the digital noise floor (dither)! This is because dither does NOT change, it's of constant level (equivalent to about the LSB) and constant frequency/spectral distribution (pure white noise). In other words, if you were hearing the digital noise floor (dither), then the hiss would NOT be different. However, both the acoustic noise floor (the noise floor of the recording venue) and the analogue noise floor on a digital recording are very highly variable both in level and spectral distribution. In fact, the acoustic noise floor varies very audibly even in the SAME recording venue, due to slight differences in relative mic positions and slightly different locations within the recording venue, which presents particular problems/difficulties with TV/film sound. Bare in mind that the noise floor of standard dither is about -92dB, while the recording noise floor (acoustic + analogue noise) of commercial music recordings is typically between -40dB and -60dB.
1a. Different songs on an album will have somewhat different playing styles, often different instruments and/or different instrument settings, will have been recorded and mixed somewhat differently and will therefore have a different acoustic (and probably a different analogue) noise floor. There's not much that can be done about this.

2. That's very loud, unless: 
A. You have headphones that do not isolate much and you use them in a relatively high noise floor environment or 
B. If you're listening to material that rarely reaches peak level, only for very short periods and the vast majority of the recording is significantly lower than peak, EG. Certain relatively uncommon pieces of classical music, that have not had any audio compression applied.

Even though it's probably uncomfortably loud for most people, 90dB SPL is still within "reasonable listening levels". With 16bit, using standard dither, the digital noise floor would be about 92dB below peak, so if your peak is 90dBSPL the dither noise would be at about -2dBSPL. Bare in mind that a 100w incandescent light bulb produces about 4 times more noise (~10dBSPL at 1m)! Even with a somewhat higher listening level, world class anechoic chamber conditions, a hypothetical recording with an acoustic + analogue noise floor lower than -92dB and an amp/speakers or HPs that have a combined noise floor lower than -92dB, you'd still be very near the threshold of audibility. However, a somewhat higher listening level (than 90dB) would be very close to an ureasonable listening level in world class anechoic conditions AND, recordings with a low noise floor would/should ALWAYS have noise-shaped dither applied (putting the dither noise at about -120dB), so the dither noise is well below audibility even at the hypothetical/theoretical extremes of "reasonable listening levels" (which of course is why it was invented in the first place)!

Still think you're hearing dither noise on commercial 16bit music recordings reduced from 24bit? 

G


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## gregorio

magicscreen said:


> It is a known story that Philips wanted CD audio to be 14-bit while Sony wanted 16-bit.
> Then would be a scientific topic here: 14 bits are enough, we do not need 16 bits.



There wouldn't be much of a scientific topic here that 14bits is enough, because science can prove that 14 bits are way more than enough. Is SACD (1 bit) many times lower audible fidelity than CD (16 bit)? If not and therefore 1 bit is enough, how would 14 bits not be enough? 

However, while there wouldn't be much of a "scientific topic" here, I'm sure there would be a very protracted discussion with those who don't understand and/or reject the science!

G


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## bigshot (Jul 21, 2020)

Castle, he is lying. I don't have to believe him. And he's going after you now. This guy is trouble and he doesn't belong here. I'm just calling it as I see it. I'm not the one causing the trouble. I'm just the one pointing at it. Correlation does not imply causation.


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## jlawler (Jul 21, 2020)

jlawler said:


> when I compare 24/192 to 16/44, the overall presentation of the sound to my ears is much less harsh.  For example, cymbals, trumpets and saxophones are much more lifelike & pleasing to my ears.  I can abx a difference in these rates with foobar2000.



My claim was that I could abx 24/192 compared to 16/44.  I was challenged by several members to provide proof with foobar signed logs - which I believe I did (albeit 24/96 to 16/44 - proctored testing was not part of the original challenge.  I think it would be really cool to do proctored testing and I am open to it, but to me - this is a secondary test going beyond the original scope.  I think it's a bit unfair to set a goal & then move the goal line once the original requirement is achieved and yet at the same time take the position that the original test was not fulfilled).  I considered redoing the test on a video recording to assure there is no cheating, but I honestly don't think that would eliminate the suspicion of cheating - so I don't think it adds much.  Some people will surely assume there is some cheating off camera even though I commit to you that is not the case.  I have also offered to redo the test with "properly" converted files provided by others.  That offer remains open but unfulfilled.

I certainly do understand requiring extraordinary proof.  If I were in the other position, I would demand that same.  There is a way to state such without impugning someone.  Clearly, the opposite has occurred in some of the communication around this.  It's fine to point out ways a test fails the integrity check.  It's not fine to make baseless accusations and insinuations based on suspicions of cheating (if such an integrity based confrontation is required - it should be based on facts rather than suspicions). 

Regarding the other proposed new test:  I have never tried to abx 24/96 to 16/96 as 16/96 is not a readily available format.  I am also not sure of the purpose of this test since this format is not available to us as consumers.  I would suggest a better test would be 24/44 to 16/44 as both formats are available.  In any case, I do not think the results should reflect negatively on me since this was not my original claim.  If there is interest, I will try it for the benefit of science.


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## bigshot (Jul 21, 2020)

Are you going to get an experienced third party to set up the test for you, or are you going to do it yourself again? The purpose of Foobar ABXes isn't to prove things to other people. They are to find out for yourself. If you want to prove it to others, you need to document your process and have independent oversight of the testing procedures. Then you will want to get someone to try to replicate your results, which might be difficult because what you are claiming has already been pretty well established as impossible with everyone else.

This isn't a test about 16/44.1 vs 24/96. This is a test about your personal hearing ability. If what you claim is true, your hearing is unique. The question is, "Why is your hearing different than everyone who has been tested in the past?" That is what you should be focused on, not proving stuff on internet forums.


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## KeithPhantom

gregorio said:


> There wouldn't be much of a scientific topic here that 14bits is enough, because science can prove that 14 bits are way more than enough. Is SACD (1 bit) many times lower audible fidelity than CD (16 bit)? If not and therefore 1 bit is enough, how would 14 bits not be enough?
> 
> However, while there wouldn't be much of a "scientific topic" here, I'm sure there would be a very protracted discussion with those who don't understand and/or reject the science!
> 
> G


1 bit by itself isn't enough, there is a need of huge oversampling and noise shaping techniques just to lower quantization errors to acceptable levels. It is both inefficient and unnecessary. 14 bits PCM is enough to have a good dynamic range, but not striving for the best would have been a mistake I'm thankful we didn't.


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## bigshot

And although it isn't throwing away bit depth, there is always compressed audio to transparently reduce file size if that is necessary. 16/44.1 works fine for consumers. So does high bitrate lossy for that matter.


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## KeithPhantom (Jul 21, 2020)

gregorio said:


> Still think you're hearing dither noise on commercial 16bit music recordings reduced from 24bit?


When I listen at 90 dB SPL, I do it just to better hear something I might missed (usually recordings errors and such) that at my long-term listening level (75-80 dB SPL) I don't hear them well enough. I supposed it wasn't the dither noise floor, but it was worrying that I could hear that much detail (I only heard hiss from records or transfers of records) from transducers at a relatively decent level. Also, some recordings have some huge high-frequency parasitic tones that annoy me (like Superunknown's Spoon man with a peak of around -10 dBFS at ~20 kHz, as reported by Foobar) that I would like to hear just up to 17 kHz just to get rid of them.


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## bigshot

What you are hearing on specific albums may just be lousy engineering. You can always run it through a dynamic noise filter and get rid of it.


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## KeithPhantom

bigshot said:


> What you are hearing on specific albums may just be lousy engineering. You can always run it through a dynamic noise filter and get rid of it.


This is what I mean by a huge spike at a really high frequency. It sounds eery and ugly, especially in the parts where there is some kind of silence. If this is what old audiophiles want to hear with a "highly resolving system", I feel sorry for their ears


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## bigshot (Jul 21, 2020)

A fella who was here a year or two ago sent me spectrum analysis of a dozen different SACDs that had huge spikes like that. I'm not sure why it's there. Gregorio might know what causes an ultra high frequency spike like that. Maybe it's just to show that there is actually signal up that high, even if the signal is noise. A low pass filter would fix that right up. I thought you were talking about a noise floor, which would be more like white noise.


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## KeithPhantom

bigshot said:


> I thought you were talking about a noise floor, which would be more like white noise.


Previously, yes, I was talking of my whole library and when you turn up the volume, something like hiss starts to be audible. I just showed another issue I have with this specific song. About the previous issue, is hiss and it sounds different depending on the album. As you said, it could be the room noise floor, but I am not really sure. I tested the files in the different sources with my other headphones, and I could hear the same, albeit I had to increase the volume even more, that's when I determined my main source wasn't the issue.


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## gregorio

KeithPhantom said:


> 1 bit by itself isn't enough, there is a need of huge oversampling and noise shaping techniques just to lower quantization errors to acceptable levels.



Any bit depth by itself is not enough, even 24bit is dithered, even the earliest consumer CD players oversampled, all professional ADCs oversample and dither, same with DACs and standard procedure for converting from 24/96 or 24/192 to 16/44 involves oversampling and noise-shaping techniques. In other words, in practice you never hear bit depth "by itself" there is ALWAYS some oversampling and dithering involved, so 1 bit, 32 bit or anything in between is irrelevant, assuming the appropriate oversampling and dithering is used.

G


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## old tech

magicscreen said:


> It is a known story that Philips wanted CD audio to be 14-bit while Sony wanted 16-bit.
> 
> Then would be a scientific topic here: 14 bits are enough, we do not need 16 bits.


The first two CD players on the market were Sony and Phillips (makes sense as they were the inventors of CD) which were 16bit and 14bit respectively. Interestingly, the Phillips 14bit player was the more highly regarded of two sound quality wise.  14bits is enough for high fidelity playback of commercial music but a bit limiting when it comes to recording.  As most digital recordings were 16bit until the 1990s it certainly made sense to standardise CD players to 16bits rather than 14.


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## old tech

jlawler said:


> My claim was that I could abx 24/192 compared to 16/44.  I was challenged by several members to provide proof with foobar signed logs - which I believe I did (albeit 24/96 to 16/44 - proctored testing was not part of the original challenge.  I think it would be really cool to do proctored testing and I am open to it, but to me - this is a secondary test going beyond the original scope.  I think it's a bit unfair to set a goal & then move the goal line once the original requirement is achieved and yet at the same time take the position that the original test was not fulfilled).  I considered redoing the test on a video recording to assure there is no cheating, but I honestly don't think that would eliminate the suspicion of cheating - so I don't think it adds much.  Some people will surely assume there is some cheating off camera even though I commit to you that is not the case.  I have also offered to redo the test with "properly" converted files provided by others.  That offer remains open but unfulfilled.
> 
> I certainly do understand requiring extraordinary proof.  If I were in the other position, I would demand that same.  There is a way to state such without impugning someone.  Clearly, the opposite has occurred in some of the communication around this.  It's fine to point out ways a test fails the integrity check.  It's not fine to make baseless accusations and insinuations based on suspicions of cheating (if such an integrity based confrontation is required - it should be based on facts rather than suspicions).
> 
> Regarding the other proposed new test:  I have never tried to abx 24/96 to 16/96 as 16/96 is not a readily available format.  I am also not sure of the purpose of this test since this format is not available to us as consumers.  I would suggest a better test would be 24/44 to 16/44 as both formats are available.  In any case, I do not think the results should reflect negatively on me since this was not my original claim.  If there is interest, I will try it for the benefit of science.



Although cheating is always a possibility with these things, I don’t believe the majority here are necessarily accusing you of that. Others have pointed out some issues with your test that could or would have produced an unreliable result and to your credit you seem to have taken the feedback onboard.

It is actually very difficult to conduct a proper listening test which provides maximum controls on all variables apart from the bit depth or bit depth and sample rate depending on what you are testing. Many individuals and organisations have expended a large amount of time, human and financial resources in conducting such tests and when done correctly, in all cases have found no audible differences (to humans) between 16 and 24 bits and between 44.1 and higher sample rates. So you can understand the skepticism and questioning when an individual claims that all these controlled tests and the underlying digital audio theory is flawed based on a test they did at home.

Of course that does not mean that you are an exception, as @KeithEmo from Emotiva would always say (he hasn’t posted for a while), you cannot exclude that possibility unless you test every single person in this world. All we can say is that it is unlikely given digital audio theory and known limits of human hearing perception. However, there is not enough rigour in your test to put yourself in that exceptional category.

If you redo your test based on the feedback you have here, cross all the T’s and dot all the I’s, pass peer review and so on there would be many organisations and universities that would be interested in talking with you.


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## Davesrose (Jul 21, 2020)

old tech said:


> The first two CD players on the market were Sony and Phillips (makes sense as they were the inventors of CD) which were 16bit and 14bit respectively. Interestingly, the Phillips 14bit player was the more highly regarded of two sound quality wise.  14bits is enough for high fidelity playback of commercial music but a bit limiting when it comes to recording.  As most digital recordings were 16bit until the 1990s it certainly made sense to standardise CD players to 16bits rather than 14.



Even Philips introduced 4x oversampling for their 14bit system....so oversampling and noise shaping has pretty much always been around with consumer digital audio.  I would think main limits of a digital system are dynamic range (as it is with imaging)....and that does not include boosting the volume of a quiet passage to see if you hear noise.  Our current audio systems can exceed the threshold of comfortable hearing.  We still have a ways to go with video (where the best TVs produce 10 stops of light, and we can accommodate up to 20).


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## old tech

Davesrose said:


> Even Philips introduced 4x oversampling for their 14bit system....so oversampling and noise shaping has pretty much always been around with consumer digital audio.


I know. There is a myth going around that CD players did not have oversampling until the 1990s. I had a 16bit Pioneer CD player that had 4x oversampling back in 1985.  The main improvement in DACs from the every early 1980s was in their filters - particularly the change from analog to digital filters. Not sure if noise shaping was used in digital mastering back then as redbook also specified pre-emphasis rules which were used on many early Japan CDs.


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## bigshot (Jul 22, 2020)

old tech said:


> Of course that does not mean that you are an exception, as KeithEmo from Emotiva would always say, you cannot exclude that possibility unless you test every single person in this world.



Keith Emo took my lossless vs lossy test and did a good job, but still couldn't discern about 256k.



old tech said:


> I know. There is a myth going around that CD players did not have oversampling until the 1990s. I had a 16bit Pioneer CD player that had 4x oversampling back in 1985.



If I'm not mistaken, 1985 was when oversampling was introduced.


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## sander99

The Philips CD100, the very first commercially available Philips CD player from 1983 had oversampling.


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## bigshot

It makes sense that Philips would have it a year or so before Pioneer. Thanks!


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## gregorio

bigshot said:


> [1] Castle, he is lying. I don't have to believe him. And he's going after you now. This guy is trouble and he doesn't belong here. I'm just calling it as I see it.
> [2] Correlation does not imply causation.



1. Bigshot, you're not doing yourself or this subforum any favours. This is the sound science subforum not the "bigshot calls it as he sees it" forum. You could well be right, he may be lying but it's also possible that he's not lying and has made an honest, inadvertent error. Of course, you can personally choose not to believe him but unless/until you have some compelling evidence that he IS lying you CANNOT assert that he is, without being guilty of exactly what you criticise others for!

2. This assertion is true but does it only apply to others and not you? Sure, many who come here making similar claims do so in order to "troll" and commonly lie to achieve that aim but just because jlawler's claims correlate with those who troll/lie does NOT imply that lying is the cause of his claims!!!



jlawler said:


> My claim was that I could abx 24/192 compared to 16/44. I was challenged by several members to provide proof with foobar signed logs ...



But of course, signed logs from foobar is NOT proof, it's not even reliable evidence, it's just *potentially" reliable evidence, the first step towards reliable evidence. 9 times out of 10, requesting an abx with the signed logs is enough to demonstrate that the claim is false. The other 1 time out of 10 it doesn't but then some error in the methodology/materials turns out to be the cause. That's it, that accounts for all 10/10, there isn't an 11/10, an example of abx which supports the claim that does not have some error.

Your original claim "_when I compare 24/192 to 16/44, the overall presentation of the sound to my ears is much less harsh. For example, cymbals, trumpets and saxophones are much more lifelike & pleasing to my ears._" - cannot be true. Digital audio theory (which has been mathematically proven) predicts that the ONLY difference between 16/44 and 24/96 or 24/192 is ultrasonic content (above 20kHz) and a higher digital noise (dither) floor. There is nothing else, there is no distortion in the 3kHz - 8kHz range that could cause a more "harsh" sound or distortion elsewhere in the audible band that could cause cymbals, trumpets and saxophones to sound more lifelike. It's easy to verify this prediction occurs in practice by objectively measuring and comparing a converted file with the original, and such objective verification has been performed countless tens of thousands of times over many years, by both scientists and sound/music engineers. The only rational explanation for what you're experiencing is some perceptual bias or if that's been eliminated, some fault/error with the conversion process or test procedure. Any other explanation has a huge mountain to climb, because it would necessitate invalidating a wealth of objective measurements, a wealth of controlled listening tests and proving some failure in the digital audio sampling theory!



KeithPhantom said:


> [1] When I listen at 90 dB SPL, I do it just to better hear something I might missed (usually recordings errors and such) that at my long-term listening level (75-80 dB SPL) I don't hear them well enough. I supposed it wasn't the dither noise floor, but it was worrying that I could hear that much detail (I only heard hiss from records or transfers of records) from transducers at a relatively decent level.
> [2] Also, some recordings have some huge high-frequency parasitic tones that annoy me (like Superunknown's Spoon man with a peak of around -10 dBFS at ~20 kHz, as reported by Foobar) that I would like to hear just up to 17 kHz just to get rid of them.



1. That's much more reasonable, though even 75-80dBSPL is still very loud with headphones, excepting the two conditions I mentioned previously. It's even very loud with speakers, with most music recordings (popular genres) of the last 25 years or so. In my studio, for me personally, around 68dB is comfortable for most modern popular genre music recordings, I might go as high as 78dB for short periods to examine very fine/quiet details, 90dB is well beyond uncomfortable for me though. These figures might appear quite low to some but then my studio has a noise floor well below that of a typical consumer listening environment.

2. TBH, it's extremely unlikely you can hear that "_huge high-frequency parasitic tone_". It's fairly unlikely even with your listening level set at 90dBSPL (and therefore the parasitic tone would be at 80dBSPL) during otherwise silent (barring the noise floor) parts of the recording, but it's far less likely still, if the parasitic tone is 65-70dBSPL (at your typical  listening level) AND in the presence of a musical signal many times higher in overall energy, as shown in your spectral analysis. Far more likely is that you are being annoyed by IMD much lower in the spectrum, caused by the parasitic tone. Your observation highlights a particular issue for engineers, especially with higher sample rates, it's very difficult to avoid, correct or even identify spurious tones/sounds that we can't hear.



Davesrose said:


> [1] Even Philips introduced 4x oversampling for their 14bit system....so oversampling and noise shaping has pretty much always been around with consumer digital audio.
> [2] I would think main limits of a digital system are dynamic range (as it is with imaging)....



1. Oversampling and dither "yes" but not noise-shaped dither. Noise-shaped dither only came about in the early 1990's, when higher than 16bit recording became available. Even the theory of noise-shaped dither was not published until 1989.

2. I'm not very "au fait" with digital imaging but the dynamic range of digital audio is NOT a main limit, in fact, it couldn't be less of a limit! Digital audio devices capable operating at a bit depth of 32bit (float) started appearing nearly 30 years ago, Neve's Capricorn mixing desk being the earliest example, and 32bit float has a dynamic range of over 1500dB! This of course cannot be realised outside the digital domain because EVERY other limit is massively more restrictive (more "main"), for example: The acoustic limit of a sound pressure wave in air is only 194dBSPL and the dynamic range limits of analogue components (transducers, etc.) are miniscule in comparison and of course we can't forget the dynamic range limitations of human hearing, which are also miniscule in comparison.



old tech said:


> [1] 14bits is enough for high fidelity playback of commercial music but a bit limiting when it comes to recording.
> [1a] As most digital recordings were 16bit until the 1990s it certainly made sense to standardise CD players to 16bits rather than 14.
> [2] Not sure if noise shaping was used in digital mastering back then as redbook also specified pre-emphasis rules which were used on many early Japan CDs.



1. Even 16bit is "_a bit limiting when it comes to recording_"! When we're recording live musicians, we obviously don't know in advance what the peak level is going to be. Typically during a run through or even a sound check, the musicians will play at a significantly lower level than during a performance/take, often deliberately (in order to avoid fatigue or physical damage), sometimes inadvertently (due to the extra adrenalin during a take/performance) and sometimes a bit of both. In other words, when recording musicians, singers or actors' dialogue, we may have to allow 20dB or so of headroom, which effectively reduces our recording to about 12-13bit and therefore the dither noise floor could be as high as about -66dB relative to the peak level of the recording. And, after mixing/processing (compression and other effects), that dither noise floor could end-up 10-20dB higher, which could put it in the realm of the acoustic and analogue noise floors and therefore audibility. That's why 20bit and then 24bit recording was invented, we can have the equivalent of 4bits headroom and the dither noise will never be an issue, even after mixing/processing. HOWEVER, this headroom requirement obviously only applies to recording a signal when it's peak level is highly unpredictable (such a recording a live performer), it does NOT apply to the recording of the mix, the master or the distribution copy, where we know exactly what the peak level will be, therefore it also does not apply to consumers.

2. No it wasn't, because there was no digital mastering back then! All mastering was in the analogue domain, that didn't even start changing until around the mid/late 1990's and it was well into the 2000's before digital mastering was mainstream. The same is true of mixing, although the change and when it became mainstream occurred slightly earlier than with mastering. The only potential exception to this is some classical labels who designed their own bespoke digital mixing systems around the mid 1980's but as far as I'm aware, it was just simple summing and EQ and then mastered in the analogue domain or not mastered. Details of exactly how they worked are a bit sketchy because it was a trade secret at the time. Yamaha were first to market with a digital mixer in 1987 (the DMP7) but that was a fairly simple unit designed for use in live sound to provide cue mixes or keyboard submixes. If memory serves, Sony released Super Bit Mapping (noise-shaped dither) in about 1993 a couple of years or so after 20bit recording first became available. This is when noise-shaping first started being applied but it wasn't really until 24bit recording and digital mixing became common in the late 1990's that noise-shaped dither became standard practice.

G


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## bigshot (Jul 22, 2020)

I never claim that people should take my tests as gospel. I share what I’ve found and I encourage them to check for themselves in good faith. Massive amounts of time are wasted in this forum replying to trolls and disingenuous people. I’d say 80% of the threads here are trolled regularly, including the most important and useful ones. I’m barred from posting in the rest of head-fi on subjects that fall under the topic of sound science. I think people who are out to undermine science should be barred from posting here.

I know exactly where this is going. He’ll keep taking everyone’s comments and say, “Yup, I did that now. Still hear a night and day difference.” The problem here isn’t the controls and variables. It’s a fundamental lack of commitment to integrity. Someone else needs to oversee the test and administer it to him- someone who knows what they’re doing and is on the lookout for tricks. Until then, it’s just talk.

I’ll step back if you want and let you guys go back and forth with him, but I think all know where this is going to end up because we all know that what he is claiming isn’t possible. Either he is doing his test wrong or he is gaming it. Either way, he’s never going to admit it, and we’re never going to get any solid proof of what he is claiming- just post after post after post of derailing an otherwise useful thread. 

If you have a solid strategy for pinning down the truth, have at it. Good luck. I think this whole thing is too wiggly to get anywhere.


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## Hifiearspeakers

bigshot said:


> I never claim that people should take my tests as gospel. I share what I’ve found and I encourage them to check for themselves in good faith. Massive amounts of time are wasted in this forum replying to trolls and disingenuous people. I’d say 80% of the threads here are trolled regularly, including the most important and useful ones. I’m barred from posting in the rest of head-fi on subjects that fall under the topic of sound science. I think people who are out to undermine science should be barred from posting here.
> 
> I know exactly where this is going. He’ll keep taking everyone’s comments and say, “Yup, I did that now. Still hear a night and day difference.” The problem here isn’t the controls and variables. It’s a fundamental lack of commitment to integrity. Someone else needs to oversee the test and administer it to him- someone who knows what they’re doing and is on the lookout for tricks. Until then, it’s just talk.
> 
> ...



You need a vacation.


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## castleofargh

bigshot said:


> I never claim that people should take my tests as gospel. I share what I’ve found and I encourage them to check for themselves in good faith. Massive amounts of time are wasted in this forum replying to trolls and disingenuous people. I’d say 80% of the threads here are trolled regularly, including the most important and useful ones. I’m barred from posting in the rest of head-fi on subjects that fall under the topic of sound science. I think people who are out to undermine science should be barred from posting here.
> 
> I know exactly where this is going. He’ll keep taking everyone’s comments and say, “Yup, I did that now. Still hear a night and day difference.” The problem here isn’t the controls and variables. It’s a fundamental lack of commitment to integrity. Someone else needs to oversee the test and administer it to him- someone who knows what they’re doing and is on the lookout for tricks. Until then, it’s just talk.
> 
> ...


So your gut feeling about his motives is all you need, to know that you're right. And when confronted by some of us about the weakness of that rational, you double and triple down on it without any form of evidence.
where have I seen this behavior before? I can't put my finger on it...


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## Davesrose (Jul 24, 2020)

gregorio said:


> 1. Oversampling and dither "yes" but not noise-shaped dither. Noise-shaped dither only came about in the early 1990's, when higher than 16bit recording became available. Even the theory of noise-shaped dither was not published until 1989.
> 
> 2. I'm not very "au fait" with digital imaging but the dynamic range of digital audio is NOT a main limit, in fact, it couldn't be less of a limit! Digital audio devices capable operating at a bit depth of 32bit (float) started appearing nearly 30 years ago, Neve's Capricorn mixing desk being the earliest example, and 32bit float has a dynamic range of over 1500dB! This of course cannot be realised outside the digital domain because EVERY other limit is massively more restrictive (more "main"), for example: The acoustic limit of a sound pressure wave in air is only 194dBSPL and the dynamic range limits of analogue components (transducers, etc.) are miniscule in comparison and of course we can't forget the dynamic range limitations of human hearing, which are also miniscule in comparison.



Once again you take a post of mine out of context.  I guess I should respond with your odd nomenclature:

1.  Notice I wrote "noise shaping".  If you're going to argue that the Philips DAC didn't implore it, you should contact all the numerous sources that say it did (including Philips themselves).  Evolution of DAC & digital filter ,  The history of the CD - Technology

" The conversion of the digital ‘zeros’ and ‘ones’ into an analogue signal also proved to be a tougher challenge than was at first thought. And it was also very difficult to keep the conversion process linear at lower signal levels, for example between -60 dB and -100 dB.At the introduction of the CD player, every player had a so called 'ladder' D/A converter, followed by a steep analogue filter to remove frequencies above 20 kHz. Philips was the only company to use four times oversampling, with a digital filter, right from its first player. Because four times oversampling means that four samples are taken every 1/44,100th of a second instead of just one, this in combination with first-order noise shaping, which Philips was also the first to apply, allowed 16 bit resolution to be achieved with a 14 bit D/A converter."

2.  Notice I said *system* and not *processing*.  I can generate 32bit per channel image files (used for light simulation and lots of post processing leeway with developing an image).  But the current best *system* for displaying dynamic range is 10 stops (or just over 1000 nits).

Another bit of info: current consumer standards for video can have up to 12bit color space.  Most prominently Dolby Vision, that video that's playing at 12bit gets dynamically "tone mapped" to the display's 10bit space (IE constantly changing contrast range from 12bit's 4096 shades of tone compared to 10bit 1024).  I would think that most people reading this would understand that this is a form of processing, and that the output is still 10bit/stops 1000 nit display.


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## KeithPhantom

gregorio said:


> Far more likely is that you are being annoyed by IMD much lower in the spectrum, caused by the parasitic tone. Your observation highlights a particular issue for engineers, especially with higher sample rates, it's very difficult to avoid, correct or even identify spurious tones/sounds that we can't hear.


I also thought it was IMD, but I compared by ear the pitch of the tone to a pure sine wave around that frequency, they seemed to be really similar.


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## gregorio

Davesrose said:


> Once again you take a post of mine out of context.  I guess I should respond with your odd nomenclature:
> 1.  Notice I wrote "noise shaping".  If you're going to argue that the Philips DAC didn't implore it, you should contact all the numerous sources that say it did (including Philips themselves).  Evolution of DAC & digital filter ,  The history of the CD - Technology
> 2.  Notice I said *system* and not *processing*.
> 2a. I can generate 32bit per channel image files (used for light simulation and lots of post processing leeway with developing an image).  But the current best *system* for displaying dynamic range is 10 stops (or just over 1000 nits).



1. I wasn't aware Philips used noise shaping in their early CD players, although it appears to be very simple 1st order shaping. As far as I'm aware (which obviously isn't as much as I thought!) noise shaping in consumer DACs didn't become widespread until the late 1980's - early 1990's, when 1 bit DS converters became the dominant topology. However, it's hardly surprising I took your post out of context, because it WAS out of context! The context of this thread is obviously 24bit vs 16bit and therefore I took your post in the context of the (high order) noise-shaped dither applied as standard procedure when converting between "high-res" and 16bit.

2. How could I notice you "said *system* and not *processing" *when that's NOT what you said?! What you actually said was: "I would think main limits of a *digital system* are dynamic range ...". An early digital system, one comprising the Neve Capricorn for example, had a dynamic range of 144dB, it's internal processing was 32bit float but it's internal routing and connections to other equipment in the digital system were 24bit ... but 144dB dynamic range is certainly NOT the "main limit", it's the opposite, the least of all the limits! By the mid/late 1990's we had mixing desks, such as the famous Sony Oxford, which maintained 32bit not only for it's processing but also for all it's internal routing/patching and with internal EQ, compression/expansion, limiting, delay and noise gates, it was more of an integrated digital system, although external connections to MTRs and external processors (such as reverbs) was still limited to 24bit (144dB). However, in the 2000's DAWs made it possible to integrate the entire digital system "in the box" and maintain the 32bit float format (~1500dB dynamic range) throughout the digital system, even the resultant audio files. Of course, if we're going to actually hear anything from a digital system it has to be connected to an analogue/acoustic system, which presents a far more massive dynamic range limitation than the digital system.
2a. Maybe the limitation of "_just over 1000 nits_" is an analogue/optical limitation and maybe in the video world it's typical to refer to a digital/analogue/optical system as just a "digital system" because the analogue/optical part isn't considered "a system" but regardless, discussion of video/images is off topic!

G


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## Davesrose (Jul 27, 2020)

gregorio said:


> 1. I wasn't aware Philips used noise shaping in their early CD players, although it appears to be very simple 1st order shaping. As far as I'm aware (which obviously isn't as much as I thought!) noise shaping in consumer DACs didn't become widespread until the late 1980's - early 1990's, when 1 bit DS converters became the dominant topology. However, it's hardly surprising I took your post out of context, because it WAS out of context! The context of this thread is obviously 24bit vs 16bit and therefore I took your post in the context of the (high order) noise-shaped dither applied as standard procedure when converting between "high-res" and 16bit.
> 
> 2. How could I notice you "said *system* and not *processing" *when that's NOT what you said?! What you actually said was: "I would think main limits of a *digital system* are dynamic range ...". An early digital system, one comprising the Neve Capricorn for example, had a dynamic range of 144dB, it's internal processing was 32bit float but it's internal routing and connections to other equipment in the digital system were 24bit ... but 144dB dynamic range is certainly NOT the "main limit", it's the opposite, the least of all the limits! By the mid/late 1990's we had mixing desks, such as the famous Sony Oxford, which maintained 32bit not only for it's processing but also for all it's internal routing/patching and with internal EQ, compression/expansion, limiting, delay and noise gates, it was more of an integrated digital system, although external connections to MTRs and external processors (such as reverbs) was still limited to 24bit (144dB). However, in the 2000's DAWs made it possible to integrate the entire digital system "in the box" and maintain the 32bit float format (~1500dB dynamic range) throughout the digital system, even the resultant audio files. Of course, if we're going to actually hear anything from a digital system it has to be connected to an analogue/acoustic system, which presents a far more massive dynamic range limitation than the digital system.
> 2a. Maybe the limitation of "_just over 1000 nits_" is an analogue/optical limitation and maybe in the video world it's typical to refer to a digital/analogue/optical system as just a "digital system" because the analogue/optical part isn't considered "a system" but regardless, discussion of video/images is off topic!
> ...



So when you try to draw analogies with photography in the very first paragraph in this thread, it's not off topic then???  Granted, throughout this thread, you have shown ignorance of photographic *systems*...in how you don't know standard bit depth layers and how that's relational with dynamic range in a visual system.  How then also in the very response to my use of "digital system", you prove the point about 32bit audio *processing *in relation to the *system* in which output does not meet the same specs (I'm sure other posters in this thread who have been talking about whether they can hear any differences in AQ with their own personal digital stereo systems, know my context of *digital system*)!  My analogy of HDR with imagery was that RAW video can be 12bit (or 14 or 16 bit), and then it has to get reduced to the 10bit space digital TVs can display (0-1023 values, up to 1024 nits). A similar analogy to your own argument of a theoretical dynamic range with a 24bit source vs audio system!!   I notice you tend to not just take posts out of context, but you also conflate terms.  For example, I did look up that the 14bit DAC Philips introduced in 1982 introduced 4x oversampling and noise shaping (apparently used in theirs and Magnovox CD players)...when the context of my response was people thought these early CD players sounded the best (the context of the above posts were about the early 14bit and 16bit DACs, not 16bit vs 24bit): and that you are clearly still going out of context with other noise shaping methods.  The best example of your conflation for me has been a past claim that Dolby's marketing for 3D audio isn't correct because even though it's comprised of X,Y, and a Z axis....its overhead array of speakers isn't overhead (and that in your meaning it would also need positional audio towards your feet).


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## pinnahertz

Philips noise shaping and 4X over sampling was an attempt to utilize 14 bit DAC chips in a world moving to 16 bit DACs.  It only sort of worked, the players had OK SQ, but had so many other issues, particularly in the clunky control concept, that they became legendary, and not always in a good way.  Initially they were welcomed to a niche of audiophiles, then rapidly rejected.  I had one, and learned to hate it really quickly.  Their DAC system did not sound noticeably better than the 16 bit/analog filter machines of the era. This was the era that stimulated better data recovery from the CD, and better error correction/concealment implementation.  Glad it's all over.


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## Hifiearspeakers

pinnahertz said:


> Philips noise shaping and 4X over sampling was an attempt to utilize 14 bit DAC chips in a world moving to 16 bit DACs.  It only sort of worked, the players had OK SQ, but had so many other issues, particularly in the clunky control concept, that they became legendary, and not always in a good way.  Initially they were welcomed to a niche of audiophiles, then rapidly rejected.  I had one, and learned to hate it really quickly.  Their DAC system did not sound noticeably better than the 16 bit/analog filter machines of the era. This was the era that stimulated better data recovery from the CD, and better error correction/concealment implementation.  Glad it's all over.



That’s fine if you didn’t like them, but it wasn’t his point that they necessarily sounded good. His point was that the technology existed and was implemented by them back then, which was earlier than what Gregorio had quoted.


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## Davesrose (Jul 27, 2020)

Hifiearspeakers said:


> That’s fine if you didn’t like them, but it wasn’t his point that they necessarily sounded good. His point was that the technology existed and was implemented by them back then, which was earlier than what Gregorio had quoted.



Pinnahertz wasn't arguing with me.  He specifically did say Philip's use of noise shaping and oversampling wasn't his cup of tea.  I didn't have much personal knowledge.  I was just a few years old, but I remember my parents and grandfather got an early CD player to plug into their stereo...and they sounded great at the time (in comparison to the audio cassette and record players they had).  Really doubt you could tell any audio difference between Sony's 16bit or Philips 14 bit DACs with those systems (and Pinnahertz says more advantages with CD technology were error correction).  My main point was just that there was an apparent difference with processing in even the earliest consumer DACs.


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## Hifiearspeakers

Davesrose said:


> Pinnahertz wasn't arguing with me.  He specifically did say Philip's use of noise shaping and oversampling wasn't his cup of tea.  I didn't have much personal knowledge.  I was just a few years old, but I remember my parents and grandfather got an early CD player to plug into their stereo...and they sounded great at the time (in comparison to the audio cassette and record players they had).  Really doubt you could tell any audio difference between Sony's 16bit or Philips 14 bit DACs with those systems...just that there was an apparent difference with processing in even the earliest consumer DACs.



Ok, fair enough. I just felt like you’ve made some good points in here that have been taken out of context and subsequently, flippantly dismissed. I’m just trying to keep that from continually happening so that the conversation can progress.


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## pinnahertz

Having the technology and utilizing it well don't typically fall at the same point in the time line.


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## davidtriune (Jul 31, 2020)

I used to think audio clipping in a track is due to a lack of dynamic range, that there was not enough bits to record very loud sound. But now I learned that clipping is an artist preference and it has nothing to do with 16/24bit.
16/24 bit determines how quiet a sound goes, not how loud it goes.


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## gregorio

Davesrose said:


> [1] Granted, throughout this thread, you have shown ignorance of photographic *systems*...
> [2] How then also in the very response to my use of "digital system", you prove the point about 32bit audio *processing *in relation to the *system* in which output does not meet the same specs.
> [3] For example, I did look up that the 14bit DAC Philips introduced in 1982 introduced 4x oversampling and noise shaping (apparently used in theirs and Magnovox CD players)...when the context of my response was people thought these early CD players sounded the best (the context of the above posts were about the early 14bit and 16bit DACs, not 16bit vs 24bit): and that you are clearly still going out of context with other noise shaping methods.
> [3a] The best example of your conflation for me has been a past claim that Dolby's marketing for 3D audio isn't correct because even though it's comprised of X,Y, and a Z axis....its overhead array of speakers isn't overhead (and that in your meaning it would also need positional audio towards your feet).



1. Clearly that's false, as I obviously haven't mentioned or discussed "photographic systems" THOUGHOUT this thread! The discussion throughout this thread has been 24bit vs 16bit digital audio and NOT photographic systems. I mentioned half a sentence worth of photography in the OP as a simple analogy: _"It's easy to see in a photograph the difference between a low bit depth image and one with a higher bit depth_" - Nothing controversial or wrong about that statement,  that readers a dozen years ago wouldn't have easily understood.

2. Huh? I explained the exact opposite! Even by 20 years ago, 32bit was NOT ONLY the bit depth of the processing, it was ALSO the bit depth of the patching/routing between all the various processors AND of the output (digital audio files). 32bit was utilized THROUGHOUT the digital audio chain. And even before that point in time, going back nearly 30 years to the earliest use of 32bit in pro-audio equipment, the output was limited to 24bit, which with a dynamic range of 144dB is clearly NOT the "main limit", as you stated: "_I would think main limits of a digital system are dynamic range_"!

3. "Other noise shaping methods", the high order noise-shaped dither required by delta-sigma sampling/conversion and standard procedure for converting 24bit to 16bit, is obviously NOT out of context, given the title of this thread!
3a. Clearly that's nonsense, I certainly did not claim that the overhead array of speakers in a Dolby Atmos system "isn't overhead". Obviously, the soundfield of a Dolby Atmos system is a hemishpere, it effectively only allows half the Z axis.



Hifiearspeakers said:


> His point was that the technology existed and was implemented by them back then, which was earlier than what Gregorio had quoted.



Sort of. It was a very basic form of the technology (only 1st order noise-shaping), the only consumers who encountered it were those few who owned that particular model/models of DAC and, it was used to just recreate the 16bit dynamic range limitation already existant on the CD, not improve on it. But, in the 1990's all consumers encountered noise-shaping, typically 3 applications of high order noise-shaping, but at least one: Noise-shaping during initial recording/sampling, noise-shaping during conversion from 20bit or 24bit to 16bit (manually applied during mastering) and noise-shaping during conversion from digital to analogue. Most decent CD players were capable of the equivalent of about 18bit dynamic range and by the late 1990's many/most CD's had the equivalent of about 20 bits dynamic range. Of course though, this was the DIGITAL dynamic range, the range from digital peak to the digital noise (dither) floor. The actual dynamic range of CDs was limited by factors OTHER than the digital dynamic range and almost never exceeded the equivalent of about 10bits (60dB), these other (non-digital) factors include: The acoustic noise during recording and the analogue noise floor of electric guitar amps/cabs, vintage effects processors, mics, etc.



davidtriune said:


> [1] I used to think audio clipping in a track is due to a lack of dynamic range, that there was not enough bits to record very loud sound.
> [2] But now I learned that clipping is an artist preference and it has nothing to do with 16/24bit.
> [3] 16/24 bit determines how quiet a sound goes, not how loud it goes.



1. Yep, that assumption was incorrect.

2. It can be an artistic preference but there are also other reasons clipping occurs, Inter-Sample Peaks being one example. Still nothing to do with 16/24bit though.

3. Correct. The maximum (loudest) allowed value in digital audio is 0dB (FS) and that value is always the same regardless of whether the file is 8bit, 16bit or 24bit. What changes between the bit depth formats is the minimum (quietest) value: -48dBFS, -96dBFS and -144dBFS respectively. Although, with dither and noise-shaped dither, these minimum values are even lower.

G


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## Davesrose (Aug 2, 2020)

gregorio said:


> 1. Clearly that's false, as I obviously haven't mentioned or discussed "photographic systems" THOUGHOUT this thread! The discussion throughout this thread has been 24bit vs 16bit digital audio and NOT photographic systems. I mentioned half a sentence worth of photography in the OP as a simple analogy: _"It's easy to see in a photograph the difference between a low bit depth image and one with a higher bit depth_" - Nothing controversial or wrong about that statement,  that readers a dozen years ago wouldn't have easily understood.



That first paragraph says "photograph" (that means a whole photographic system that produced a photograph).  In this very thread we have had exchanges on how DR relates to bit depth, and the method of ADC with digital cameras.  You demonstrated to me that you thought bit depth had to do with total channels (IE 32bit means 8 bit per channel RGBA). I found it something else, then, that you said that you have sound mixing experience with cinema, and therefore needed some familiarity with video formats.  Even years past the date of this thread, digital cameras have shot in a RAW format that has a higher bit depth than the 8bpc jpeg image you try to allude to.



gregorio said:


> 2. Huh? I explained the exact opposite! Even by 20 years ago, 32bit was NOT ONLY the bit depth of the processing, it was ALSO the bit depth of the patching/routing between all the various processors AND of the output (digital audio files). 32bit was utilized THROUGHOUT the digital audio chain. And even before that point in time, going back nearly 30 years to the earliest use of 32bit in pro-audio equipment, the output was limited to 24bit, which with a dynamic range of 144dB is clearly NOT the "main limit", as you stated: "_I would think main limits of a digital system are dynamic range_"!



Huh back!!?????  How do you not understand that 32bit master source has to go through conversion for audio playback on a 16bit CD digital system??  You yourself in previous posts admitted that a 32bit processor has DR reduced by significant margins when it gets to transducers (and then also the below quote where you also include realized DR from analog stages)!!  Again you're conflating terms.  Maybe given that you arbitrarily post these absurd outlines that take out other key content (like my point that most people would realize I'm talking about a digital audio system), you only understand your out of context bullet points?



gregorio said:


> 3. "Other noise shaping methods", the high order noise-shaped dither required by delta-sigma sampling/conversion and standard procedure for converting 24bit to 16bit, is obviously NOT out of context, given the title of this thread!
> 
> Sort of. It was a very basic form of the technology (only 1st order noise-shaping), the only consumers who encountered it were those few who owned that particular model/models of DAC and, it was used to just recreate the 16bit dynamic range limitation already existant on the CD, not improve on it. But, in the 1990's all consumers encountered noise-shaping, typically 3 applications of high order noise-shaping, but at least one: Noise-shaping during initial recording/sampling, noise-shaping during conversion from 20bit or 24bit to 16bit (manually applied during mastering) and noise-shaping during conversion from digital to analogue. Most decent CD players were capable of the equivalent of about 18bit dynamic range and by the late 1990's many/most CD's had the equivalent of about 20 bits dynamic range. Of course though, this was the DIGITAL dynamic range, the range from digital peak to the digital noise (dither) floor. The actual dynamic range of CDs was limited by factors OTHER than the digital dynamic range and almost never exceeded the equivalent of about 10bits (60dB), these other (non-digital) factors include: The acoustic noise during recording and the analogue noise floor of electric guitar amps/cabs, vintage effects processors, mics, etc.



This is a great example of how you do respond to posts and take them out of context and conflate with your own terms.  I think in this case, it's just that you simply couldn't admit to being mistaken about the early Philips DAC.  I said that the early DAC had 4x oversampling and "noise shaping" (because the content of the thread then was the first 16bit and 14bit DACs).  You then responded with a disagreement with the statement about "noise shaping".  When presented with the literature, you're still going on ad nauseam of other noise shaping methods (and not accepting written fact about the 14bit DAC).  Just a side note for other folks interested in facts...apart from the CD players that used this DAC, I saw a video from Techmoan that Philips first introduced their 14bit system with a digital tape system (that used VHS cassettes).


----------



## bigshot

Welcome back Gregorio. Keep fighting the good fight.


----------



## Brahmsian

old tech said:


> I understand dithering and the points made by Monty.  The question I have is more around noise shaped dither, particularly moving the energy into the higher frequencies when the band width is limited to 22.05 khz.
> 
> I appreciate that even so, being limited to 22khz that our hearing is less sensitive at higher frequencies doesn't that the extra energy have an effect?  If not, why would noise shapes 8bits require a higher bandwidth than 22khz?  More fundamentally, why is noise shaping beneficial at all for 16bits?


When I got to the noise shaping part of Monty’s demonstration, I immediately thought that I much prefer the more audible midrange dithering noise to the less audible but high frequency one. To me, the former is more soothing. Of course, in both cases, it’s audible only because he brings the gain up for the sake of demonstrating it. Were it normally audible, however, I think I wouldn't find the higher pitch noise the more inoffensive of the two even if it’s harder to hear.


----------



## Lazysnakes

gregorio said:


> It seems to me that there is a lot of misunderstanding regarding what bit depth is and how it works in digital audio. This misunderstanding exists not only in the consumer and audiophile worlds but also in some education establishments and even some professionals. This misunderstanding comes from supposition of how digital audio works rather than how it actually works. It's easy to see in a photograph the difference between a low bit depth image and one with a higher bit depth, so it's logical to suppose that higher bit depths in audio also means better quality. This supposition is further enforced by the fact that the term 'resolution' is often applied to bit depth and obviously more resolution means higher quality. So 24bit is Hi-Rez audio and 24bit contains more data, therefore higher resolution and better quality. All completely logical supposition but I'm afraid this supposition is not entirely in line with the actual facts of how digital audio works. I'll try to explain:
> 
> When recording, an Analogue to Digital Converter (ADC) reads the incoming analogue waveform and measures it so many times a second (1*). In the case of CD there are 44,100 measurements made per second (the sampling frequency). These measurements are stored in the digital domain in the form of computer bits. The more bits we use, the more accurately we can measure the analogue waveform. This is because each bit can only store two values (0 or 1), to get more values we do the same with bits as we do in normal counting. IE. Once we get to 9, we have to add another column (the tens column) and we can keep adding columns add infinitum for 100s, 1000s, 10000s, etc. The exact same is true for bits but because we only have two values per bit (rather than 10) we need more columns, each column (or additional bit) doubles the number of vaules we have available. IE. 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024 .... If these numbers appear a little familiar it is because all computer technology is based on bits so these numbers crop up all over the place. In the case of 16bit we have roughly 65,000 different values available. The problem is that an analogue waveform is constantly varying. No matter how many times a second we measure the waveform or how many bits we use to store the measurement, there are always going to be errors. These errors in quantifying the value of a constantly changing waveform are called quantisation errors. Quantisation errors are bad, they cause distortion in the waveform when we convert back to analogue and listen to it.
> 
> ...



Three issues I have with this, but I'm here to learn not counter. 
#1 I don't understand the DB references  made. 50 db doesn't add to the sound floor of the 144 db range, if
You take 2x 112 db speaker systems and add them together you get about 116 db not 224.
Adding 50 db to 144 gets you about 4 extra or 148db, maybe less. You CAN NOT just add db, it makes no scientific sense. 
#2 another scientific DB problem.  Dynamic range and dynamic reproduction are different measurements.  Say you hear a sound 80 db and another 88 DB, more dynamic range definitely means the difference is greater, so although the recording difference is only 8 db, you will hear at the low volumes most headphones use only a difference of 1db with your 16bit and 1.5 db or so with 24bit.  THIS IS why people tend to turn up their music, because listening to a 122 db recording with only 60db to your ear canals with 16 bit and lower severely stubs the experience.   Furthermore,  an 8bit sound with 122db recording and 122db out doesn't mean an accurate sound of 122db, it will also be much quieter depending on the db caps of the audio, typically 100 or so (less than 10% of what 122db is, you need 10x 100 db to get to 122 roughly) so when a dynamic range increases the subtle differences are exacerbated.  Even to the point of only changing perception of difference in an unrealistic way, it still helps recreate music differently at low volumes. 
#3 I believe you underestimate how much dynamic range effects percieved SQ. To my understanding no matter equipment reproducing it,  the ability to hear variations in music especially listening at lower volumes than real life makes it nessicary. 

#4, another big problem.  Compression.  Compressing especially 10k Hertz plus audio into digital amplifies by the force multiple of the bit when going into bit rate. So a 24 bit streamed by 244bit rate will show less decay than 16bit and so on. This gives a great force multiplayer to anything using a dac. ESPECIALLY  with any CD player. Those all have EQ a d DAC  onboard   which always compress the read because there is no technology that exists which can read discs perfectly without digital reading.
If you had analog disc readers you may be onto something,  but the digital disc readers must lose something by their own compression  causing tremendous dynamic range lose. You're right,  equipment that can produce huge dynamic range doesn't exist, and not have wider source material promotes are larger equipment based problem. The DAC that reads the data compresses audio always, and then readers, and then equalizers (always present in any audio equipment,  the myth that flat EQ exists astounds me), until you lose all kinds of stuff, even with straight 100,000 dollar electrostatic reproduction the amount of loses is unreal.

Perfect examples of why I'm right are: lightning recording (difference between 24bit and 16 is unreal), dynamic range on lightning is ridiculous,  well over 144 db (afterall,  190 DB lightning shockwaves indeed kill people)
Gunshot recordings, gunshots sound  like silenced shots with 16 vs 24bit, and silencers sound like mouse farts. 
And race car recordings. Which never sound like the real thing.

COMPRESSION technology is at fault because even 24bit never comes close to real life, and others dynamic audio equipment which frankly sucks and trying to reproduce those sounds. But when the technology comes my #4th point is that th 24bit is nessicary to minimize recording losses. Refer to point #1 and #2 for why lightning with 24 sounds better than 16 and 16 better than below.


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## bigshot (Aug 22, 2020)

The purpose of commercially recorded music isn't to reproduce reality. It is to present an optimized and organized sound mix that sounds better than reality to human ears.

Just a few specs to put everything in proper context... Most recorded music rarely exceeds 50dB of dynamic range. Beyond that, it's uncomfortable to listen to even in a quiet listening room. The studios and concert halls musicians record in have a noise floor of just under 30dB or so. Your living room where you listen to music is likely between 35 and 40dB. The listening level most people would consider to be as loud as is comfortable to listen to is about 80dB. The threshold of pain (and hearing damage) is 120dB.

So if you take a CD with 96dB of potential dynamic range and boost that potential range up above the noise floor of your living room so you can hear the quietest potential stuff, even if we assume your living room is as quiet as a recording studio, that would put the peaks over the threshold of pain. If you take commercially recorded music and raise it above the room noise floor, you end up with about 80dB peaks, which is as loud as most people want to listen to music. That's why commercially recorded music tops out at about 50dB.

When you remove all of the real world from the equation, too much is never enough. It's easy to point at hyper extremes and use them as benchmarks and come up with crazy results. But the truth is that for the purposes of listening to commercially recorded music in the home, 16 bit is already overkill by a significant measure. The lowly redbook CD outperforms the best analogue studio tape recorders. 24 bit might be useful if you need to bring up quiet elements in a mix. But if that quiet instrument is down in the 40dB range (100dB below peak), you are going to be pulling up a lot of the room tone from the recording venue along with it.  Assuming the band is playing at 140dB (which is highly unlikely) that means that only about 95 to 100 dB of the range is usable. We're back in the realm of CD again. 24 bit is overkill for recording too.

The best sounding album I have ever heard, Donald Fagen's The Nightfly, was recorded and mixed 16/44.1. Potential sound isn't what matters. The quality of the miking and mixing is what matters.


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## Lazysnakes

bigshot said:


> The purpose of commercially recorded music isn't to reproduce reality. It is to present an optimized and organized sound mix that sounds better than reality to human ears.
> 
> Just a few specs to put everything in proper context... Most recorded music rarely exceeds 50dB of dynamic range. Beyond that, it's uncomfortable to listen to even in a quiet listening room. The studios and concert halls musicians record in have a noise floor of just under 30dB or so. Your living room where you listen to music is likely between 35 and 40dB. The listening level most people would consider to be as loud as is comfortable to listen to is about 80dB. The threshold of pain (and hearing damage) is 120dB.
> 
> ...


I fundamentally misunderstand the mathematical reference hear.
Say your room is 50 db. Adding 96 db to 50 db gets you to about 100 db.

You are adding 30/35db to 96 in db, you cannot add that number together it's a physical miscalculation.


----------



## sander99

@Lazysnakes:
#1 You interpret it wrong. G is not saying 50 dB background noise + 96 dB music gives 146 dB sound, he is saying that if you set the volume such that the quitest possible sound on a cd can be heard while there is 50 dB background noise in the room, then the loudest possible sound on a cd will be 96 dB higher than 50 dB, hence 146 dB.


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## bigshot (Aug 22, 2020)

If your dynamic range is 96dB, and the noise floor in your listening room is 30dB, you need to raise the volume above the noise floor to be able to hear the quietest sounds. That means raising 96dB to 126dB which is into the range of hearing damage.

Commercially recorded music has a dynamic range of about 50dB. To raise the quietest details above the 30dB room tone, you raise it to 80dB, which is as loud as most people are comfortable with listening to. That is why music is mixed that way. It is optimized for human ears.

Dynamic range extends downward from the peak level, not upwards. 24 bit has the exact same sound for the top 96dBs. The added details are in the stuff quieter than that. 
Is that clearer?


----------



## sander99

@Lazysnakes:
#2


Lazysnakes said:


> Say you hear a sound 80 db and another 88 DB, more dynamic range definitely means the difference is greater, so although the recording difference is only 8 db, you will hear at the low volumes most headphones use only a difference of 1db with your 16bit and 1.5 db or so with 24bit.


How did you get this idea? It is simply not correct at all. There can be two sounds on a recording with 8 dB level difference. At playback they will also be 8 dB different. No matter whether it is a 16 bit or a 24 bit recording. And no matter what your volume setting is (except that if the volume is too low one or both sounds will not be audible above the background noise, and if the volume - or rather the gain - is too high the amp will clip and/or the transducers will be overloaded).


----------



## Lazysnakes

sander99 said:


> @Lazysnakes:
> #1 You interpret it wrong. G is not saying 50 dB background noise + 96 dB music gives 146 dB sound, he is saying that if you set the volume such that the quitest possible sound on a cd can be heard while there is 50 dB background noise in the room, then the loudest possible sound on a cd will be 96 dB higher than 50 dB, hence 146 dB.


That is inaccurate.  If the quietest sound that can be heard is heard, it will be around the room sound but minus the noise  canceled. So probably only 15 db higher than normal, but even @50, say 4db to 10db can be heard, 4db (at the most ridiculous sensitivity)  will sound like 50db, and 50db plus 96db is about 100db.
Your suggestion is fictitious,  that isn't how DB works, it is algorithmic. If that isn't how it works I'd like to see the evidence,  I may certainly be wrong.


----------



## sander99

You still don't get it. It is not about adding two sounds with an "absolute" dB level, it is about going 96 dB up relative to an "absolute" level of 50 dB.


----------



## Lazysnakes

sander99 said:


> @Lazysnakes:
> #2
> 
> How did you get this idea? It is simply not correct at all. There can be two sounds on a recording with 8 dB level difference. At playback they will also be 8 dB different. No matter whether it is a 16 bit or a 24 bit recording. And no matter what your volume setting is (except that if the volume is too low one or both sounds will not be audible above the background noise, and if the volume - or rather the gain - is too high the amp will clip and/or the transducers will be overloaded).


No, to your ears the difference isn't linear. If a track has sounds recorded at 80 db, then the reproduction can boost it or lose it to 100db or say 50 db.
If played at 50db, then a piece that's 8+ will be 58db but at 58 db the difference compares to 80db plus 4 or 5db or so, so you're absolutely right,  but except for playing the track right were it lands on 80db the difference will be less or more than it should be. This improves between bit bases like 8 and 16bit, of to24 bit. At lower volumes it is heard louder and at higher it is reproducible at a rate that makes more sense. I don't see a world in which 24bit base is not beneficial at all, and audio is so distinctly different than real life putting any restraints on mixing is shackled potential.


----------



## Lazysnakes

sander99 said:


> You still don't get it. It is not about adding two sounds with an "absolute" dB level, it is about going 96 dB up relative to an "absolute" level of 50 dB.


Say the min is 4db, amplified up to 50db. 96db amplified by the same amount is still distantly below the algorithm for multiplication of sound. 
96 times 2 is roughly 110. 110 by to to make 4 times louder is 116, again is 120 that's 8. Again is 123 is 16 and 32x96 is roughly 126db. 4db amplified to 50 is only about 10/20 times louder. That puts 96db at 122db

You would be correct if 1db was in the track and audible but it isn't distinguished from white noise. 1db amplified to 50db would make 96db 50db amplified or roughly 200 times louder.
 This is indeed 146db that's impossible no track could ever have such quiet sounds and it has zero relevance to the topic of using 96db of dynamic range,  no merit that is.

Plus you 1-96db calculation is assuming an awful high 50db ambient noise and listening with transparent headphones. Using IEMs and headphones that reduce noise and basic quiet room elements most people into TOTL equipment will have, room noise to the ears is down to 15 from 50, or 4 from 25.

Putting your volume boost down to nearly nothing. This invalidates 90% of the argument,  but I don't mean to be argumentative it is of my opinion 24bit just is not snake oil it has a substantial benefit.


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## sander99

@Lazysnakes:
It could be that an 8 dB difference is perceived different at different levels, but it still remains an 8 dB difference.

#4 is so full of misconceptions and haziness that I wouldn't know where to start...
But I will pick one thing out:


Lazysnakes said:


> If you had analog disc readers you may be onto something, but the digital disc readers must lose something by their own compression causing tremendous dynamic range lose.


A digital disc reader reads digital data, 0s and 1s, it does this correct (if it works, otherwise it's broken).
The digital data describes the signal. If the data is read correct no changes to the signal or it's dynamic range are made.
If the data is not read correct (and not re-read, corrected or whatever) then the result won't be dynamic compression but random artifacts (like a tic if one or two bits flipped) or total chaos (in case of many errors).


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## sander99

Lazysnakes said:


> This is indeed 146db that's impossible no track could ever have such quiet sounds and it has zero relevance to the topic of using 96db of dynamic range, no merit that is.


It has all the relevance in the world! It is the whole point! Of course no track ever has such quit sound that's why 96 dB of dynamic range is more than enough and 24 bits are competely unnecessary!

@gregorio didn't make this up. And he knows more about audio than the rest of us here together.
You, @Lazysnakes, however are clearly confused about the dB scale. Amplifying a signal by 10 dB means it gets 10 dB louder. Period. It has nothing to do with adding 2 sounds together, in which case indeed you can not just add the levels in dB's together to get the total level. And it is exacly because of this logarithmic scale that the relative level differences between different sounds in the signal stay constant (if expressed in dB's) after amplification or attenuation.


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## Davesrose (Aug 22, 2020)

Lazysnakes said:


> Your suggestion is fictitious,  that isn't how DB works, it is algorithmic. If that isn't how it works I'd like to see the evidence,  I may certainly be wrong.



What you're talking about is loudness and the power requirements of increasing dB in an amplification circuit. With a digital audio file, audio engineers are mixing for loudest undistorted signal to noise floor of the microphone. Un-dithered, a 16bit file can reach 96dB...and with dithering, reaches 120dB.  That's certainly enough as that's reaching immediate pain threshold with healthy hearing.  Your premise that digital audio mixes don't get loud enough for a real life gun or canon could be more for safety/standards in mixing than limitation with the file formats.


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## Lazysnakes

sander99 said:


> It has all the relevance in the world! It is the whole point! Of course no track ever has such quit sound that's why 96 dB of dynamic range is more than enough and 24 bits are competely unnecessary!
> 
> @gregorio didn't make this up. And he knows more about audio than the rest of us here together.
> You, @Lazysnakes, however are clearly confused about the dB scale. Amplifying a signal by 10 dB means it gets 10 dB louder. Period. It has nothing to do with adding 2 sounds together, in which case indeed you can not just add the levels in dB's together to get the total level. And it is exacly because of this logarithmic scale that the relative level differences between different sounds in the signal stay constant (if expressed in dB's) after amplification or attenuation.


I don't understand but I concede


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## Lazysnakes (Aug 24, 2020)

sander99 said:


> @Lazysnakes:
> It could be that an 8 dB difference is perceived different at different levels, but it still remains an 8 dB difference.
> 
> #4 is so full of misconceptions and haziness that I wouldn't know where to start...
> ...



There are no CD readers that don't use decomposition algorithms since 2009. Before then there were uncompressed CDs with about SeVenTy FoRe MiNuTeS of voice on them. Today most are compressed to make it uncopyable. This decompression is nessicary or are there normal CDs still in use with 10 code instead of compressed code?

Why /how do they make money if copying them is so easy?

Also or further, if you have a CD with no compression and the same laser cut depth and size not blue ray then the CD with no compression will have a limit of a couple hours of sound instead of 6-24 hours most 2019 or latter CDs can keep with a modern compression algorithm.


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## Davesrose (Aug 23, 2020)

Lazysnakes said:


> There are no CD readers that don't use decomposition algorithms since 2009. Before then there were uncompressed CDs with about 20 minutes to an hour of voice on them. Today most are compressed to make it uncopyable. This decompression is nessicary or are there normal CDs still in use with 10 code instead of compressed code?
> 
> Why /how do they make money if copying them is so easy?



Where have you been getting your info?  The thing about a disc standard is that they are created before the first players come to market, and they have to stick with the specifications (to stay compatible with all installed players) from there on out.  Redbook CDs did not have any copy protection (data discs could implement tricks).  In 2005, Sony/BMG did try implementing a form of copy protection with 52 artists (an extra data track recognized on Windows computers that acted like a DRM).  They were sued and lost as the software was deemed malware.

Edit to your edit with reference to blu-rays.  You do understand that DVDs and blu-rays are all seperate formats?  The audio CD was a particular standard for audio (and has a run time of up to 74 minutes).  DVD means digital versatile disc: it's a different file format than CD and can be a data system for computers, and also has a specific standard for DVD video players (that used the video compression of the era: MPEG-2).  Blu-ray has a higher density and can hold more data than DVD...with the video disc standards, it also utilizes a newer video compression: MPEG-4 (which is more efficient and can help fit HD lossless audio movies on a disc).  Now we have UHD movies on blu-ray.  Because it's a different standard, you have to have a UHD player to be able to decode its video compression: h.265.


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## bigshot (Aug 23, 2020)

The peak is 0dB. That is because that is the loudest something can be. 0dB at 24 bit is EXACTLY the same as 0dB at 16 bit. The difference is the sound below -96dB.

Dynamic range extends DOWNWARD, not upward. The peak is the peak. Only the stuff below the noise floor of the lower bitrate is different. I know math is hard. I'm lousy at it myself. But to cut to the chase, you would need to incur hearing damage to hear the entire dynamic range of a CD on the system in your living room. It would not be a pleasant experience at all.

16 bit is 16 bit. A CD is just a container for a 16 bit PCM file. It isn't applying compression that wasn't there in the original mix and mastering. And file compression isn't the same thing as dynamic compression at all. You can decrease the data rate of an audio file and create a smaller file size, but that wouldn't be kosher with the redbook standard that CDs have to adhere to. CDs are all 16/44.1.


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## Brahmsian

bigshot said:


> The studios and concert halls musicians record in have a noise floor of just under 30dB or so. Your living room where you listen to music is likely between 35 and 40dB.


Not that it makes much difference to your argument, but I wonder what the noise floor is when you're wearing closed headphones.


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## bigshot

Less, but probably still there. 20dB? Just a guess. Hard to measure that.


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## Lazysnakes (Aug 24, 2020)

hearing damage from peaks with volume tuned for the average of the song to have a high peak doesnt let you hear a higher peak without clipping?

lets then explore the hearing damage opinion. say your hearing a drum-set at 1/100th second, exactly where it should be; 120db or 128db, why doesnt that damage hearing? well because hearing doesn't occur instant damage until you get to nearly 166 (a lightning bolt). even gunshots dont come close to it at over 150db.
drums tend to be the main piece to my music, and it is disappointing nothing in real life sounds close to most music I hear.

USING the LOWEST sound audible in the track is a ridiculous idea to set your volume.

SET your volume so that the loudest recorded sound sounds life like, and you can then have the rest be what it is 20-90db. this will let drums actually hit instead of sounding like, - lolol, mouse farts...

I distinctly disagree with the opinion hearing damage matters to everyone. as a bass head maybe it matters less.

further what of people with shell-shocked ears? what are they gonna _DO, huh? _listen to _your opinion _that music damages hearing instantly at 120 or even 135 db?
my friend cant hear bass less than 110db, period. 8 years in the military. he has a vehicle with nearly 155db, but can still understand everything I say.

this off-premise idea of hearing caps helps me understand the fundamental scientific bass-lessness (puns lol) of the argument:you can actually have too much dynamic range, ONE OF THE ONLY REMAINING rooms of improvement with audio, and a huge problem with speakers that are not dynamic.

I listen to hardtrap a lot, probably the most ridiculous High dynamic range and lowest dynamic range of any music produced. youtube caps this and makes it necessary to find 32 bit files, which always have an obvious immediate difference, but LUBLUBLUBLUBLU YEAH YEAH YEAHYEAYEAYEYHHEYAHEY Causes hearing damage of course it does, but not that much for a 11 minute listen who cares I want it.

its like this forum post about too many lumens: https://www.candlepowerforums.com/v...ng-too-bright-Lumens-race-getting-out-of-hand : IF I want it dont stop meh please.
music example: https://sound cloud.com/szunakachan/colossus
remove space after cloud, and then look up the original under a Japanese server (no-Japanese versions do not exist in 32 bit) should be very _very VERY _loud if done right.
I damaged my hearing and had an tempatic membrane recovery after hearing this over 150db in an auto, but that's okay. it *clips* on soundcloud, AND uh, you know why? it isnt 32 or even 24 bit.



P. S. the downward full caps is the ONLY reason I had to make this. also this is relevant, but not to my original point which I still believe in despite my resignation, this post brings up a different equally important view.


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## bigshot (Aug 24, 2020)

The volume control on your amp doesn’t cause the digital audio file itself to clip. Your posts are beginning to fall below the threshold where they aren’t worth my time to read. I’ve patiently explained it to you. It’s all there. I refuse to have a battle of wits with an unarmed man. -Groucho


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## Lazysnakes

bigshot said:


> The *volume control on your amp doesn’t cause the digital audio file itself to clip*. Your posts are beginning to fall below the threshold where they aren’t worth my time to read. I’ve patiently explained it to you. It’s all there. I refuse to have a battle of wits with an unarmed man. -Groucho


OF course not, the audio itself is clipped.
I refuse to have a battle with someone with the hypocrisy to not even listen to the track, no matter your setup, the audio in that stream will clip because of soundcloud limitations because of lacking dynamic range, anything mixed with the original dynamic range will clip below 24 bit. -Grouchie


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## Lazysnakes

bigshot said:


> The volume co*ntrol on your amp doesn’t *cause the digital audio file itself to clip. Your posts are beginning to fall below the threshold where they aren’t worth my time to read. I’ve patiently explained it to you. It’s all there. I refuse to have a battle of wits with an unarmed man. -Groucho


*Digital clipping*


 

This PCM waveform is clipped between the red lines

In digital signal processing, clipping occurs when the signal is restricted by the range of a chosen representation. For example, in a system using 16-bit signed integers, 32767 is the largest positive value that can be represented. If, during processing, the amplitude of the signal is doubled, sample values of, for instance, 32000 should become 64000, but instead cause an integer overflow and saturate to the maximum, 32767. Clipping is preferable to the alternative in digital systems—wrapping—which occurs if the digital processor is allowed to overflow, ignoring the most significant bits of the magnitude, and sometimes even the sign of the sample value, resulting in gross distortion of the signal.

Directly bouroughed for fair use EDUACATIONAL PURPOUSESSES: https://en.wikipedia.org/wiki/Clipping_(audio)

TheRe yUh gO.
duh doi.

weapons drawn: wIkiPeDiA


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## Lazysnakes

bigshot said:


> 16 bit is 16 bit. A CD is just a container for a 16 bit PCM file. It isn't applying compression that wasn't there in the original mix and mastering. And file compression isn't the same thing as dynamic compression at all. You can decrease the data rate of an audio file and create a smaller file size, but that wouldn't be kosher with the redbook standard that CDs have to adhere to. CDs are all 16/44.1.




THIS BIT I didnt understand at all, thanks for the enlightenment. 
my terminalogy for CD is compressed disc. all DVD/blueray formats are CDs, your semantics are revolutionizing mine, I apreasiate the leshon papa.


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## chef8489

Lazysnakes said:


> THIS BIT I didnt understand at all, thanks for the enlightenment.
> my terminalogy for CD is compressed disc. all DVD/blueray formats are CDs, your semantics are revolutionizing mine, I apreasiate the leshon papa.


Cds are not compressed dics. DVD and bluray are quite different from each other and from cds. 

More you post the more evidence you know very little and are either trolling or talking out your butt.


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## Lazysnakes

Lazysnakes said:


> There are no CD readers that don't use decomposition algorithms since 2009. Before then there were uncompressed CDs with about SeVenTy FoRe MiNuTeS of voice on them. Today most are compressed to make it uncopyable. This decompression is nessicary or are there normal CDs still in use with 10 code instead of compressed code?
> 
> Why /how do they make money if copying them is so easy?
> 
> Also or further, if you have a CD with no compression and the same laser cut depth and size not blue ray then the CD with no compression will have a limit of a couple hours of sound instead of 6-24 hours most 2019 or latter CDs can keep with a modern compression algorithm.




so ya man actually has original 74 minute CDs? wow. arent we living in the past. I have 800,000 songs of space on my 32GB phone SD card. lets not stand on ceremony here. move with the times.


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## Lazysnakes

chef8489 said:


> Cds are not compressed dics. DVD and bluray are quite different from each other and from cds.
> 
> More you post the more evidence you know very little and are either trolling or talking out your butt.


hmm, its a combination of all three, like you, compared to what ther is to know about audio, like you, I know quite little, I am talking out of my buthead, and importantly, you are being trolled back after obviously fruitlessly trolling me for no reason. please STOP ALL CAPPING, STOP YOUR SEMANTICS, and contribute to the forum WITHOUT NAME CALLING. 

   pretty sure that's against the rules. Calling my a troll is name calling, especially when you just openly admitted there's other possibles.

play my sample track, let me know how that clipping sounds, K?


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## chef8489 (Aug 24, 2020)

Lazysnakes said:


> so ya man actually has original 74 minute CDs? wow. arent we living in the past. I have 800,000 songs of space on my 32GB phone SD card. lets not stand on ceremony here. move with the times.


You are full of crap. even at 128kb/s mp3 is around 5mb. You are looking at les than 7k songs for 32gb. A cd in wave format is around 700mb uncompressed audio.
Let's just say you actually converted and compressed 800,000 songs to fit in a 32gb sd card they would sound awful. He'll 128gb sound awful. Maybe that's why you hear clipping.

You giving a link to a compressed file streaming is nothing like a cd, true flac, or true Redbook recording. Your point and example is moot. The quality is not true Redbook 16/44.1k


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## sander99

Lazysnakes said:


> *Digital clipping*
> 
> 
> 
> ...


Indead the more you post the clearer it gets you understand practically nothing of all this stuff (or are deliberately playing the fool).
A digital signal can clip (or wrap if you like) yes. But that has nothing to do with the bit depth.
Your drawing suggests that you think that for example 24 bit audio is converted to 16 bit by removing the 8 most significant bits. That is nonsense. One correct way would be to remove the 8 least significant bits. And maybe apply dither to the result. In general: of course you convert - or produce a new track - in such a way that the result fits in 16 bits without clipping, which is always possible. And as explained by G the only thing that changes is the noise floor, and because the noise floor of a 16 bit signal is not audible except when putting the volume ridiculously high it doesn't matter. (And even if you set it ridiculously high you probably still won't hear the noise floor because your hearing will adapt to the loud levels and become - temporarely if you are lucky - less sensitive).



Lazysnakes said:


> I have 800,000 songs of space on my 32GB phone SD card.


But this thread is about 24 bit versus 16 bit _uncompressed PCM_. It doesn't say anything about massively compressed MP3. Indeed, if that is what you are talking about that is something else entirely. And that could sound bad yeah. So you are barking up the wrong tree here, it is not the fault of 16 bit uncompressed PCM/WAV.
Like @chef8489 already said.


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## Lazysnakes (Aug 24, 2020)

sander99 said:


> Indead the more you post the clearer it gets you understand practically nothing of all this stuff (or are deliberately playing the fool).
> A digital signal can clip (or wrap if you like) yes. But that has nothing to do with the bit depth.
> Your drawing suggests that you think that for example 24 bit audio is converted to 16 bit by removing the 8 most significant bits. That is nonsense. One correct way would be to remove the 8 least significant bits. And maybe apply dither to the result. In general: of course you convert - or produce a new track - in such a way that the result fits in 16 bits without clipping, which is always possible. And as explained by G the only thing that changes is the noise floor, and because the noise floor of a 16 bit signal is not audible except when putting the volume ridiculously high it doesn't matter. (And even if you set it ridiculously high you probably still won't hear the noise floor because your hearing will adapt to the loud levels and become - temporarely if you are lucky - less sensitive).
> 
> ...


hmm. I can see that.
still



with loud enough difference, you can hear the difference in higher frequencies, and with loud enough 1db to 155db changes in music.
if you aint into 155db car audio, your loss bud. the human ability to recognize the difference, and the complete and totally irrelevant idea that the hardrive space it takes up is ridiculous with 1-32 terabyte hard drives, by 2025 to be replaced with 32 terabyte drives shackles with 16bit in hi-res is pointless. might as well have 24 and even 32 bit with over 700khtz, whatever-the-spec.

in otherwords, humans can hear the differnce, probably, but regardless, we should not dismiss 24 or 32bit. it should be embraced as standard or the future of car audio, home and public speaker audio (including wave clipping in theaters with explosions)  and used for self-increasing audio volume changes and sampling for dog training and child training (10-40khtz) etc.


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## chef8489 (Aug 24, 2020)

He has been messaging me and still doesn't get it.
"
I don't understand why you are saying this. I have a 32bit file and a 16bit file, (no 24) the 32 bit sounds louder without distorting, period.
the 16 distantly lacks the punch.
you could master it and then rack the volume up higher, but then the intro will sound way too loud."

"favorite.
with loud enough difference, you can hear the difference in higher frequencies, and with loud enough 1db to 155db changes in music.
if you aint into 155db car audio, your loss bud. the human ability to recognize the difference, and the complete and totally irrelevant idea that the hardrive space it takes up is ridiculous with 1-32 terrabyte hardrives, by 2025 to be replaced with 32 terabyte drives shackles with 16bit in hi-res is pointless. might as well have 24 and even 32 bit with over 700khtz, whatever-the-spec."


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## sander99

@Lazysnakes:
That video you posted is exactly the kind of marketing bull that we like to warn people for here in the Sound Science forum. Like someone before me nicely put it: MQA is about unfolding dollars from your wallet.


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## bigshot (Aug 24, 2020)

I'm sorry, I've had my allotment of Dunning Kruger for this month. Have a nice day, troll. Same old tricks I see.


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## castleofargh

@Lazysnakes 
I have tried to understand your posts and failed. Your reasoning often doesn't seem to make sense based on my understanding of digital audio and decibel. 
You need to make an effort so that we can have a chance to follow your explanation. but my best guess so far is that you have some profound misconceptions about how things really work.
When you have an amplitudes encoded in PCM, if you have 3bits, 111 will be 0dB(the loudest amplitude you can record). all the other permutations will be for lower amplitude values. Now if you have 8 bit PCM, 11111111 will be 0dB. With 8bit we have many more permutations possible, allowing to encode much quieter signals. But the maximum any given bit depth allows is always referred to as 0dB! Then the DAC might turn that 0dB amplitude into 2volt, or maybe 2,5volt, or maybe closer to one volt for portable devices. There is no rule about that. 
0dB is only relevant as being the highest amplitude in digital, and it's an important reference for the sound engineers who want the music to be recorded at less than 0dB(to avoid clipping). and it's important for us if we use some EQ or other DSP to boost a signal already close to 0dB. That's when clipping would occur and it would be bad. Which is why most EQs have a global gain slider to keep our EQ from actually boosting the signal in the digital domain, as it's not something we can do. We don't have bit permutations to go above a line of ones AKA 0dB.

It's fair to consider that bit depth defines how quiet the quantization noise will be(assuming no dither). And really it doesn't do much else. Bit depth most certainly does not decide how loud a signal will be, as again, no matter how many bits you use, the maximum will still be the 0dB digital reference. It can seem arbitrary but that's how digital audio is handled.


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## sander99

castleofargh said:


> if you have 3bits, 111 will be 0dB ...  if you have 8 bit PCM, 11111111 will be 0dB.


The core of your story is correct. 0 dB is the max value.

However to be completely correct:
For PCM the samples are coded in two's complement, see link below. In that coding scheme 011 and 01111111 would be the respective maximal positive values,
and 100 and 10000000 would be the minimal (or max negative) values.
Actually 100 (-4) and 1000000 (-128) are 1 larger in absolute value than 011 (-3) and 0111111 (-127), so I assume they would normally never be used in a 3 and 8 bit digital signal respectively.

https://en.wikipedia.org/wiki/Two's_complement


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## castleofargh

sander99 said:


> The core of your story is correct. 0 dB is the max value.
> 
> However to be completely correct:
> For PCM the samples are coded in two's complement, see link below. In that coding scheme 011 and 01111111 would be the respective maximal positive values,
> ...


I admit that I was thinking about physical R2R gates in a DAC, because it's the easiest way to model(at least in my mind) the passage from binary(switches) to actual voltage amplitude. 
Never hesitate to point out when I'm full of crap, it benefits me too.


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## bigshot

I guess when we respond, it gives them attention. Their name is mentioned. They get email notification that someone has replied. They see multiple people take time out of their day to explain the same idea in a range of ways from simple and direct to complicated and full of footnotes and numbers. It isn’t to their benefit to understand because the second they acknowledge it, the attention stops. The internet brings out interesting behavior in people.


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## old tech

I think part of @Lazysnakes issue is that he references that pseudoscience Hans beek you tube videos as a source.


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## Lazysnakes (Aug 24, 2020)

I don't see a reason, if even humanly indistinguishable,  to limit audio from 16bit instead of hi res files. With terabyte and exabytes of storage there is no reason to waste time mixing and tuning, when you can simply store thousands of 700mb 5 minute songs with little impact on your drive.

In a previous edit I made a serious mistake. There are micro sd cards right now eith 500GB of storage. This is physical storage.
You can then use arithmetic or algorithm coding to compress files further a d depending on the genius that invents the code, this can be quite substantial. Lossless compression isnt the same as a file that is readable quickly,, but as it continues to advance and distantly is more advanced the the oldness of this forum O.P. post I'd say handicapping your files if you produce them yourself, not necessarily buying 24bit, but production should still be 24 bit depth to 32 bit depth.

Compression using 2025 algorithm coding or even 2020 vs the O.P. date, should be enough to store millions of songs or thousands of songs 50MB to 125MB in length or even 32 bit depth or about 500 MB in physical length on mobile SD cards with 500 GB of storage or even just 32MB. This storage power makes any argument for 24 bit depth being pointless and limiting your library invalid.

And I certainly don't understand why one would say the files are damaging to audio quality.


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## bigshot

This is going in circles. Here is an article that goes into great detail about how 16/44.1 is all you need to accurately reproduce everything human ears can hear.
https://web.archive.org/web/20200426202431/https://people.xiph.org/~xiphmont/demo/neil-young.html


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## dazzerfong

Lazysnakes said:


> All micro sd cards are limited to computer chip space of only about 32kbytes. They have no physical storage beyond that, your 32gb sd card can store songs compressed,  which can then be read later.
> 
> I don't see a reason, if even humanly indistinguishable,  to limit audio from 16bit instead of hi res files. With terabyte and exabytes of storage there is no reason to waste time mixing and tuning, when you can simply store thousands of 700mb 5 minute songs with little impact on your drive.



I have no idea why you pulled up allocation unit size - that has zero bearing on the discussion here. It also isn't limited to 32 KB: it's adjustable when you format it. You select a size depending on if you have many small files or some big files. Also, compression isn't what's making storage a non-issue: there's a limit to how much you can compress (research entropy) and SD cards just keep getting bigger. 

No-one's arguing against keeping large files. They're just arguing for the audibility of it over a standard Redbook file.

In other words, sure it doesn't really harm it, but you can't hear the difference anyway. So while on my NAS I have a 24-bit 192 kHz HDTracks version, on my DAP which has 500GB I'm gonna scale it down to 320 kbps so I can have 20,000 songs on it.


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## Lazysnakes

dazzerfong said:


> I have no idea why you pulled up allocation unit size - that has zero bearing on the discussion here. It also isn't limited to 32 KB: it's adjustable when you format it. You select a size depending on if you have many small files or some big files. Also, compression isn't what's making storage a non-issue: there's a limit to how much you can compress (research entropy) and SD cards just keep getting bigger.
> 
> No-one's arguing against keeping large files. They're just arguing for the audibility of it over a standard Redbook file.
> 
> In other words, sure it doesn't really harm it, but you can't hear the difference anyway. So while on my NAS I have a 24-bit 192 kHz HDTracks version, on my DAP which has 500GB I'm gonna scale it down to 320 kbps so I can have 20,000 songs on it.



Why? It's like arguing that lossy compression is okay or a good thing even if-even then, 99% of audiophiles can still hear no difference? 
Why stop there? Why not cap your highs to your hearing limit? If your 35 just cap your songs to 16khtz?

It does not go along with what the same people argue for lossless files.

And I standby the idea, although I concede the argument,  some people can still use the difference in signal.


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## dazzerfong (Aug 24, 2020)

Lazysnakes said:


> Why? It's like arguing that lossy compression is okay or a good thing even if-even then, 99% of audiophiles can still hear no difference?
> Why stop there? Why not cap your highs to your hearing limit? If your 35 just cap your songs to 16khtz?
> 
> It does not go along with what the same people argue for lossless files.
> ...



Lossy compression is a good thing for convenience. The folks who invented the principles of lossy compression were geniuses.  What's achieved now with lossless codecs is a good compromise between portability and hitting the threshold of audibility.

What you're using now is called the slippery slope fallacy. Don't. The reason that's a bad idea is that it means that you'll have to re-encode all your files as your hearing ages.

I personally vouch for lossless files because I'm a data hoarder, but day to day I don't use said lossless files. FLAC doesn't universally support replay gain across all my devices whereas MP3 gain works no matter what.

Difference in signal between 16 bit and 24 bit? You literally cannot. It's like saying what's the difference between 1010000000000000 and 101000000000000000000000. Answer is 8 zeroes that were never going to be used in the first place anyway.


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## Lazysnakes

dazzerfong said:


> Lossy compression is a good thing for convenience. The folks who invented the principles of lossy compression were geniuses
> 
> ...agreed...
> 
> ...



Is it true that conveying a higher sampling rate like that of a CG sample from 500khtz is possible WITHOUT a bit depth of 24bit?
My understanding is that there is still information in those zeros. And to some tracks and people it is applicable. 

The slippery slope fallacy absorbs your comment. Nothing your saying refutes the idea capping audio or reducing size fro. A bit depth is a bad thing generically  speaking. Especially in recording.
In 3085 when we do have the technology to expand our brains to hear music in 80khtz range or what ever, it would be nice to have the content lot limited by those who claim a visible difference is pure snake oil or is not useful in some way.

Unless I'm wrong about sampling and smoothness of the actual sound is achievable with a different technique.


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## Lazysnakes

"So, 24bit does add more 'resolution' compared to 16bit but this added resolution doesn't mean higher quality, it just means we can encode a larger dynamic range. This is the misunderstanding made by many. There are no extra magical properties, nothing which the science does not understand or cannot measure. The only difference between 16bit and 24bit is 48dB of dynamic range (8bits x 6dB = 48dB) and nothing else. This is not a question for interpretation or opinion, it is the provable, undisputed logical mathematics which underpins the very existence of digital audio."

From O.P. original forum comment.

This bit I have a problem with. If you want to limit dynamic range do it to your own music. Using the median sound to maximum sound loudness difference,  ignoring the lowest sound which no one wants anyway (except for tou jazz-class jazzercise fans that want to hear clanking of both tables and covid coughs in the background,  that's not my point) the larger bit depth, the more difference here or as is that wrong.


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## bigshot (Aug 24, 2020)

With modern codecs like AAC and MP3 LAME, at a certain data rate, lossy files are audibly transparent to 100% of audiophiles. It may theoretically possible, but I've never heard of anyone identifying a difference between AAC 320 VBR, 16/44.1 or 24/96 or above. They all reproduce everything we can hear. For the purposes of listening to music in the home, high data rate lossy files can sound just as good as lossless. All of the benefits of 24 bit over 16 bit are beyond the range of human hearing.

But don't take my word for it. Set up a simple test and see for yourself. The people here can tell you how to do that if you are interested.


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## sander99

Lazysnakes said:


> This bit I have a problem with. If you want to limit dynamic range do it to your own music. Using the median sound to maximum sound loudness difference, ignoring the lowest sound which no one wants anyway (except for tou jazz-class jazzercise fans that want to hear clanking of both tables and covid coughs in the background, that's not my point) the larger bit depth, the more difference here or as is that wrong.


I have a feeling here you confuse the actual dynamic range of the music (or whatever content) on the recording with the dynamic range of the recording format. As if dynamic compression would occur when converting 24 bits to 16 bits (without additional processing): that is not true. The level differences between different sounds in the content remain exactly the same. Only if the actual dynamic range of the content were too large to fit in 16 bits then the softest parts would drop below the noise floor.


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## dazzerfong (Aug 25, 2020)

Lazysnakes said:


> This bit I have a problem with. If you want to limit dynamic range do it to your own music. Using the median sound to maximum sound loudness difference,  ignoring the lowest sound which no one wants anyway (except for tou jazz-class jazzercise fans that want to hear clanking of both tables and covid coughs in the background,  that's not my point) the larger bit depth, the more difference here or as is that wrong.



Unless you're trying to reproduce a cannon shot (1812 comes to mind), music never goes that loud. The way bits work, it's a relative system: if your music ever gets to -60 dB, 16 bit or 24 bits would sound exactly the same. It's only if the quietest and loudest sound is more than 96 dB that you'll need more than 24 bits.

The way you're thinking about it is like 'steps' that, with more dB, you'll get a more 'accurate' reproduction of the original sound. Probably something like this sith the number of 'steps':






Doesn't work that way in reality. Like, yes, if you plot it it will look less 'jagged'. But that 'jaggedness' represents a signal that is so quiet you can't hear it.


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## sander99

dazzerfong said:


> if you plot it it will look less 'jagged'. But that 'jaggedness' represents a signal that is so quiet you can't hear it.


More than that: The 'jaggedness' or steps are not even really there in the first place. The digital samples give a number of points and the reconstruction filter creates a fluent curved line through those points. (And the deviation of that reconstructed signal from the original sampled signal - that is the noise - will be a little more with less bits, but with 16 bits still quiet enough.)


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## dazzerfong

sander99 said:


> More than that: The 'jaggedness' or steps are not even really there in the first place. The digital samples give a number of points and the reconstruction filter creates a fluent curved line through those points. (And the deviation of that reconstructed signal from the original sampled signal - that is the noise - will be a little more with less bits, but with 16 bits still quiet enough.)



Yep - that's why those stairstep plots are incredibly misleading. Those inaccuracies manifest as noise, and if you apply dithering, it's something that's even less of a problem.


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## KeithPhantom

dazzerfong said:


> Yep - that's why those stairstep plots are incredibly misleading


Sample and hold. Samples aren't stairsteps at all, they are *points* in the curve representing the signal (+ quantization errors). Oversampling takes care of that easily.


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## 71 dB

old tech said:


> I think part of @Lazysnakes issue is that he references that pseudoscience Hans beek you tube videos as a source.



Hans Beekhuyzen's job is to sell snakeoil. It's kind of sad these guys on the Youtube sound so convincing to those who don't undertand and know digital audio well.


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## Lazysnakes

dazzerfong said:


> Unless you're trying to reproduce a cannon shot (1812 comes to mind), music never goes that loud. The way bits work, it's a relative system: if your music ever gets to -60 dB, 16 bit or 24 bits would sound exactly the same. It's only if the quietest and loudest sound is more than 96 dB that you'll need more than 24 bits.
> 
> The way you're thinking about it is like 'steps' that, with more dB, you'll get a more 'accurate' reproduction of the original sound. Probably something like this sith the number of 'steps':
> 
> ...



Wouldn't you have to change the mixing and thus the sound ratios? Why does 8 bit sound so much worse than 16 bit if the recording is in 16 bit and gets compressed,  and why 
are go u totally and completely backwards when it comes to that. Because the difference between 8 bit and 16 bit is NOT in the quietest sound, it is the median to loudest values that have the most relevant change, I'm arguing the same goes for 24 bit depth.

Furthermore cannons are well over 150db. And I'd love to be able to actually here one so the point still stands.

The record for human audio is 171db in an automobile recorded early this year, 2020 with a custom setup. In order to achieve the record a human must withstand the audio present and he did.


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## 71 dB

Lazysnakes said:


> This bit I have a problem with. If you want to limit dynamic range do it to your own music. Using the median sound to maximum sound loudness difference,  ignoring the lowest sound which no one wants anyway (except for tou jazz-class jazzercise fans that want to hear clanking of both tables and covid coughs in the background,  that's not my point) the larger bit depth, the more difference here or as is that wrong.



When my listening room is VERY quiet the background noise level is maybe 30 dB. If I listen to music quite loud, the peaks go to 100 dB and if I go crazy the peak go to 110 dB. This extreme situation "needs" technically speaking 80 dB of dynamic range (about 13 bits), except if I listen to music so loud the peaks go to 110 dB, my hearing threshold raises temporarily and I won't hear 30 dB sounds, not even close! So the "needed" dynamic range is even less than 80 dB. Vinyl nuts never complain about the limitations of dynamic range with vinyl when it's about 60 dB (10 bits) at best. 60 dB of dynamic range starts to be enough in consumer audio. Since you want some dynamic headroom (say 12 dB or 2 bits worth) for the highest short peaks and also some margin (say 6 dB) in the least significant bit, 10+2+1 bits = 13 bits is what you "need" in consumer audio. So, 16 bit digital audio is OVERKILL by 3 bits or so. Overkill is overkill. Going from 3 bits of overkill to 11 bits of overkill is just overkill with more bits. There are no practical benefits.


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## Davesrose (Aug 29, 2020)

Lazysnakes said:


> Wouldn't you have to change the mixing and thus the sound ratios? Why does 8 bit sound so much worse than 16 bit if the recording is in 16 bit and gets compressed,  and why
> are go u totally and completely backwards when it comes to that. Because the difference between 8 bit and 16 bit is NOT in the quietest sound, it is the median to loudest values that have the most relevant change, I'm arguing the same goes for 24 bit depth.
> 
> Furthermore cannons are well over 150db. And I'd love to be able to actually here one so the point still stands.
> ...



The subject was if 16bits is enough for reproducing all ranges of human hearing.  As has been addressed, with dithering, 16bits allows 120dB DR (with peak being up to 0dB and noise floor being negative value).

When it comes to the loudness of a cannon or gun, you do realize it diminishes the further you stand from it?  If you want a new standard in sound reproduction to produce levels that are harmful after very short intervals....good luck with your one man crusade.


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## 71 dB

Lazysnakes said:


> Wouldn't you have to change the mixing and thus the sound ratios? Why does 8 bit sound so much worse than 16 bit if the recording is in 16 bit and gets compressed,  and why
> are go u totally and completely backwards when it comes to that. Because the difference between 8 bit and 16 bit is NOT in the quietest sound, it is the median to loudest values that have the most relevant change, I'm arguing the same goes for 24 bit depth.



Going from 16 bits to 8 bits truncates the last 8 bits. What this does it raises the noise floor by 48 dB. It also introduces distortion if dither noise is not used.
You mix data compression with volume compression.


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## Lazysnakes

71 dB said:


> Hans Beekhuyzen's job is to sell snakeoil. It's kind of sad these guys on the Youtube sound so convincing to those who don't undertand and know digital audio well.



Mmm I may not, but I have no interest in believing in something with no evidence,  just physical proof that 24 bit is absolutely better than 16 bit depth, regardless of human abilities to tell the difference. 
Likewise I am not a high-end hi-fi audiophile,  and my interest in pointing out the glaring flaws of the original post is purely scientific. 

I believe the relative factor is still not of your understanding here.  The idea that the change from normal music to the maximum is present in going from 8 to 16bit,
Even there most people never use the range in record broad music, but narrower music interest as well as home and public theater do use difference of this and more. And  on recording that comes from rock and such still have a difference that could be widened with 24 bit.

The use of other techniques is still sound dampening by some degree, and although I wouldn't have this huge expensive push toward things that are not really making a difference for your hi-fi music, I personally don't see such a ridiculous hostility toward larger bit depth technology and not utilizing it to a further degree.

The military and metal detection use sampling above 1gigahertz (1,000,000 hertz) in order to produce accurate measurements of sound waves around 1/10 of that, or 100,000 htz which are used for echo location and for slowing down of the recording.  The latter which could actually help a lot with pitch control and redefinition.

Also more advanced techniques for algorithm coding to shrink the size of such recording after first making it, including lossy compression of dense music that has as you say "more information than nessicary for human hearing " could be used down the road to keep files the same size as lossless 16 bit and have the same quality, with possibly more relevant information.  And even compressed smaller, lossy, and still sound the same because of the information gained from the curves and the algorithmic curve prediction for frequencies of sound being predicted, due to 4 times the recording points. This can not be improved without wider acceptance of 24 bit depth recording and even consumption. 

Although considered irrelevant by some here, no one has provided evidence that the future is not brighter with 24 bit depth, mark my words, 24 bit will be used to noticeably make audio better.  And as of 2021/2020 there is no longer a compelling counter that we shouldn't use it. 

I was messaged by someone that said it themselves:  some recordings are only available in 24 bit depth, there is plenty of storage for the increase, there is less need for dithering and other labor techniques,  less mixing nessicary,  and a broader range for deliberate manipulation of the sounds. 

Cannons and such sounds, bass boosting without using amplification, and public theater explosion sounds can also accept an improvement. I don't understand all of this yet but I will unfollow this forum because I have reduced the responses from the majority helping my understanding and contributing,  to jabs at my reputation and blatant "no you" and "you're wrong" "you knownothing" statements,  which besides being unhelpful and far from the truth are damaging to the spirit of education and science.


----------



## Lazysnakes

KeithPhantom said:


> Sample and hold. Samples aren't stairsteps at all, they are *points* in the curve representing the signal (+ quantization errors). Oversampling takes care of that easily.


Using  higher sampling allows better definition of the curve. With practice,  and a genius interested in music with an immeasurable IQ (none such genius has existed in known history, unfortunately), I suspect it is easier to compress 24 bit to a smaller size than 16 bit, or to be more specific,  196k htz to be compressed to 44k htz with wave forms that make music sound  better because the quantinization errors make lower sampling rates harder to do lossy and lossless compression without losing information.

It is the same with 4k 1080p picture. It actually compresses to a smaller 1080p file than 1080p does, invented recently, 1080p that is recorded at twice the definition (4k) has a cleaner downscale and  a smaller lossless algorithm compression due to the wider range of pixel colors, especially when taking HDR into the mix. A case of compressing a more complete information index into a small size rather than not being able to compress files too well because of the 

(Genius)  earlier who suggested entropy causes a problem.  He was wrong and totally missed my point this is a better way to explain my philosophy on compressing audio.


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## Davesrose

Lazysnakes said:


> It is the same with 4k 1080p picture. It actually compresses to a smaller 1080p file than 1080p does, invented recently, 1080p that is recorded at twice the definition (4k) has a cleaner downscale and  a smaller lossless algorithm compression due to the wider range of pixel colors, especially when taking HDR into the mix. A case of compressing a more complete information index into a small size rather than not being able to compress files too well because of the
> 
> (Genius)  earlier who suggested entropy causes a problem.  He was wrong and totally missed my point this is a better way to explain my philosophy on compressing audio.



This is utterly false.  I was specific about blu-ray and UHD standards in a previous response to you where you didn't understand the difference between an uncompressed audio CD standard vs CD containing compressed mp3s.  It seems you're trying to understand video codecs: formats for compressing video to a smaller file size.  1080P blu-rays are encoded in MPEG-4 H.264.  A good video encoding program will let you specify properties such as 2 pass encoding (where the software first goes through the video to analyze and then finalize a variable bitrate that's high enough to maintain quality, but small enough to save disc space).  If all things were equal with a 1080P resolution video compared to a UHD resolution (and barring frame rate or color depth differences), the 4K video takes up 4 times the data. H.265 was implemented with UHD so that UHD movies could still fit on a blu-ray (with up to 100GB storage).  If you compress a 4K movie with the h.264 codecs common with HD, you still wind up with a much larger file size.  If you encode a 1080P movie in h.265, you wind up with a much smaller file size than UHD.


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## KeithPhantom (Aug 29, 2020)

Lazysnakes said:


> quantinization errors make lower sampling rates harder to do lossy and lossless compression without losing information.
> 
> It is the same with 4k 1080p picture. It actually compresses to a smaller 1080p file than 1080p does, invented recently, 1080p that is recorded at twice the definition (4k) has a cleaner downscale and a smaller lossless algorithm compression due to the wider range of pixel colors, especially when taking HDR into the mix. A case of compressing a more complete information index into a small size rather than not being able to compress files too well because of the


There is a problem with your analogy. More points above the ones needed by Nyquist are not needed if we account for the accepted limitations of the human ear. 16 bit undithered is more than enough since errors and the noise floor are ~ -90 dBFS at worst from the fundamental. In terms of 4K and 1080p, the differences are apparent when screen size is increased, but we can't do that with audio (we can't zoom with audio. I'malso ignorant in video, so correct me if needed). For me, I would worry more about transducers, with the distortion of the best ones averaging -60 to -70 dB occasionally in the best of the cases. Electronic, and especially, distortion caused by the files and encoding methods are way down in the list of things to take care of.


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## Davesrose

KeithPhantom said:


> There is a problem with your analogy. More points above the ones needed by Nyquist are not needed if we account for the accepted limitations of the human ear. 16 bit undithered is more than enough since errors and the noise floor are ~ -90 dBFS at worst from the fundamental. In terms of 4K and 1080p, the differences are apparent when screen size is increased, but we can't do that with audio (we can't zoom with audio. I'malso ignorant in video, so correct me if needed). For me, I would worry more about transducers, with the distortion of the best ones averaging -60 to -70 dB occasionally in the best of the cases. Electronic, and especially, distortion caused by the files and encoding methods are way down in the list of things to take care of.



I think we should also be specific about audio compression (compressing dynamic range) vs @Lazysnakes miss-use of an analogy with 4K video compression: which is about file compression.


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## KeithPhantom

Davesrose said:


> I think we should also be specific about audio compression (compressing dynamic range) vs @Lazysnakes miss-use of an analogy with 4K video compression: which is about file compression.


I would like to get more into video, do you have anything like sound science but for video? It could a forum, but a YouTube channel/video would be perfect.


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## Davesrose (Aug 29, 2020)

KeithPhantom said:


> I would like to get more into video, do you have anything like sound science but for video? It could a forum, but a YouTube channel/video would be perfect.



When it comes to consumer products, I think AVS is pretty good: they do have forums on firmware updates and settings for TVs, receivers, etc.  Rtings is also a great site for TV reviews.  I'm involved with 3D graphics and video editing....so I'm used more with sites for professional topics, training!  There is pretty good info out there for resolution benefits vs certain viewing distances. Photography has also been a serious hobby with me (starting with the film days).  There are sites like Head-Fi for that (and less biased like headphone reviews, as photography is applied recording).  They do have more threads now for videography...which can be a separate application than cinema specific camera equipment.  There's also YouTube channels devoted to photography news (some get more money with their own beginers guides to photography).

I notice Samsung is really advertising their 8K QLED TVs, but there isn't going to be true 8K film sources for awhile (there have been just a few 8K film restorations, while many 4K...only the most recent big production movies are done in 4K digital intermediates).  Like plasma was rated highly before, the 4K OLEDs still score higher than QLEDs because of clearly defined pixels (QLED is brighter and better for brighter rooms, with OLED being better with blacks and great for rooms you can control lighting).  I also think 4K standards are great for cinephiles as it reaches the realized limits of 35mm film formats (may be last older film you need to double dip with....and some new Dolby Atmos/DTS:X remixes have been nice).


----------



## bfreedma

Davesrose said:


> When it comes to consumer products, I think AVS is pretty good: they do have forums on firmware updates and settings for TVs, receivers, etc.  Rtings is also a great site for TV reviews.  I'm involved with 3D graphics and video editing....so I'm used more with sites for professional topics, training!  There is pretty good info out there for resolution benefits vs certain viewing distances. Photography has also been a serious hobby with me (starting with the film days).  There are sites like Head-Fi for that (and less biased like headphone reviews, as photography is applied recording).  They do have more threads now for videography...which can be a separate application than cinema specific camera equipment.  There's also YouTube channels devoted to photography news (some get more money with their own beginers guides to photography).
> 
> I notice Samsung is really advertising their 8K QLED TVs, but there isn't going to be true 8K film sources for awhile (there have been just a few 8K film restorations, while many 4K...only the most recent big production movies are done in 4K digital intermediates).  Like plasma was rated highly before, the 4K OLEDs still score higher than QLEDs because of clearly defined pixels (QLED is brighter and better for brighter rooms, with OLED being better with blacks and great for rooms you can control lighting).  I also think 4K standards are great for cinephiles as it reaches the realized limits of 35mm film formats (may be last older film you need to double dip with....and some new Dolby Atmos/DTS:X remixes have been nice).




Agree on all points regarding commercial 8k, but there is some limited but very high quality content out there for anyone looking for 8k native demo content.  Check   https://drive.google.com/drive/folders/1TSdV36G_npDtjRJze54GEYpdxpBt7nCK.  (sorry about the OT, but figured some might be interested...)

Before anyone asks, I would not spend more for an 8k TV this year than a 4k set.  The only reason I have an 8k set is that Samsung was having a very hard time moving them late last year and I actually negotiated a better price for a TOTL 8k Samsung than was possible on the 4k equivalent.  The store really wanted to move stock..


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## 71 dB (Aug 30, 2020)

Lazysnakes said:


> 1. Mmm I may not, but I have no interest in believing in something with no evidence,  just physical proof that 24 bit is absolutely better than 16 bit depth, regardless of human abilities to tell the difference.
> 
> 2. Likewise I am not a high-end hi-fi audiophile,  and my interest in pointing out the glaring flaws of the original post is purely scientific.
> 
> ...



1. This is not about what is "absolutely better." Infinite sampling rate together infinite bit depth ( ∞ Hz / ∞ bits) is "absolutely" best, but any serious person agrees what counts here is human hearing and what we can hear. What is good enough? What is not?

2. The facts of digital audio don't care about whether you or anyone else is a high-end hi-fi audiophile.

3. What is "normal music"? What is "maximum music"? What are "narrow" and "broad" music? 8 bits is not considered enough for high quality consumer audio. I don't consider it either. It's 5 bits less than what I consider adequate for transparent consumer audio. 8 bit is low quality digital audio. Going from 8 bits to 16 bits means going to the other side of the threshold line. Once you are on the other side, it doesn't matter how far you go.

4. This is not hostility toward larger bit depth technology. This is about calling out the snakeoil bs about 24 bit providing higher fidelity than 16 bit. In consumer audio this is not the case. In music production things are different, there are additional things to consider so in music productio 24 bit is definitely better than 16 bit, but ones the music is produced and scaled optimately adding 16 bit dither and dropping the last 8 bits (most of them probably just noise from various sources) out means zero loss of perceptual fidelity. *In other words 24 bit technology in studios allows us to take the full potential out of 16 bit technology. *

5. Well, I am not detecting enemy tanks when I listen to music. I am not an expert on "metal detection." If 1 MHz (megahertz, not gigahertz I believe) sampling frequency is needed then that's what is used.

6. Lossy coding is whole another issue. This is about non-data-compressed music. This topic explodes when we start to talk about lossy compression.

7. So music sounds better to you when there's stuff you can't hear? Have you ever listened to the difference of 16 and 24 bit audio (same master?) It's noise. Insanely quiet noise. You can't hear it at any rational listening level and even if you could, the music masks it.

8. 24 bit serves it's purpose. So does 16 bit. They sound the same (if from same master). Use 24 bit all you want, but when you say 16 bit gives less perceptual fidelity you are wrong. Is 24 bit the magic number for you? No need for more? 32 bit? 64 bit? 1348457 bit? How far do you want to go?

9. Considering human hearing and how loud sounds must be mixed to make any sense 16 bit is enough. Public theaters are noisy places. Heck, the audience might not notice if the theater sound was (dithered) 8 bit sound because the noise from pop corn eaters would mask the dither noise.


----------



## castleofargh

Lazysnakes said:


> Wouldn't you have to change the mixing and thus the sound ratios? Why does 8 bit sound so much worse than 16 bit if the recording is in 16 bit and gets compressed,  and why
> are go u totally and completely backwards when it comes to that. Because the difference between 8 bit and 16 bit is NOT in the quietest sound, it is the median to loudest values that have the most relevant change, I'm arguing the same goes for 24 bit depth.
> 
> Furthermore cannons are well over 150db. And I'd love to be able to actually here one so the point still stands.
> ...


Little of what you said makes sense. Again, when you go from one bit depth to a smaller one, what we do is truncation. We get rid of the lower bits. It is not a compression!
As to 8bit sounding "so much worse than 16bit", instead of repeating what I already told you in PM that you didn't understand or didn't believe, here is an actual example to download(3.8Mo flac):
https://www.dropbox.com/s/jyzibehg3bhe4ux/The Vampyre Of Time And Memory .flac?dl=1
I took a sample of the song, saved it to 16/44 and 8/44, then I put both into the same 16/44 track, one after the other. So the container is 16bit but the second sample is a signal that only had 8bit of data when it was added to the track. the missing 8bit for that second part can't magically come back from the bit cemetery, so you get to hear the quantization noise at 8bit  on one of the versions in the file. Without getting whatever issues people may get because their audio app doesn't know how to play 8bit, or whatever.



"does anyone ever get this right?"


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## 71 dB (Aug 30, 2020)

Lazysnakes said:


> 1. Using  higher sampling allows better definition of the curve. With practice,  and a genius interested in music with an immeasurable IQ (none such genius has existed in known history, unfortunately), I suspect it is easier to compress 24 bit to a smaller size than 16 bit, or to be more specific,  196k htz to be compressed to 44k htz with wave forms that make music sound  better because the quantinization errors make lower sampling rates harder to do lossy and lossless compression without losing information.
> 
> 2. It is the same with 4k 1080p picture. It actually compresses to a smaller 1080p file than 1080p does, invented recently, 1080p that is recorded at twice the definition (4k) has a cleaner downscale and  a smaller lossless algorithm compression due to the wider range of pixel colors, especially when taking HDR into the mix. A case of compressing a more complete information index into a small size rather than not being able to compress files too well because of the
> 
> (Genius)  earlier who suggested entropy causes a problem.  He was wrong and totally missed my point this is a better way to explain my philosophy on compressing audio.



1. No no no!! Higher sampling frequency allows HIGHER FREQUENCIES. Properly bandlimited signal are represented (within the dynamic range) with 100 % accuracy. NOTHING can be better definition than 100 %. When the signal is bandlimited (for example all frequencies above 20 kHz are filtered away because humans can't hear above that) and we use the sampling frequency on 44.1 kHz (at least twice the highest frequencies in the signal) the signal can't make any kind of curves BETWEEN the sample points. In fact there is only ONE solution for the signal to have it's curvy shape while going through the sample points. That is the WHOLE point of *sampling theorem*, something you clearly have not studied. Sampling is not about taking samples of the signal to get a somewhat accurate picture of what the signal does. It is about capturing 100 % of the information about what the signal is doing. Because the sample points must be quantized to certain amount of bits, noise gets added to the signal. If that noise is quiet enough so you don't hear it the signal is perceptually 100 % perfect. Higher sampling rates allows music for bats if that's something important to you.

The real thing that makes music sound better is to produce it better. Compose better, play better, mix better, etc. That's what makes better sound, not ridiculous bitrates.

2. In video heavy compression is a must, because without it the raw video feed is MASSIVE. With audio you don't really need data compression. Video compression is different. It hapens spatially and temporarily. Eyes are different from ears. So these comparisons are dangerous.


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## tansand (Aug 30, 2020)

"Because the sample points must be quantized to certain amount of bits, noise gets added to the signal. If that noise is quiet enough so you don't hear it the signal is perceptually 100 % perfect."

Isn't quantization noise just a fancy word for distortion though?


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## danadam (Aug 31, 2020)

tansand said:


> Isn't quantization noise just a fancy word for distortion though?


AFAIK, if quantization was done without dither, producing quantization error correlated to the input signal, this would be called a distortion. When quantization is done with dither, producing quantization error not correlated to the input signal, it is called noise.

The difference is most audible with quiet signals. Listen to the attached files:

"02_8bit_no_dither.flac" sounds distorted when compared to the "01_16bit_orig.flac",
"03_8bit_dither.flac" sounds the same, only with added noise, when compared to the "01_16bit_orig.flac",
"04_8bit_dither_shaped.flac" sounds the same, with only a little noise, when compared to the "01_16bit_orig.flac".
(and why would you ever want to do quantization without dither is anyone's guess)


----------



## old tech

Here's another good video explaining bit depths and the effect on listening.  The demonstration should be enough to convince the most ardent skeptics unless they just don't want to know.

The difference between 8bits and 16bits is noise, the signal is all there, nothing is missing.  With dither, most people wouldn't be able to tell the difference between 8bits and 16bits for most commercial music, let alone 16bits compared with 24bits!


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## old tech

71 dB said:


> 3. What is "normal music"? What is "maximum music"? What are "narrow" and "broad" music? *8 bits is not considered enough for high quality consumer audio.* I don't consider it either. It's 5 bits less than what I consider adequate for transparent consumer audio. 8 bit is low quality digital audio. Going from 8 bits to 16 bits means going to the other side of the threshold line. Once you are on the other side, it doesn't matter how far you go.



High bias cassette tape is equivalent to around 5bits digital and LP records around 10-11bits.  The truth is even 8 bits is enough for most commercial music and for the bulk of listeners.  After all, pre-recorded cassettes were the best selling music format from the late 1970s to the early 1990s.


----------



## tansand

danadam said:


> AFAIK, if quantization was done without dither, producing quantization error correlated to the input signal, this would be called a distortion. When quantization is done with dither, producing quantization error not correlated to the input signal, it is called noise.



Thank you for this, I appreciate it.


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## bigshot

16/44.1 is overkill. The fact that people argue it isn’t nearly enough makes me shake my head. I think it’s the “more is always better theory”.


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## SoundAndMotion

old tech said:


> Here's another good video explaining bit depths and the effect on listening.  The demonstration should be enough to convince the most ardent skeptics unless they just don't want to know.
> 
> The difference between 8bits and 16bits is noise, the signal is all there, nothing is missing.  With dither, most people wouldn't be able to tell the difference between 8bits and 16bits for most commercial music, let alone 16bits compared with 24bits!



I like that video. Many similar videos come from a more engineering/signal theory perspective, but this video is something music lovers can appreciate. If one watches the whole thing and listens to what Mr. Shepherd says, a nice understanding of how bit-depth relates to music recording is available. Nice find.

One can't just pick out bits (lol) of what he says, though. Early on (0:42) he says "if you do it right", which he later (3:52) explains and demonstrates to mean adding appropriate dither. You do say: "With dither, most people ... most.. music," which is true. We agree. If you add noise shaping and oversampling, you can safely replace one of the "most"s with "all".

But the sentence "The difference between 8bits and 16bits is noise, the signal is all there, nothing is missing." is demonstrably false on its own. A passage hovering around -55 to -60dBFS will disappear if recorded at or truncated to 8 bits. So the signal is not all there. Am I being hypercritical or nit-picking? For most readers who'll never face this issue, yes. For the occasional DIYer who floats through here, and may have an Arduino board with 8-bit ADCs and some music project in mind, no, he/she must "do it right".


----------



## castleofargh

danadam said:


> AFAIK, if quantization was done without dither, producing quantization error correlated to the input signal, this would be called a distortion. When quantization is done with dither, producing quantization error not correlated to the input signal, it is called noise.
> 
> The difference is most audible with quiet signals. Listen to the attached files:
> 
> ...


A word of warning about this. It's probably obvious for most who tried the files, but they are recording a very quiet signal that's close to and reaching into the 8bit cut. Which is why the hiss is really loud relatively to the music content even though it's 8bit down, and why the correlated noise(no dither) has so much impact and isn't masked much if at all by the music content.
Otherwise those are very nice examples.


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## 71 dB

old tech said:


> High bias cassette tape is equivalent to around 5bits digital and LP records around 10-11bits.  The truth is even 8 bits is enough for most commercial music and for the bulk of listeners.  After all, pre-recorded cassettes were the best selling music format from the late 1970s to the early 1990s.



I believe C-casettes can do much better than 30 dB of dynamic range when noise reduction is used. "Best selling" doesn't mean "hight quality." Effectively dithered 8 bit might be enough for _some_ music such as commercial music with heavy dynamic compression, but we must use the most demanding music out there, not the easiest.


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## old tech

SoundAndMotion said:


> I like that video. Many similar videos come from a more engineering/signal theory perspective, but this video is something music lovers can appreciate. If one watches the whole thing and listens to what Mr. Shepherd says, a nice understanding of how bit-depth relates to music recording is available. Nice find.
> 
> One can't just pick out bits (lol) of what he says, though. Early on (0:42) he says "if you do it right", which he later (3:52) explains and demonstrates to mean adding appropriate dither. You do say: "With dither, most people ... most.. music," which is true. We agree. If you add noise shaping and oversampling, you can safely replace one of the "most"s with "all".
> 
> *But the sentence "The difference between 8bits and 16bits is noise, the signal is all there, nothing is missing." is demonstrably false on its own. A passage hovering around -55 to -60dBFS will disappear if recorded at or truncated to 8 bits. *So the signal is not all there. Am I being hypercritical or nit-picking? For most readers who'll never face this issue, yes. For the occasional DIYer who floats through here, and may have an Arduino board with 8-bit ADCs and some music project in mind, no, he/she must "do it right".


The signal is all there, even if drowned by noise.  After all, DSD is 1 bit and the entire signal is recovered - albeit at very high sample rates and noise shaping.  In any event, I think we are referring to playback rather than recording.


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## 71 dB

SoundAndMotion said:


> But the sentence "The difference between 8bits and 16bits is noise, the signal is all there, nothing is missing." is demonstrably false on its own. A passage hovering around -55 to -60dBFS will disappear if recorded at or truncated to 8 bits. So the signal is not all there. Am I being hypercritical or nit-picking?



It's not false. The signal is all there, part of it is just masked by the noise. If you have the original 24 bit file from what the 8 bit version was made, you can produce the dither noise by substracting the 24 bit version from the 8 bit version (as done in the video). If you substact this noise (now in 24 bits) from the 8 bit version you get the original 24 bit version. All the signal is there. In music instruments mask each other to same extent, but that doesn't mean signal gets lost. It's all there, just partially masked.


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## old tech

71 dB said:


> I believe C-casettes can do much better than 30 dB of dynamic range when noise reduction is used. "Best selling" doesn't mean "hight quality." Effectively dithered 8 bit might be enough for _some_ music such as commercial music with heavy dynamic compression, but we must use the most demanding music out there, not the easiest.


I don't disagree, just saying most listeners of the greater music consuming public would be satisfied.  Most pre-recorded cassette tapes were not even high bias and the masses flocked to the dreadful early 128 MP3 codecs.


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## 71 dB

The genius of dither is any 8 bit signal can pretend being 24 bit by pushing all the "last 16 bits stuff" onto the dither noise to do and when the dither can't do that because it's only 8 bits too, it gets truncated and badly distorted, but distorted noise is just noise. Nobody hears the difference. That's how we get rid of the distortion of loosing bit depth. Instead of distorting the signal we distort the noise were it doesn't matter. All we lose is dynamic range because the noise floor raises.


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## 71 dB

old tech said:


> I don't disagree, just saying most listeners of the greater music consuming public would be satisfied.  Most pre-recorded cassette tapes were not even high bias and the masses flocked to the dreadful early 128 MP3 codecs.



You are correct. I just don't see the link to 24 bit vs 16 bit debate. 
Those who think 16 bit is not enough are certainly not fans of pre-recorded cassettes or 128 kbps mp3s!


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## SoundAndMotion (Aug 31, 2020)

old tech said:


> The signal is all there, even if drowned by noise.





71 dB said:


> It's not false. The signal is all there, part of it is just masked by the noise.





SoundAndMotion said:


> [snip]
> One can't just pick out bits (lol) of what he says, though. Early on (0:42) he says *"if you do it right",* which he later (3:52) explains and demonstrates to mean *adding appropriate dither.*
> [snip]
> A passage hovering around -55 to -60dBFS will disappear if *recorded at or truncated to 8 bits*. So the signal is not all there.


I tried to be clear that "doing it right" means adding dither. If you don't add dither to a signal that is -60dBFS, whether recorded at 8 bits or truncated to 8 bits from 16 or 24 bits, there is no signal. There's also no noise. *At 8 bits, -60dBFS without dither is all zeroes!!*
Where is the signal and where is the noise in all zeroes?

Edit: Also, I question the concept that a signal buried deep in noise is "there". There are tricks one can use, like signal averaging, dither and noise shaping, *if one prepares ahead of time.* But if I give you a file with a signal to which I've added enough noise to destroy the signal, the signal is no longer there in a real sense, perhaps in an abstract form (mathematically) or philosophically, but... really?
If a book is completely burned, and you save the ashes and the smoke, is the book still there?


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## 71 dB

SoundAndMotion said:


> I tried to be clear that "doing it right" means adding dither. If you don't add dither to a signal that is -60dBFS, whether recorded at 8 bits or truncated to 8 bits from 16 or 24 bits, there is no signal. There's also no noise. *At 8 bits, -60dBFS without dither is all zeroes!!*
> Where is the signal and where is the noise in all zeroes?



When your signal gets completely below the least significant bit, the quantization noise starts to correlate with your signal so much it's actually -100 % correlation. In other words your signal and quantization noise cancel each other completely and the result is silence.


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## SoundAndMotion (Aug 31, 2020)

71 dB said:


> When your signal gets completely below the least significant bit, the quantization noise starts to correlate with your signal so much it's actually -100 % correlation. In other words your signal and quantization noise cancel each other completely and the result is silence.


That's philosophical drivel. The quantization error *IS* the signal. They don't both exist and cancel each other. The signal is not digitized at all, or is completely discarded (if truncated from a higher bit level).


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## 71 dB

SoundAndMotion said:


> That's philosophical drivel.



So? It works and explains what happens. I find these philosophical drivels useful in understanding digital audio (and many other things).


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## SoundAndMotion (Aug 31, 2020)

71 dB said:


> So? It works and explains what happens. I find these philosophical drivels useful in understanding digital audio (and many other things).



I have no problem with pedagogic tricks to help understanding something. You need to be careful with definitions of "noise" and "error", though. Go for it. But when you write (and others read):



old tech said:


> The signal is all there, even if drowned by *noise*.





71 dB said:


> The signal is all there, part of it is just masked by the *noise*.



Arguing that a file of all zeroes is really the signal fully and completely there, except the noise has perfectly cancelled it to all zeroes, would defy anyone's definition of "noise".


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## 71 dB

SoundAndMotion said:


> 1. I have no problem with pedagogic tricks to help understanding something. You need to be careful with definitions of "noise" and "error", though. Go for it. But when you write (and others read):
> 
> 2. Arguing that a file of all zeroes is really the signal fully and completely there, except the noise has perfectly cancelled it to all zeroes, would defy anyone's definition of "noise".



1. Sure, but in this context noise and error are the same thing.

2. What I said applies to the _truncation process_. 10 years later when everybody has forgotten _why_ the data is all zeros it's just digital silence. The truncation process functions the same regardless of the amplitude of the signal to be truncated. What changes when the signal gets below the LSB of the truncated version is the error = noise in this context starts to correlate with he signal negatively in a very strong way. For example:

Signal = 0.25 LSB of 8 bit => Gets truncated to 0 LSB of 8 bits => error = noise = -0.25 LSB of 8 bit = -signal.

This is not only a "pedagogic trick." It's what happens mathematically. Error ( = noise) correlates with the signal and this correlation is signal amplitude dependent. The advantage of thinking this way is you don't need two _models_ for signals strong enough to not truncate into zeros and signals so weak the result is all zeros.


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## theaudiologist1

I still prefer 48khz over 44.1 due to compatability. I think the 44.1 multiples should just be dicontinued.

Also, is there a point of releasing digital recordings recorded at 24bits to vinyl? Imo only analog recordings should be on vinyl. It's pointless.


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## theaudiologist1

Does 24bit/32bit have an advantage over 16bit in terms of digital volume control? If you lower the volume of 16bit you will get a terrible 8-10bits but if you lower the volume of 24bit audio you get 20bits?


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## 71 dB (Sep 1, 2020)

theaudiologist1 said:


> Does 24bit/32bit have an advantage over 16bit in terms of digital volume control? If you lower the volume of 16bit you will get a terrible 8-10bits but if you lower the volume of 24bit audio you get 20bits?


Zeros get added to 16 bit so what comes out is not "terrible 8-10 bits" for example -6 dB (bit shift) you have still 16 bits of information it's just shifted


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## 71 dB

theaudiologist1 said:


> I still prefer 48khz over 44.1 due to compatability. I think the 44.1 multiples should just be dicontinued.
> 
> Also, is there a point of releasing digital recordings recorded at 24bits to vinyl? Imo only analog recordings should be on vinyl. It's pointless.



What "compatability" are you talking about? Typically digital audio devices support several samplerates.
The point of vinyls is to have vinyl distotion (many people like it) and large cover art.


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## dazzerfong

theaudiologist1 said:


> I still prefer 48khz over 44.1 due to compatability. I think the 44.1 multiples should just be dicontinued.
> 
> Also, is there a point of releasing digital recordings recorded at 24bits to vinyl? Imo only analog recordings should be on vinyl. It's pointless.



Then there are practically going to be no modern recordings are going to be on vinyl. Almost everything is recorded digitally.

People buy modern vinyl recordings for a whole host of reasons not related to sound quality itself. These include:

- Nostalgia
- Hipsters
- Vinyl album collection


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## bigshot (Sep 1, 2020)

theaudiologist1 said:


> I still prefer 48khz over 44.1 due to compatability. I think the 44.1 multiples should just be dicontinued. Also, is there a point of releasing digital recordings recorded at 24bits to vinyl? Imo only analog recordings should be on vinyl. It's pointless.



You can master high data rate audio for LP release and it really doesn't make any audible difference. I would think a digital master would be less subject to dropouts or noise. But there wouldn't be any difference between a tape master or a digital copy of a tape master. Likewise, any data rate above audible transparency doesn't matter either. I guess humans have an urge to make numbers and formats line up nice and tidy, but what really matters is what you hear. When it comes to that, perfect sound is perfect sound.


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## Davesrose (Sep 2, 2020)

bigshot said:


> You can master high data rate audio for LP release and it really doesn't make any audible difference. I would think a digital master would be less subject to dropouts or noise. But there wouldn't be any difference between a tape master or a digital copy of a tape master. Likewise, any data rate above audible transparency doesn't matter either. I guess humans have an urge to make numbers and formats line up nice and tidy, but what really matters is what you hear. When it comes to that, perfect sound is perfect sound.



I know you like to always simplify by saying perfect sound is always perfect (and anything recorded 16 bit is perfect).  This seems an absolute....I'm not arguing that 16bit is a good audio container.  I'm just seeing that many folks judge sound by their sound system: that often does have EQs and DSPs tied in with said compressed mp3/FLAC/however high bit-depth audio file.  IMO also true for latest slight trend in vinyl uptick: albums that are from a digital source and folks enjoying the pleasurable distortions of playback equipment.


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## bigshot (Sep 3, 2020)

Perfect for the purposes of listening to commercially recorded music with human ears. Sound processing has enough headroom to work audibly perfect at 16/44.1. If it needs more, it can uprez, perform the processing, then downrez. There isn’t anything in commercially recorded music to require more.

in my experience, even digitally mastered LPs have all the noise and distortion slathered on. That comes from the limitations of the format, not the master.


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## Davesrose

bigshot said:


> Perfect for the purposes of listening to commercially recorded music with human ears. Sound processing has enough headroom to work audibly perfect at 16/44.1. If it needs more, it can uprez, perform the processing, then downrez. There isn’t anything in commercially recorded music to require more.
> 
> in my experience, even digitally mastered LPs have all the noise and distortion slathered on. That comes from the limitations of the format, not the master.



I'm not really sure your argument that anything that can process 16/44.1 has enough headroom and is perfect for any ears.  My point was about how it's processed (through further DSPs and EQs)....to reach that final output.  Or are you saying all bluetooth headphones sound the same if they receive 16/44.1?


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## bigshot (Sep 3, 2020)

The most recent Bluetooth codecs I’ve heard are transparent. They weren’t when I first investigated Bluetooth. With DSPs, I think processing power is more important than data rate. If you’re starting with a CD, which is transparent and encompasses just about all commercial music, the ability of the processor to do the processing is likely much more of an impediment than whether a recording is 16/44.1 or 24/96. I run CDs through all kinds of DSPs and I’ve never run into problems. Have you?


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## Davesrose

You keep rambling on about transparency with any system independent of what extra EQ or DSP (especially now, virtual surround systems) exist.  Again, I’m not going to argue that a CD is a good carrier system.  My current argument is with this idea that anything that accepts 16bit is unadulterated.  I think it’s something else you say you run DSPs with your CDs...that proves my point.  If anything is perfect by itself (and any audio is the same)...why are you running DSPs to begin with?


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## bigshot (Sep 3, 2020)

When did I say everything that does 16/44.1 is transparent? I said THE FORMAT is transparent. Obviously if you use a CD player in a cheap boom box or clock radio, it isn’t transparent. The CD player built into it probably is, but the rest of the system isn’t so it doesn’t matter.

I use DSPs to correct errors in mastering and to calibrate my system.


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## Davesrose

bigshot said:


> When did I say everything that does 16/44.1 is transparent? I said THE FORMAT is transparent. Obviously if you use a CD player in a cheap boom box or clock radio, it isn’t transparent. The CD player built into it probably is, but the rest of the system isn’t so it doesn’t matter.
> 
> I use DSPs to correct errors in mastering and to calibrate my system.



I'm not going to try to review all your posts, but you're doing it in this post: assuming the CD player built into a cheap boom box is audibly "transparent" (which I would understand as fully utilizing all capabilities of the digital file).  Bose would be a good example of audio technology that has a history of using EQ to make up for deficiencies of frequency range with their  speakers.  If we're theorizing about a cheap system's CD player: who knows, there could be issues with correct tracking....and there's certainly more stages of the signal going from DAC, whatever EQ settings the device has for balancing audio, to transducers.  Or as you're now indicating: you side with changing that "transparent" audio signal with your own preferences (IE opinion about mastering or what sounds best with your system).  There's also the issue of theoretical limits of a format vs actual utilization of recording (most all recordings falling short of absolute DR or boundaries of FR of a 16bit container).


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## bigshot (Sep 3, 2020)

If you build an all in one component like a clock radio, there are certainly things limiting the fidelity (primarily speakers). Bose can try to EQ to take the curse that, but it's still going to be just as inaccurate. I'm not arguing that. I am saying the digital player hardware itself is all audibly transparent because that is what it's designed to be. The standards for digital players playing 16/44.1 ensures audible transparency. If they aren't able to do that, they aren't meeting spec. I have a $40 Walmart DVD player that is audibly transparent through line out. Every player I've ever owned has been audibly transparent through line out. If a cheap all-in-one piece of kit isn't audibly transparent, it is MUCH more likely to be due to the limitations of the mechanical components like speakers, and to a much lesser extent the analog components like capacitors or filters, not the digital ones. Digital disk player hardware- DACs and transports- are mass produced to meet specs as off-the-shelf parts. If you go out and buy a bunch of players, odds are the electronics inside of them are very similar if not identical. Transparent DAC chips cost just a few dollars. It isn't like there is a range of audible quality that relates to the price point. If a CD player doesn't produce sound to spec it is either defective from improper manufacture or deliberately hobbled to create a house sound, which isn't the fault of the player. That is defective by design.

The problem is that even if an audibly different digital player exists somewhere, it would account for what? .01% of the market? Probably considerably less. When people focus on the rare and undocumented exceptions so much, they give a misleading impression. No one needs to worry about whether a CD player is audibly transparent or not when they shop at Amazon for a player for their home. But there are many, many audiophiles that believe they all sound different because of people who hammer away at minute and meaningless exceptions instead of giving them real practical advice. The tendency to be over-focused on things that just aren't important isn't just limited to audiophools. It's common around here too. Sometimes I feel like we are playing out the story of the "Princess and the Pea". No matter how incredibly small it is, it still matters to some people.

I think we both are saying the same thing. You're just adding "wiggle room" for exceptions to the rule that none of us have ever run across personally. Every player I have ever owned and tested has been audibly transparent, since the very early days of the format when they made players without oversampling. (And then the difference, although audible, was negligibly small.) All the back and forth over this just muddies the water for the truth. There really isn't any reason to worry about digital components when it comes to audibility. If there is a problem, it is not likely at all to be because of the digital disc player hardware itself.

I'm afraid I don't understand what you are getting at with EQ and DSPs. That is precisely adjustable coloration to correct for acoustic and mechanical problems in transducers or the room. Signal purity isn't fidelity anymore once it reaches the real world. Although digital audio can easily be audibly perfect, when it comes to headphones, speakers and rooms there's no such thing as complete fidelity... only combinations of compromises that have better fidelity than other ones. DSPs and EQ can help to correct for inevitable acoustic problems that exist in the real world of living rooms. You can correct error, or you can add it for aesthetic reasons. There's nothing wrong with that at all. We all hold the car keys to our own systems.


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## Davesrose

I agree that on most concepts we have no disagreements.  The only point I'm disagreeing with is an assumption that any digital audio device will sound the same.  I'm not disagreeing that 16bit audio format is capable, or that a cheap DAC chip can render everything within human hearing.  It's just that in my experience, a device might sound different due to enabled DSPs.  Say a TV device: it could have a dynamic range limiter enabled, or some kind of virtual surround.  I've also noticed the same with current Windows 10 standards.  Certain software might have a set EQ or surround scheme that also has a different tonal balance.


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## bigshot

Yes, a device will sound different with DSPs. They will sound different with tone controls and different volume levels too. But that’s signal processing. That’s changing the sound by definition. Bypass the signal processing and they’re all transparent. At least every one I’ve run across.


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## BigZ12 (Sep 14, 2020)

theaudiologist1 said:


> I still prefer 48khz over 44.1 due to compatability. I think the 44.1 multiples should just be dicontinued.
> 
> Also, is there a point of releasing digital recordings recorded at 24bits to vinyl? Imo only analog recordings should be on vinyl. It's pointless.


So..please tell a "noob" like me:
What do I set my computer to, when using a Audioquest Dragonfly Cobalt, Philips Fidelio X3 and play music with Apple Music? 24bit/44.1, 48, 88.2 or 96khz??
In the manual, Audioquest suggest 44.1khz mostly.

There's also an internal setting in Apple Music: Windows Audio Session or Direct sound. 16/24bit and 44.1 to 192khz. What should I use?

I have a trial with Tidal HiFi, where the Cobalt autodetect the bitrate and sampling freq. itself it seems. (green dragonfly (44.1khz) with "hifi" tracks, and purple for MQA/master)


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## KeithPhantom

BigZ12 said:


> So..please tell a "noob" like me:
> What do I set my computer to, when using a Audioquest Dragonfly Cobalt, Philips Fidelio X3 and play music with Apple Music? 24bit/44.1, 48, 88.2 or 96khz??
> In the manual, Audioquest suggest 44.1khz mostly.
> 
> ...


Can you use something like ASIO or WASAPI? I don't trust the internal audio processing Windows has implemented. But if you don't have any option, use the sample rate/bit rate most of your music has so you limit the amount of processing to be done and any possible nonlinearity introduced by bad software.


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## BigZ12 (Sep 14, 2020)

KeithPhantom said:


> Can you use something like ASIO or WASAPI? I don't trust the internal audio processing Windows has implemented. But if you don't have any option, use the sample rate/bit rate most of your music has so you limit the amount of processing to be done and any possible nonlinearity introduced by bad software.


Asio and WASAPI are not possible with Apple Music. As I said, just settings under preferences with the following options:
Windows Audio session and Direct Sound - 16/24bit - 44.1-192khz

When connecting the Dragonfly Cobalt to Windows 10, I get 24bit/44.1 to 96khz as options.


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## theaudiologist1 (Sep 17, 2020)

BigZ12 said:


> So..please tell a "noob" like me:
> What do I set my computer to, when using a Audioquest Dragonfly Cobalt, Philips Fidelio X3 and play music with Apple Music? 24bit/44.1, 48, 88.2 or 96khz??
> In the manual, Audioquest suggest 44.1khz mostly.
> 
> ...


always stay bitperfect. If you have 44.1khz files (most CD rips are), go with 44.1. If you want to upsample, upsample in multiples of 2 (44.1 ---> 88.2/176.4/352.8, 48 ---> 96/192/386) to not cause quantization errors.


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## Brahmsian

Audirvana ups everything to 32 bits, so that’s what I listen to. Having said that, if you have a good recording to start with and decent headphones or speakers that are driven properly, even MP3 and AAC files sound quite good. Was listening to my old MP3 of Dudamel conducting Mahler’s 9th with the Los Angeles Philharmonic. Really enjoyed it. Far from bad sound despite the compression. And we’re talking a large-scale orchestral piece.


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## theaudiologist1

Brahmsian said:


> Audirvana ups everything to 32 bits, so that’s what I listen to. Having said that, if you have a good recording to start with and decent headphones or speakers that are driven properly, even MP3 and AAC files sound quite good. Was listening to my old MP3 of Dudamel conducting Mahler’s 9th with the Los Angeles Philharmonic. Really enjoyed it. Far from bad sound despite the compression. And we’re talking a large-scale orchestral piece.


what's the best lossy format? AAC, OGG Vorbis, or Opus?


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## bigshot

For high data rate, the best is AAC. And always use VBR. AAC can even go past 320 in VBR if necessary. (I can't imagine it being necessary, but it can.)


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## theaudiologist1

bigshot said:


> For high data rate, the best is AAC. And always use VBR. AAC can even go past 320 in VBR if necessary. (I can't imagine it being necessary, but it can.)


but I heard OGG/Opus can have higher quality at lower bitrate. Also, AAC is proprietary.


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## bigshot

AAC is not proprietary. It’s an open standard, and it’s widely supported. If you want super small files and don’t care so much about sound quality, opus is better. If you are looking for something that sounds as good as lossless with a smaller file size AAC is the best. I use AAC 256 VBR and it is as good as lossless at a much more convenient size.


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## SoundAndMotion

AAC is patented (link). End users don’t need to pay, but manufacturers or developers of codecs do. That’s why supporters of free and open source software (link) often won’t use it.


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## bigshot (Oct 8, 2020)

I'm pretty sure the Fraunhofer FDK AAC codec is open source and it encodes and decodes AAC files. (I think it's backward engineered or something...) In effect though, it doesn't really matter to users though because AAC is the second most supported codec for compressing high fidelity music, second only to its predecessor MP3. It's not going anywhere. I think he was most concerned with compatibility, We don't want to confuse this guy. We all agree that AAC is definitely the best choice for a lossy codec with both the highest fidelity sound quality and compatibility.


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## Steve999 (Oct 8, 2020)

AAC is excellent. I am lousy at hearing lossy artifacts (and hope to stay that way) so anything above 192 kbps in any moderately effective codec is fine with me. I honestly don’t know what’s better in stereo in high bitrates, AAC or Opus. I’ve seen some streaming services turning to Opus if they sense one way or another (the tech is beyond me!) that the downstream gear can handle it. I’d bet the house on either one being transparent to me for the great majority of music under real world listening conditions at 192 kbps or above.

One thing I’ve worried about for a few years is how lossy codecs are handled by simulated surround sound setups, a.ka. upmixing to 2.1, 3.1, 5.1, etc. Industry seems to be quietly addressing this. I‘ve read that newer Bluetooth distributions can be better for that, since they distribute the lossy bits more evenly over the spectrum, so the upmixing algorithm steers the signal to the different channels more  or less as hoped. I’ve also read that at higher bitrates Opus specifically tunes its bitrates so that it can be used as a stream for surround sound. Here, from the horse’s mouth: https://wiki.xiph.org/index.php?title=Opus_Recommended_Settings&mobileaction=toggle_view_desktop

A great deal of my listening is lossless streaming now (Amazon HD, Qobuz) (16-bit will do, the rest is marketing voodoo!), so a lot of this falls away as any kind of concern for me.  Although right now I am listening to Apple Music (AAC) because, well, I feel like it. It was right there on my Apple TV menu and I’m so sick of the political junk on TV so I said what the hey. And in the meantime I did a head-fi Sound Science drive-by.


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## Brahmsian

Maybe someone can react to this:

”when a musician is being mixed in, say, 96kHz and and 24-bit, the output of that mixing session, the master tape or file that’s released for mastering to CD, tape, vinyl, etc. is very likely going to be 96kHz, 24-bit. So anything that’s done in mastering, including dithering down from 24-bit to 16-bit, downsampling from 96kHz to 44.1kHz, that’s all going to affect sound. These days, the changes are minor, possibly not even audible unless you’re really trained in critical listening. But there’s also the presumption that’s being done right. That there’s not record company mandate for crazy compression levels to make the music sound louder on the radio, etc.”
https://www.quora.com/Can-you-hear-...ty-between-24bit-and-32bit-192khz-music-files


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## KeithPhantom (Oct 17, 2020)

Brahmsian said:


> So anything that’s done in mastering, including dithering down from 24-bit to 16-bit, downsampling from 96kHz to 44.1kHz, that’s all going to affect sound.


Again, changes -120 dBFS from the fundamental are NOT noticeable by humans, including outliers, even if you play the fundamental at 120 dB SPL (which I do not recommend), the differences are going to be all the way down to 0 dB SPL (assuming that 1 dBFS == 1 dB SPL).


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## Brahmsian

KeithPhantom said:


> Again, changes -120 dBFS from the fundamental are NOT noticeable by humans, including outliers, even if you play the fundamental at 120 dB SPL (which I do not recommend), the differences are going to be all the way down to 0 dB SPL (assuming that 1 dBFS == 1 dB SPL).


He grants that they are ”possibly not even audible,” but then he says that it presumes that the dithering and downsampling to 16/44.1 are being done right--for instance, that there is no ”record company mandate for crazy compression levels to make the music sound louder on radio, etc.”

My own thinking on this is that if given the choice between a dithered and downsampled file and one without dithering and downsampling, I would choose the latter, all else being equal, even if I can’t hear a difference. I have the space for it, my CPU can handle the extra processing, and when they’re on sale on Qobuz the price difference is often negligible. But this is based on the assumption that Qobuz is selling me a copy of the ”master” and isn’t simply adding bits and upsampling redbook files. That is, I assume that when I buy a 24/88.2 file it was recorded that way. Of course, my DAC converts all my music to 32 bit anyway, and I’m still not sure what the point of that is.


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## bigshot (Oct 18, 2020)

Phrases like "possibly not even audible" are weasel words. There isn't a chance in hell that any human being is going to hear something down that low no matter how well they're trained. Human hearing has limits and -120dB is so far beyond them, it's pitiful.

I've found that people use vague phrases like this for one of two reasons... 1) they want to protect themselves from someone speaking up with a crazy unlikely exception like "if you were in the atmosphere of Venus and the sound was inside your ear canal you could hear it!" or 2) they want to instill doubt so you say to yourself, "If he says I *probably* don't hear it unless I'm trained, then I believe I can hear it because I try really really hard!"

Audiophiles spend more time talking about inaudible sound than they do things they can actually hear... and that makes sense... Digital audio and modern electronics have solved the problems of fidelity to the level of human hearing. The only think left to sell you is equipment that improves things beyond your ability to hear.

The truth is that plain vanilla CD sound with dithering is already far into the range of overkill. Even without dithering it's more than you need.


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## Brahmsian

bigshot said:


> Phrases like "possibly not even audible" are weasel words.


To be fair to the person who wrote what I quoted from, I think the ”possibly not even audible” is in relation to the rare instances where the high resolution master was tampered with when dithered/downsampled to CD specs. He provides the following example:

”If you’re able to get the exact data that’s released by the artist, you can be certain that you’re hearing that music as-intended. This really started to become a well known problem when a few artists, including Metallica, released ridiculously overcompressed albums… and then those same tracks showed up on video game and sounded so much better.”


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## bigshot (Oct 19, 2020)

Iit didn’t sound bad because of the technical specs, but because of lousy engineering. You can add that as a third reason if you want.


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## Brahmsian

bigshot said:


> Iit didn’t sound bad because of the technical specs, but because of lousy engineering. You can add that as a third reason if you want.


Correct but in the end it makes no difference if, for whatever reason, they're going to degrade the sound on the redbook compared with the master as recorded. I don't imagine it happens very often though.


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## 71 dB

Brahmsian said:


> Maybe someone can react to this:
> 
> ”when a musician is being mixed in, say, 96kHz and and 24-bit, the output of that mixing session, the master tape or file that’s released for mastering to CD, tape, vinyl, etc. is very likely going to be 96kHz, 24-bit. *So anything that’s done in mastering, including dithering down from 24-bit to 16-bit, downsampling from 96kHz to 44.1kHz, that’s all going to affect sound*. These days, the changes are minor, possibly not even audible unless you’re really trained in critical listening. But there’s also the presumption that’s being done right. That there’s not record company mandate for crazy compression levels to make the music sound louder on the radio, etc.”
> https://www.quora.com/Can-you-hear-...ty-between-24bit-and-32bit-192khz-music-files



Reacting to the *bolded* part: For human ear the sound is not affected unless the dithering and downsampling is somehow done completely wrong, but if they are done half-decently there is no affect on sound for human ears (bat ears perhaps?).


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## bigshot

"Possibly not even audible..." You're right. Whenever I've supervised a mix and we monitor the bounce down, it hasn't been audible. It shouldn't be audible.


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## Brahmsian

Here is the sort of thing I expect from truly audiophile-grade music download services. I don't care if it's PCM or DSD or what the resolution is as long as a) it's a good recording and b) it's an exact copy of the master in whatever resolution it was recorded in. This way, I know I'm getting the best version possible.


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## bigshot (Oct 20, 2020)

The first paragraph just says they add a data fork to tag the music. That is nothing special at all. You can do that with iTunes. I don't know what that second paragraph means. It sounds like sales pitch. First of all, there is no such thing as a "native DSD" master. You can't mix in "native DSD". It has to be bumped to PCM to do that, and each song is a separate project file. Until an album is mastered, the songs aren't in sequential order and the levels and EQ aren't all matched to play through as an album. There is absolutely no audible difference between a 24/96 mastered album and a 16/44.1 bounce down to a CD, so the file on the optical CD is audibly identical to the mastered file. If music is poorly mastered, that has nothing to do with how early in the process it is. Digital has no generation loss. This whole paragraph is made up.

Every download service has the same kind of boilerplate saying they have the best master. But the fact is, they get what the distribution network of the record company sends them. In many cases, that is taken from CD rips. (Even a lot of the ones that claim to be HD audio.) There's nothing wrong with that at all if the CD has been well mastered. If you then take that rip and put it in a DSD file, it's just going to be a massively large file that sounds exactly the same as the CD and the 24/96 master. It doesn't matter what file format it is.


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## reginalb (Oct 21, 2020)

Brahmsian said:


> Here is the sort of thing I expect from truly audiophile-grade music download services. I don't care if it's PCM or DSD or what the resolution is as long as a) it's a good recording and b) it's an exact copy of the master in whatever resolution it was recorded in. This way, I know I'm getting the best version possible.



Uh, I don't believe for one moment that the source for most music for download or streaming today is optical media. They aren't ripping CDs, unless they're for some reason caught up in some dinosaur of a process.




bigshot said:


> ...But the fact is, they get what the distribution network of the record company sends them. In many cases, that is taken from CD rips...



I mean, seriously? How can that be? I work in tech, so maybe the recording industry is about 2 decades behind, but I just can't believe that would be the case. What the hell would be the point of writing it to optical media just to remove it from that optical media and send it out that way? That makes zero sense from a cost perspective.


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## colonelkernel8

reginalb said:


> Uh, I don't believe for one moment that the source for most music for download or streaming today is optical media. They aren't ripping CDs, unless they're for some reason caught up in some dinosaur of a process.
> 
> 
> 
> ...


Maybe not an actual CD rip, but the raw image for a CD.


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## KeithPhantom

colonelkernel8 said:


> Maybe not an actual CD rip, but the raw image for a CD.


Or master tapes. I haven't found many direct-to-DSD 1-bit PCM (or PDM/PWM if you prefer or decimation D/S) ADCs. Usually, these companies have "audiophile-standard" equipment, it wouldn't be weird of them using tapes at this point and mixing/mastering in analog equipment.


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## KeithPhantom

Here some articles about DSD and how it works:

https://www.grimmaudio.com/site/assets/files/1088/dsd_myth.pdf
http://users.ece.utexas.edu/~bevans...ersion/AP_Understanding_PDM_Digital_Audio.pdf


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## colonelkernel8

bigshot said:


> The first paragraph just says they add a data fork to tag the music. That is nothing special at all. You can do that with iTunes. I don't know what that second paragraph means. It sounds like sales pitch. First of all, there is no such thing as a "native DSD" master. You can't mix in "native DSD". It has to be bumped to PCM to do that, and each song is a separate project file. Until an album is mastered, the songs aren't in sequential order and the levels and EQ aren't all matched to play through as an album. There is absolutely no audible difference between a 24/96 mastered album and a 16/44.1 bounce down to a CD, so the file on the optical CD is audibly identical to the mastered file. If music is poorly mastered, that has nothing to do with how early in the process it is. Digital has no generation loss. This whole paragraph is made up.
> 
> Every download service has the same kind of boilerplate saying they have the best master. But the fact is, they get what the distribution network of the record company sends them. In many cases, that is taken from CD rips. (Even a lot of the ones that claim to be HD audio.) There's nothing wrong with that at all if the CD has been well mastered. If you then take that rip and put it in a DSD file, it's just going to be a massively large file that sounds exactly the same as the CD and the 24/96 master. It doesn't matter what file format it is.


Doesn't Pyramix keep everything in DSD when mixing?


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## bigshot (Oct 21, 2020)

colonelkernel8 said:


> Maybe not an actual CD rip, but the raw image for a CD.



I have a friend who works in music licensing. He worked on building the library for the company of the files that would be licensed out. For legacy titles on analog recordings, it can be a royal pain to pull the masters and make a transfer. A CD is all signed off on and approved by definition. I remember hearing him saying that they were ripping CDs to PCM for a lot of older material. Stuff that was produced digitally probably does exist as a mastered digital file, but the older stuff wasn't always archived in that form. Also, just because a company has the rights to license a recording, it doesn't mean that they possess the physical master for it.

It doesn't really matter though because you can take a mastered 16/44.1 PCM file used in CD replication and rip a 16/44.1 PCM file ripped off the CD and they will be identical. The format doesn't matter.


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## bigshot (Oct 21, 2020)

colonelkernel8 said:


> Doesn't Pyramix keep everything in DSD when mixing?



I went looking for a "direct DSD" recording back when I was doing my tests on SACD vs CD. I found a few on the Pentatone label. There aren't many of them because all the recording and mixing has to be done live on the fly since the DSD stream isn't able to be modified without converting to PCM. It doesn't matter though, because direct DSD doesn't sound any different to human ears than 16/44.1 PCM.


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## KeithPhantom

bigshot said:


> I went looking for a "direct DSD" recording back when I was doing my tests on SACD vs CD. I found a few on the Pentatone label. There aren't many of them because all the recording and mixing has to be done live on the fly since the DSD stream isn't able to be modified without converting to PCM. It doesn't matter though, because direct DSD doesn't sound any different to human ears than 16/44.1 PCM.


You can always record and mix in the analog domain if having "pure DSD" is your goal. It won't be purely digital though...


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## bigshot

Kind of defeats the purpose.


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## Davesrose

bigshot said:


> I have a friend who works in music licensing. He worked on building the library for the company of the files that would be licensed out. For legacy titles on analog recordings, it can be a royal pain to pull the masters and make a transfer. A CD is all signed off on and approved by definition. I remember hearing him saying that they were ripping CDs to PCM for a lot of older material. Stuff that was produced digitally probably does exist as a mastered digital file, but the older stuff wasn't always archived in that form. Also, just because a company has the rights to license a recording, it doesn't mean that they possess the physical master for it.
> 
> It doesn't really matter though because you can take a mastered 16/44.1 PCM file used in CD replication and rip a 16/44.1 PCM file ripped off the CD and they will be identical. The format doesn't matter.



But are there many studios that archive their album with a consumer pressing of the CD?  I'm just wondering that it may not be that archival, as there has been some notorious batches of plants not maintaining standards and having discs that are susceptible to CD rot.

I'm not familiar with the audio industry, but I do think it's interesting what TV broadcast and film industries have done for archiving.  Older TV shows might have been filmed on 35mm film, and if there's demand for a HDTV master, they can go back to that stock.  Before the realization of home distribution, TV studios wouldn't have an archive.  Some of the oldest episodes of Dr Who have survived because people have found original 16mm films in an attic (which BBC sent for international distribution that was supposed to be destroyed after broadcast).  Star Trek Next Gen was popular enough for the studio to go back to all film elements for a new digital edit with the HD mastering (that during broadcast were just scanned to video and edited in analog SD).  Likewise, for film restorations, its been quite a few years that a company will scan 35mm film to 4K and 70mm film to 8K.  They do their color correcting, de-specking, etc.  Then they've had a 4K master in a digital format, and they still print the final edit to new film stock.  They've spent the money and effort, so they want to be safe in having the digital backup and analog.  To also get back to Star Trek NG, it did have some shots that had early computer graphics.  They used 3D software that's now considered primitive and very hard to try to get running on a modern computer (similar also to early video codecs older folks might remember). For those, shots....especially being in HD and modern standards, it was easier to just recreate in new software.


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## bigshot (Oct 21, 2020)

Davesrose said:


> But are there many studios that archive their album with a consumer pressing of the CD?  I'm just wondering that it may not be that archival, as there has been some notorious batches of plants not maintaining standards and having discs that are susceptible to CD rot.



Well the fella I know says that their archive is PCM files on a server. I would guess they have multiple cloud backups too. But they had to source some of the music they licensed from CDs because the masters weren't available. The example he mentioned to me was a country musician who had a career that spanned over fifty years and multiple record labels. They received master files on some of his catalog, but they had the rights to license music from his entire career, so they had to source some material from commercial CDs. I believe Universal Music had to do something similar after a fire destroyed some of their vaults. A lot of music was archived there in all kinds of different formats... whatever format was standard at the time. All it took was one bad backlot fire and it was all gone.

Yeah, there is a huge push to digitize as much old film as possible. The most problematic are the shows shot on videotape, and the ones that were shot on film and finished on tape. It can be hard to pull all the elements together again to rebuild a show from the film elements. The thing about studio archives is that they are very practical. They don't pay to do any work that isn't directly profitable. That means a lot of stuff falls through the cracks. It can take sleuthing to put it all back together again. Sometimes bits only survive in work prints. And often when shows are reformatted, the original film elements get cut up and the trims are lost. It's not always as organized as one might hope.


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## Davesrose

bigshot said:


> Well the fella I know says that their archive is PCM files on a server. I would guess they have multiple cloud backups too. But they had to source some of the music they licensed from CDs because the masters weren't available. The example he mentioned to me was a country musician who had a career that spanned over fifty years and multiple record labels. They received master files on some of his catalog, but they had the rights to license music from his entire career, so they had to source some material from commercial CDs. I believe Universal Music had to do something similar after a fire destroyed some of their vaults. A lot of music was archived there in all kinds of different formats... whatever format was standard at the time. All it took was one bad backlot fire and it was all gone.
> 
> Yeah, there is a huge push to digitize as much old film as possible. The most problematic are the shows shot on videotape, and the ones that were shot on film and finished on tape. It can be hard to pull all the elements together again to rebuild a show from the film elements. The thing about studio archives is that they are very practical. They don't pay to do any work that isn't directly profitable. That means a lot of stuff falls through the cracks. It can take sleuthing to put it all back together again. Sometimes bits only survive in work prints. And often when shows are reformatted, the original film elements get cut up and the trims are lost. It's not always as organized as one might hope.



Yeah, one thing I've gathered is that archiving for distribution to home media is very piecemeal.  Many of the oldest films have been lost because they were using highly flammable nitrate film emulsions.  When it comes to music studios, it would make sense that they have a priority for the most popular albums being archived in a WAV format in redundant storage.  Assume for the smaller catalogues, if it's from CD, it's still a master pressing vs the mass produced runs.

For awhile, Warner Bros had stopped commercial printing of their classic movies, and you'd order directly from them for a DVD-R!  Now with streaming, it's a heyday for distribution of older catalogues.  You have the older movies that had been restored for blu-ray now making it to original 4K (with possibly more edits for an Atmos/DTS:X mix).  The earliest digitally filmed movies don't have as much an upgrade since they might have been filmed DV without HDR.

I have seen the workflows from ILM when it comes to VFX.  What you're mentioning about sleuthing studio archives is exactly what they did for the Star Trek NG remasters.  Even though the initial output was just going to be NTSC, and with tight deadlines they edited on analog consoles, ILM was involved with the special effects.  ILM already had a standard for filming the Star Trek movies, and applied the same workflow for the series.  They had many passes filmed in VistaVision (higher resolving power film than standard 35mm cinema standards).  The stage shots were done in Super-35mm.  Then all these film elements were scanned to video for editing.  For the blu-rays, they went back to finding all film elements.  It was a lot of work finding some of the original elements, but the outcome is pretty spectacular (the VFX still holds up).  This compared to the original Star Trek series, where they were able to do quite a bit of clean up and color grading of the stage shots.  The VFX didn't hold up as well since they did shoot and print to academy film (which the quality difference wasn't seen in 60s analog TV).  For that mastering then, they had 3D artists re-imagine all the special effects.


----------



## Brahmsian (Oct 21, 2020)

bigshot said:


> 1)The first paragraph just says they add a data fork to tag the music. That is nothing special at all.
> 
> 2) First of all, there is no such thing as a "native DSD" master. You can't mix in "native DSD". It has to be bumped to PCM to do that, and each song is a separate project file. Until an album is mastered, the songs aren't in sequential order and the levels and EQ aren't all matched to play through as an album.
> 
> ...


1) NativeDSD claims to have a custom procedure that involves ”extra work” to ensure the most accurate metadata possible. They mostly host classical music labels, and I think they realize that there’s a problem with how that metadata is handled by other services, which basically treat it like pop music. So their boast about metadata might be aimed at classical music listeners who see how the big download sites and streaming services fail to treat that metadata with the extra care it deserves.

2) They make a distinction between the DSD Edit Master and an SACD Cutting Master, and then again between the ”deliverable release DSD” and the ”DXD edited master itself.” Whether or not it’s all a distinction without a difference is another matter.

3) I’m going off what the guy I quoted above said. You’re saying that a master recording will always be just as compressed as the final commercial release? There is never a version of the recording with less compression?

4) Not only are DSD files massive, a single recording can easily cost you $40 and up. Some people will pay that and more because it’s multi-channel. If you’re happy buying the redbook version, great. Let others spend their money as they please.


----------



## Davesrose

Brahmsian said:


> 1) NativeDSD claims to have a custom procedure that involves ”extra work” to ensure the most accurate metadata possible. They mostly host classical music labels, and I think they realize that there’s a problem with how that metadata is handled by other services, which basically treat it like pop music. So their boast about metadata might be aimed at classical music listeners who see how the big download sites and streaming services fail to treat that metadata with the extra care it deserves.
> 
> 2) They make a distinction between the DSD Edit Master and an SACD Cutting Master, and then again between the ”deliverable release DSD” and the ”DXD edited master itself.” Whether or not it’s all a distinction without a difference is another matter.
> 
> ...



I would think that blu-ray concerts are more popular for the surround sound marketing aspect (I personally really like them for hi-fi sound and visual as well).  I did invest in SACD and a good quality stereo player in the last decade (since then have an Oppo BD player that can output SACD multi-channel track).  With the analog outputs, I still prefer my stereo SACD player (sounds a bit warmer and I'm sure due to coloration).  When Sony was marketing DSD for studio production, classical albums were certainly the most popular followed by jazz (and some rock that was literally just PCM mixed to surround).  Many of my SACDs aren't hybrids, so if I'd want to digitize I'd have to find some hack.  Why bother, I'll just enjoy those titles and will see about options if and when I can't have SACD playback.  SA-CD.net had been pretty comprehensive about those classical recordings that were DSD.  Looks like SACD might have officially died in 2015.  SA-CD.net


----------



## castleofargh (Oct 26, 2020)

Brahmsian said:


> 1) NativeDSD claims to have a custom procedure that involves ”extra work” to ensure the most accurate metadata possible. They mostly host classical music labels, and I think they realize that there’s a problem with how that metadata is handled by other services, which basically treat it like pop music. So their boast about metadata might be aimed at classical music listeners who see how the big download sites and streaming services fail to treat that metadata with the extra care it deserves.
> 
> 2) They make a distinction between the DSD Edit Master and an SACD Cutting Master, and then again between the ”deliverable release DSD” and the ”DXD edited master itself.” Whether or not it’s all a distinction without a difference is another matter.
> 
> ...


I like that they don't try to pretend that DXD is DSD. It's a weird amalgam I've seen a few times and it bothers me. It's completely irrelevant in term of fidelity as both PCM and DSD can do better than what we can hear, and DXD is the super overkill big brother in that respect, but I do like honesty.


----------



## bigshot (Oct 22, 2020)

Brahmsian said:


> You’re saying that a master recording will always be just as compressed as the final commercial release? There is never a version of the recording with less compression?



There could be a *different* mastering created for a different purpose that has different engineering, but that isn't an "earlier master", it's just a different master. And engineering has nothing to do with format. There's really no reason to take a recording made with PCM and transcode it to DSD for consumers. It's just a great big giant bag wrapped around the exact same candy bar. I think that whole thing was just sales pitch. Every streaming service does pretty much the same thing as that. And all the file types they discuss all sound the same to human ears... as would plain vanilla PCM 16/44.1.

Multi channel is different. That is actually something different than other streaming services can deliver. They should be talking about how the multichannel recordings are made instead of the way they tag and format the files. There are much more efficient ways to distribute multichannel than as DSD files.

Davesrose, I've noticed that classical has shifted more to live concerts on blu-ray too. I guess it just gives them one more medium to license it to. Now that I think of it, I've bought many more blu-rays and DVDs than SACDs in the past couple of years. I like the fact that you just put an SACD in and it plays like a CD with no menu screen. But everything has a bunch of extra options and content now that SACD doesn't support.


----------



## Brahmsian

castleofargh said:


> I like that they don't try to pretend that DXD is DSD. It's a weird amalgam I've seen a few times and it bothers me. It's completed irrelevant in term of fidelity as both PCM and DSD can do better than what we can hear, and DXD is the super overkill big brother in that respect, but I do like honesty.


They do however point out that there are ways to edit in DSD and avoid converting to DXD except for crossfades. I think Channel Classics is one of the labels that do that. This is probably a phenomenon that is mostly limited to acoustic music.


----------



## theaudiologist1

Are audiophile music players like poweramp, neutron and audirvana just placebo snakeoil? At least when it comes to phones since Android/iOS only samples at 48kHz? I feel like I've wasted $4.


----------



## dazzerfong

theaudiologist1 said:


> Are audiophile music players like poweramp, neutron and audirvana just placebo snakeoil? At least when it comes to phones since Android/iOS only samples at 48kHz? I feel like I've wasted $4.



Not really, at least for Poweramp. Poweramp has its own audio engine that allows for a much more precise EQ. Also allows access to direct volume control, which makes the EQ changes go more extreme with less distortion. But, if everything's turned off, is it better? Nope. EXCEPT when dealing with USB DACs: native Android has terrible support for USB DACs in general which is really only resolved by using something like USB Audio Player.

Poweramp at least can support hi-res output without resampling, for what it's worth.

I personally bought Poweramp like 8 odd years ago because the Sony default player wasn't great, forgot about it, then only came back to it out of curiosity. Gotta say, looks very good now.


----------



## bigshot

theaudiologist1 said:


> Are audiophile music players like poweramp, neutron and audirvana just placebo snakeoil? At least when it comes to phones since Android/iOS only samples at 48kHz? I feel like I've wasted $4.



I can only speak for Audivana. I tried it to play multichannel music files and it was very temperamental. It skipped and had trouble. Plex played the same files smoothly.


----------



## SennheiserNoob

All I know is that in those blind online tests, I haven't been able to differentiate


----------



## Sonic Defender

SennheiserNoob said:


> All I know is that in those blind online tests, I haven't been able to differentiate


Of course not, nobody can if they're being honest.


----------



## magicscreen

Fact 1. Recording and mastering is made by using 24bit/96kHz-192kHz etc hi-res format and then dumbed down to CD quality

Fact 2. delta-sigma DACs use oversampling for reconstruct the lost information

Think logically and you can understand that high resolution audio is better.
Philips wanted 14 bits for the CD first.
The CD quality was only an economical decision.


----------



## bigshot

Wrong. But I don't think you are here to have a conversation, so I'll let you just be wrong.


----------



## peskypesky

bigshot said:


> Audiophiles spend more time talking about inaudible sound than they do things they can actually hear... and that makes sense... Digital audio and modern electronics have solved the problems of fidelity to the level of human hearing. The only think left to sell you is equipment that improves things beyond your ability to hear.



I think I'm going to make this my signature.


----------



## bigshot

If you do fix my typo! Thanks!


----------



## peskypesky

bigshot said:


> If you do fix my typo! Thanks!


will dew


----------



## 71 dB

magicscreen said:


> Fact 1. Recording and mastering is made by using 24bit/96kHz-192kHz etc hi-res format and then dumbed down to CD quality
> 
> Fact 2. delta-sigma DACs use oversampling for reconstruct the lost information
> 
> ...


Frustratingly we are living the age of "alternative facts", but the real facts nevertheless exist.

In music_ production_ higher resolution offers benefits. Bigger dynamic range means you don't have to optimize the use of 16 bit dynamic range at every point and there is no fear of massive processing of sound to cumulate into audible levels of noise. Bigger bandwidth is perhaps less beneficial, but in some instances may offer something. Such as recording ultrasonic frequencies and playing them back at lower sample rate to make those frequencies audible to humans.

_Consumers_ of music don't need higher resolution formats. When the music has been produced using higher resolution and the benefits have been gained, the end result is easy to optimize for CD. Music consumers do not need more than about 80 dB (~13 bits) worth of dynamic range (vinyl has 60 dB at best). There's simply limits of human hearing dictating these requirements. There are limits of listening environments. Say you have a very quiet room with 30 dB background noise level. 80 dB up from that is 110 dB. How much above that do you want to go? How much pain do you want your ears to feel? How much do you want to damage your hearing? How quiet sounds do you think you are able to hear temporarily when your ear have a been exposed to sound pressure levels of over 100 dB? Loud sounds rise the threshold of hearing. The dynamic range of hearing 120-130 dB, but not in one time! When your ears have rested for hours in silence you can hear sounds as quiet as 0 dB SPL (even below that around 3-4 kHz), but that's not the case when you blast music loud and your threshold of hearing has temporarily raised. That's why during the music listening session the dynamic range of your hearing is much less than 120-130 dB, about 70 dB. That's why you do not need more than 80 dB of dynamic range. It also explains why for many even vinyls have enough dynamic range.

Technically the more bits you have the more information you have, but the question is what information is relevant for human ears? Stuff we don't hear anyway can go. I don't hear ants farting so I don't care if CD can't produce sounds that quiet. I don't hear bats chirping so I don't care if CD can't produce those ultrasonic frequencies. Within the bandwidth and dynamic range specifications, digital audio doesn't loose information. DACs don't reconstruct "lost" information. They reconstruct the analog signal from the digital information. Oversampling is a "technically smart" way to do it. Within the bandwidth and dynamic range limitations digital audio can theoretically reproduce the original signal with 100 % accuracy. Higher resolution doesn't increase accuracy (you can't go above 100 %), but they push the limits further beyond the limits of human hearing. High resolution digital sound is not better for human ears. We can't hear the effect of more bits, because 44.1/16 already gets over our limits. 

Philips wasn't crazy wanting 14 bits for the CD first. As I mentioned, about 13 bits is what's needed. CD specifications were not economical decisions as at first CD players were very price, althou the prices dropped fast. The decisions were for the most part technological. Fortunately at that point in time, four decades ago, digital technology was at a level were a "good enough" format could be created. CD has enough bandwidth and dynamic range. The real limitation of CD is the amount of audio channels. That's what CD really lacks, support for multichannel sound, but for stereo (and mono) sound it's not lacking anything sound-quality wise.

Hi-res consumer audio however is an economical decision. It's about milking audiophools out of their money. Making people buy their favourite albums again and again. Better masterings is all they can offer, but then again those better masterings sound just as good at CD quality so there is that...


----------



## Sterling2

71 dB said:


> Frustratingly we are living the age of "alternative facts", but the real facts nevertheless exist.
> 
> In music_ production_ higher resolution offers benefits. Bigger dynamic range means you don't have to optimize the use of 16 bit dynamic range at every point and there is no fear of massive processing of sound to cumulate into audible levels of noise. Bigger bandwidth is perhaps less beneficial, but in some instances may offer something. Such as recording ultrasonic frequencies and playing them back at lower sample rate to make those frequencies audible to humans.
> 
> ...


 I cannot discern AAC from CD but I can discern the benefit of multi-channel SACD.


----------



## old tech

71 dB said:


> Technically the more bits you have the more information you have, but the question is what information is relevant for human ears? Stuff we don't hear anyway can go. I don't hear ants farting so I don't care if CD can't produce sounds that quiet. I don't hear bats chirping so I don't care if CD can't produce those ultrasonic frequencies. Within the bandwidth and dynamic range specifications, digital audio doesn't loose information. DACs don't reconstruct "lost" information. They reconstruct the analog signal from the digital information. Oversampling is a "technically smart" way to do it. Within the bandwidth and dynamic range limitations digital audio can theoretically reproduce the original signal with 100 % accuracy. Higher resolution doesn't increase accuracy (you can't go above 100 %), but they push the limits further beyond the limits of human hearing. High resolution digital sound is not better for human ears. We can't hear the effect of more bits, because 44.1/16 already gets over our limits.
> 
> Philips wasn't crazy wanting 14 bits for the CD first. As I mentioned, about 13 bits is what's needed. CD specifications were not economical decisions as at first CD players were very price, althou the prices dropped fast. The decisions were for the most part technological. Fortunately at that point in time, four decades ago, digital technology was at a level were a "good enough" format could be created. CD has enough bandwidth and dynamic range. The real limitation of CD is the amount of audio channels. That's what CD really lacks, support for multichannel sound, but for stereo (and mono) sound it's not lacking anything sound-quality wise.


Not meaning to detract from anything here as it is a good post, but my understanding is that 'technically' you don't get more information with more bits.  16 bits has as much information as 24 bits, just more noise (which is below the threshold of human hearing, hence why humans cannot hear a difference between the two).  I bit has as much information as 16 bit or 24 bits, just a hell of a lot more noise.  After all, SACDs are only 1 bit, but the massive noise with 1 bit is shifted to the ultrasonics and out of human range of hearing.

Interesting snippet with the 14 bits for CD. Back in 1982 with the first CD players, Phillips had an oversampling DAC (4x I think) while Sony's 16 bits did not. The general subjective consensus back then was that the Phillips CD player sounded better, along with the other CD player brands that used the Phillips DAC.


----------



## VNandor (Feb 1, 2021)

old tech said:


> Not meaning to detract from anything here as it is a good post, but my understanding is that 'technically' you don't get more information with more bits. 16 bits has as much information as 24 bits, just more noise (which is below the threshold of human hearing, hence why humans cannot hear a difference between the two). I bit has as much information as 16 bit or 24 bits, just a hell of a lot more noise. After all, SACDs are only 1 bit, but the massive noise with 1 bit is shifted to the ultrasonics and out of human range of hearing.


If I understand Nyquist's rule for maximum channel capacity, 24 bit will carry more information than 16 bit if the sampling rate is constant and the last 8 bits can be decoded properly (which can't be done if the system is too noisy).

I bet the 1bit SACD coding isn't using a 44kHz sample rate because that just wouldn't work.


----------



## 71 dB (Feb 1, 2021)

old tech said:


> Not meaning to detract from anything here as it is a good post, but my understanding is that 'technically' you don't get more information with more bits.  16 bits has as much information as 24 bits, just more noise (which is below the threshold of human hearing, hence why humans cannot hear a difference between the two).  I bit has as much information as 16 bit or 24 bits, just a hell of a lot more noise.  After all, SACDs are only 1 bit, but the massive noise with 1 bit is shifted to the ultrasonics and out of human range of hearing.



Well, I was speaking on theoretical point of view. In_ practise_ of course having full 24 bit worth of dynamic range is quite impossible. This only adds to the lunacy of 24 bit in consumer audio. 24 bits is more information than 16 bits. Could be text encoded. 24 bits is 50 % more text than 16 bits. Information theory doesn't care if its "noise" or not. However, for human ears its not more information, because we want music, not noise and there is also the question of what do we hear to begin with. Hopely you now understand what I mean. I simplified things a lot. I get what are after. Yes, bits just tell how much music has noise. in that sense no more information in 24 bit. Sorry, but you opened this can of worms.


----------



## 71 dB (Feb 1, 2021)

VNandor said:


> If I understand Nyquist's rule for maximum channel capacity, 24 bit will carry more information than 16 bit if the sampling rate is constant and the last 8 bits can be decoded properly (which can't be done if the system is too noisy).
> 
> I bet the 1bit SACD coding isn't using a 44kHz sample rate because that just wouldn't work.



If you could use the 24 bit information fully it would mean the noise floor would drop much much lower so that extremely quiet sounds would not be masked by noise, but there is no benefit from this, because these sounds are too quiet to be heard in any practical listening scenario. 

Yes, PCM at 1 bit doesn't work. 8 bits works as low quality audio, 16 is more than enough for high quality audio while 24 is lunacy.


----------



## old tech

VNandor said:


> If I understand Nyquist's rule for maximum channel capacity, 24 bit will carry more information than 16 bit if the sampling rate is constant and the last 8 bits can be decoded properly (which can't be done if the system is too noisy).
> 
> *I bet the 1bit SACD coding isn't using a 44kHz sample rate because that just wouldn't work*.


Of course not, an extremely wide band width is needed to shape the 1 bit noise.


----------



## old tech

71 dB said:


> Well, I was speaking on theoretical point of view. In_ practise_ of course having full 24 bit worth of dynamic range is quite impossible. This only adds to the lunacy of 24 bit in consumer audio. 24 bits is more information than 16 bits. Could be text encoded. 24 bits is 50 % more text than 16 bits. Information theory doesn't care if its "noise" or not. However, for human ears its not more information, because we want music, not noise and there is also the question of what do we hear to begin with. Hopely you now understand what I mean. I simplified things a lot. I get what are after. Yes, bits just tell how much music has noise. in that sense no more information in 24 bit. Sorry, but you opened this can of worms.


Yes I understood what you mean, I was just being an annoying pedant.


----------



## bigshot

old tech said:


> Yes I understood what you mean, I was just being an annoying pedant.



If speaking about real world, practical reality makes one an annoying pedant, then add me to that club. Do we get membership cards and a secret handshake?


----------



## niharspol

I used to buy 24 bit, 96 kHZ recordings whenever they were available. I have since stopped doing that after I realized that the noise floor for 16 bit is already lower than audible


----------



## redrol

The more important part of a recording is the actual noise that has been recorded.  Plenty of older classical recordings have a massive amount of noise as well as low frequency noise which apparently no one noticed in mastering.


----------



## Sterling2

niharspol said:


> I used to buy 24 bit, 96 kHZ recordings whenever they were available. I have since stopped doing that after I realized that the noise floor for 16 bit is already lower than audible


 You’re a smart guy.


----------



## niharspol

redrol said:


> The more important part of a recording is the actual noise that has been recorded.  Plenty of older classical recordings have a massive amount of noise as well as low frequency noise which apparently no one noticed in mastering.


Agreed. I try to choose newer recordings, but sometimes I don't have much luck. 
There's a recording of Rachmaninoff's 2nd PC that I like (Royal Philharmonic Orchestra, Rafael Orozco). It has audible noise, but I really like it otherwise. Once the piece begins, my brain forgets about the noise.


----------



## bigshot

Classical music is going to have the highest noise floor. You've got 80 people all sitting under the microphones and a concert hall with traffic outside. But I agree... Who cares once the music starts?!


----------



## Slaphead

redrol said:


> The more important part of a recording is the actual noise that has been recorded.  Plenty of *older* classical recordings have a massive amount of noise as well as low frequency noise* which apparently no one noticed in mastering.*



Or did, but realised that there was not much that could be done about it without damaging the performance, given the technology at the time.


----------



## bigshot

Well, in recordings made at Carnegie Hall, sometimes the subway train goes by underneath during the performance. Not much that can be done about that.


----------



## Slaphead

bigshot said:


> Well, in recordings made at Carnegie Hall, sometimes the subway train goes by underneath during the performance. Not much that can be done about that.



And those recordings often become very much sought after due to the added character of the performance, even if those ambient sounds are unintended.


----------



## bigshot

Toscanini!


----------



## redrol

Slaphead said:


> Or did, but realised that there was not much that could be done about it without damaging the performance, given the technology at the time.


I understand.  Interesting comment however.


----------



## theaudiologist1

bigshot said:


> Classical music is going to have the highest noise floor. You've got 80 people all sitting under the microphones and a concert hall with traffic outside. But I agree... Who cares once the music starts?!


Does that mean you need powerful amps and need to drive your headphones really loud? My headphones reach 110dB-113dB max. My headphones are high impedance and my amp is not the strongest. Will that ruin my classical experience?


----------



## sander99

bigshot said:


> Classical music is going to have the highest noise floor. You've got 80 people all sitting under the microphones and a concert hall with traffic outside. But I agree... Who cares once the music starts?!





theaudiologist1 said:


> Does that mean you need powerful amps and need to drive your headphones really loud? My headphones reach 110dB-113dB max. My headphones are high impedance and my amp is not the strongest. Will that ruin my classical experience?


No, at least not because of a high noise floor. If you listen at X dB higher SPL then also the noise will be X dB louder.


----------



## bigshot

If your amp can go up loud enough for listening the way you want, it should be fine.


----------



## skhan007

This thread has been eye-opening. I have not gotten through all the posts, but really have enjoyed reading and learning and will continue to do so. I'm seeing some conclusions being asserted by many, in quite scientific terms, which is useful for me to study. 

Please correct me if I'm wrong: To improve upon and go higher than bit/sampling of 16/44.1 would likely produce inaudible differences, if any, so sticking with lossless recordings at 16/44.1 is fine. Audiophiles that assert noteworthy or superior benefits above these values are reporting a placebo effect of sorts. Buying and paying to stream higher resolution at, let's say, 24/96 or greater does not yield a higher quality listening experience. Am I understanding correctly? I haven't read the answer to this one in the thread yet, so apologies if it's already been answered- Does a higher sampling rate above 44.1 kHz (let's say 96, 128, etc.) produce appreciable and audible improvements?

I've just purchased a new DAC that's capable of 32/768 kHz to replace my 10+ year old DAC (which has a max of 24/96), but from what I'm reading, playing my ALAC 16/44.1 music will sound great and perhaps I don't need to re-purchase my favorite albums in 24/192? If so, you all have saved me a ton of cash and I can simply continue with my CD's transferred to my MacBook as ALAC files. My new DAC is a combo DAC/headphone amp and will arrive in a day or two and looking forward to it.


----------



## 71 dB

skhan007 said:


> 1) This thread has been eye-opening. I have not gotten through all the posts, but really have enjoyed reading and learning and will continue to do so. I'm seeing some conclusions being asserted by many, in quite scientific terms, which is useful for me to study.
> 
> 2) Please correct me if I'm wrong: To improve upon and go higher than bit/sampling of 16/44.1 would likely produce inaudible differences, if any, so sticking with lossless recordings at 16/44.1 is fine.
> 
> ...


1) That's good. The purpose of these threads is to educate if possible.

2) Higher bitrates than 16/44.1 alone mean inaudible differences. In order to have audible differences something else is needed such as different mastering in which case it's not the bitrate, but the mastering causing the differences.

3) Placebo is common especially among people who don't understand digital audio properly and assume audible benefits for hi-res audio based on intuition. Another explanation is some people earn their living selling snake oil and benefit financially when this myth of hi-res consumer audio being better is upheld and spread.

4) Yes assuming those higher streams/files are from the master than 16/44.1 versions. This is because 16/44.1 is technically transparent for human ear. The sampling rate 44.1 kHz is just enough and bit depth is more than needed (13 bits would be just enough). Dither noise makes digital audio 100 % accurate (distortion free) within it's dynamic range. The dynamic range of 16 bit digital audio is technically over 90 dB and perceptually 90-120 dB depending of what kind of shaping if any is used for the dither noise. The needed dynamic range for even the most demanding listening scenarios is 70-80 dB (limited by the properties of human hearing and background noise levels real life listening environments) so it's clear 16 bits is easily enough for consumer audio. In studio, music production, a lot of safety margin is beneficial and hi-res has it's place, but in consumer audio massive safety margins are not needed anymore.

5) No. All it does is make ultrasonics possible, but humans don't hear those frequencies. Dogs, cats and bats do. Those animals claim improvements. Amplifiers and speakers are generally not designed for ultrasonics. Amplifiers can even produce distortions on those very high frequencies that are modulated on lower, audible frequencies. In this sense hi-res audio can be even worse than regular 16/44.1 that takes care sure ultrasonic content is not fed to the amplifier. 

6) Yes, you don't need to re-purchase your favorite albums in 24/192 unless those are different (much better) masterings you want. Even in that case I'd downsample those 24/192 files to 16/48 files (48 = 192/4 meaning smooth downsampling even with simple algorithms) and make the files 6 times smaller without any loss of perceptual sound quality.


----------



## skhan007

71 dB said:


> 1) That's good. The purpose of these threads is to educate if possible.
> 
> 2) Higher bitrates than 16/44.1 alone mean inaudible differences. In order to have audible differences something else is needed such as different mastering in which case it's not the bitrate, but the mastering causing the differences.
> 
> ...


I cannot thank you enough for such a thorough and thoughtful response. I greatly appreciate this guidance and believe that I've potentially circumvented a rabbit hole of purchasing streaming plans, re-purchasing my favorite recordings, and otherwise believing the hype and placebo effect. I do understand that if I find a remaster of a favorite album, I could give it a go and see if this yields a better listening experience, but overall, re-buying the recordings shouldn't be necessary if I have them as ALAC 16/44.1 files.

I'm going to also try to read up and investigate the notion that FLAC is superior to ALAC. I've read in a few places that this is true, but again, I'm curious if there's an audible difference. I have not done any A/B testing of the same song in FLAC and ALAC, but I'm sure others have and I'm curious. As a Mac user, ALAC is the most convenient as I don't have to bother with another media player.


----------



## chef8489

skhan007 said:


> This thread has been eye-opening. I have not gotten through all the posts, but really have enjoyed reading and learning and will continue to do so. I'm seeing some conclusions being asserted by many, in quite scientific terms, which is useful for me to study.
> 
> Please correct me if I'm wrong: To improve upon and go higher than bit/sampling of 16/44.1 would likely produce inaudible differences, if any, so sticking with lossless recordings at 16/44.1 is fine. Audiophiles that assert noteworthy or superior benefits above these values are reporting a placebo effect of sorts. Buying and paying to stream higher resolution at, let's say, 24/96 or greater does not yield a higher quality listening experience. Am I understanding correctly? I haven't read the answer to this one in the thread yet, so apologies if it's already been answered- Does a higher sampling rate above 44.1 kHz (let's say 96, 128, etc.) produce appreciable and audible improvements?
> 
> I've just purchased a new DAC that's capable of 32/768 kHz to replace my 10+ year old DAC (which has a max of 24/96), but from what I'm reading, playing my ALAC 16/44.1 music will sound great and perhaps I don't need to re-purchase my favorite albums in 24/192? If so, you all have saved me a ton of cash and I can simply continue with my CD's transferred to my MacBook as ALAC files. My new DAC is a combo DAC/headphone amp and will arrive in a day or two and looking forward to it.





skhan007 said:


> I cannot thank you enough for such a thorough and thoughtful response. I greatly appreciate this guidance and believe that I've potentially circumvented a rabbit hole of purchasing streaming plans, re-purchasing my favorite recordings, and otherwise believing the hype and placebo effect. I do understand that if I find a remaster of a favorite album, I could give it a go and see if this yields a better listening experience, but overall, re-buying the recordings shouldn't be necessary if I have them as ALAC 16/44.1 files.
> 
> I'm going to also try to read up and investigate the notion that FLAC is superior to ALAC. I've read in a few places that this is true, but again, I'm curious if there's an audible difference. I have not done any A/B testing of the same song in FLAC and ALAC, but I'm sure others have and I'm curious. As a Mac user, ALAC is the most convenient as I don't have to bother with another media player.


Lossless is lossless. Doesn't matter if alac, flac, and so on. Its the exact same information and will sound the same.


----------



## skhan007

chef8489 said:


> Lossless is lossless. Doesn't matter if alac, flac, and so on. Its the exact same information and will sound the same.


Thank you!! In this one thread, I've learned a ton and greatly appreciate the responses to my questions. 

It appears that my hobby just became exponentially simplified and exceedingly cost-effective. My set up will now entail: MacBook with my CD's imported as ALAC into my (soon to arrive) RME ADI-2 DAC/headphone amp, into headphones that are yet to be researched and purchased. I know the headphone topic is a rabbit hole, but I love research (it's what I do for a living), so looking forward that exploration.


----------



## chef8489 (Feb 26, 2021)

skhan007 said:


> Thank you!! In this one thread, I've learned a ton and greatly appreciate the responses to my questions.
> 
> It appears that my hobby just became exponentially simplified and exceedingly cost-effective. My set up will now entail: MacBook with my CD's imported as ALAC into my (soon to arrive) RME ADI-2 DAC/headphone amp, into headphones that are yet to be researched and purchased. I know the headphone topic is a rabbit hole, but I love research (it's what I do for a living), so looking forward that exploration.


Yep headphones, amps, dacs, and other gear is a rabbit hole. Just don't fall down the cable rabbit hole. Digital cables are digital cables. As long as info gets there it will sound the same.


----------



## skhan007

chef8489 said:


> Yep headphones, amps, days, and other gear is a rabbit hole. Just don't fall down the cable rabbit hole. Digital cables are digital cables. As long as info gets there it will sound the same.


I see in your footer, you've listed two of headphones I'm currently considering: The HD650/HD6xx and Audeze LCD2.

I will absolutely avoid the rabbit hole of cables. I've experienced and experimented in this realm with instrument cables for my guitars and tube amps. Other than capacitance in very long cables, no appreciable differences.


----------



## chef8489

I thought the lcd-2c blow the hd650 out of the water. Just more precise lows, more sub bass, fuller sound, and faster then the hd650.


----------



## castleofargh

My ears preferred the LCD2 by a good margin.
My neck preferred the HD650. 
The neck won.

 Conclusion, music is really experienced with the neck. Or maybe I'm not so good at drawing conclusions, IDK.


----------



## sonitus mirus

castleofargh said:


> My ears preferred the LCD2 by a good margin.
> My neck preferred the HD650.
> The neck won.
> 
> Conclusion, music is really experienced with the neck. Or maybe I'm not so good at drawing conclusions, IDK.



That may be how the phrase "paean in the neck" originated.


----------



## 71 dB

skhan007 said:


> I cannot thank you enough for such a thorough and thoughtful response. I greatly appreciate this guidance and believe that I've potentially circumvented a rabbit hole of purchasing streaming plans, re-purchasing my favorite recordings, and otherwise believing the hype and placebo effect. I do understand that if I find a remaster of a favorite album, I could give it a go and see if this yields a better listening experience, but overall, re-buying the recordings shouldn't be necessary if I have them as ALAC 16/44.1 files.
> 
> I'm going to also try to read up and investigate the notion that FLAC is superior to ALAC. I've read in a few places that this is true, but again, I'm curious if there's an audible difference. I have not done any A/B testing of the same song in FLAC and ALAC, but I'm sure others have and I'm curious. As a Mac user, ALAC is the most convenient as I don't have to bother with another media player.



There is no need to thank. I'm glad if I can help. FLAC and ALAC are both lossless formats meaning they have identical sound quality also identical to the original not data compressed file. Lossless means reducing redundancy so that the file size become smaller, but there is zero loss of information. ALAC is "FLAC for Mac users."


----------



## Bozzunter

71 dB said:


> 1) That's good. The purpose of these threads is to educate if possible.


There is one more thing I would like to be added to this recap about the "advantages" of high resolution and 24 bits... Since all that was explained in your post is so easily defensible on a scientific level (which does not mean lots of people refuse to believe it), streaming services and the likes claim high res has another fantastic advantage on audible frequencies: do be more detailed on audible frequencies. More samples, more resolution, better quality.

It has widely been debunked in this forum, but if someone could pop in and re-post a simple explanation, it would be fantastic.

By the way, a few days ago I was reading the forum dedicated to the fantastic Audeze LCD i4. In order to explain why a very expensive DAP/DAC/Amplifier is better than the DAC provided by Audeze itself through a Lightning cable, someone said that something like "the power of the first solution is *better* than what's provided by the latter". Better, I kid you not. You couldn't even say that delusions come cheap, because with audio they don't.


----------



## 71 dB

Bozzunter said:


> There is one more thing I would like to be added to this recap about the "advantages" of high resolution and 24 bits... Since all that was explained in your post is so easily defensible on a scientific level (which does not mean lots of people refuse to believe it), streaming services and the likes claim high res has another fantastic advantage on audible frequencies: do be more detailed on audible frequencies. More samples, more resolution, better quality.
> 
> It has widely been debunked in this forum, but *if someone could pop in and re-post a simple explanation*, it would be fantastic.



I'm afraid there just isn't a _simple_ explanation for someone who doesn't understand digital audio well. A lot of this goes against intuition and to break the misleading intuition (more is better), one has to understand there things well. These things are learnable for just about anyone (it helps if you don't suck at math), but most people just don't see the trouble of learning this stuff.

At it's simplest the explanation can be broken into pieces like this:

1. Even in most demanding music listening scenarios no more than 80 dB of dynamic range and frequency range up to 20 kHz is needed. This comes from the properties and limits of human hearing and it includes "golden ears." Pain threshold, comfortable/safe listening levels and listening environment background noise play a role in this.

2. Mathematically digital audio provides* 100 % *fidelity within the limits set by sampling frequency and bit depth. This comes from Nyquist-Shannon sampling theory and the use of Dither noise.

3. Sampling frequency of 44.1 kHz gives theoretical bandwidth of 22.05 kHz and a practical bandwidth of 20 kHz.

4. The theoretical dynamic range given by 16 bits is 98 dB without dither. Dither increases noise level a little bit, but also makes the noise more comfortable because granulation (correlation with the signal) is gone. However, this happens on levels too low to be heard at any practical listening levels. Using shaped dither the noise can be made perceptually even quieter (if inaudible is not quiet enough) so that the _perceptual_ dynamic range of 16 bit, 44.1 kHz digital audio can be even 110-120 dB!

5. Based on all of the above, 16 bit, 44.1 kHz digital audio is enough of bandwidth and dynamic range. Within this frame the fidelity is 100 % (in practice the quality of ADC and DAC etc. limit the quality, but this is the same for hi-res audio also, just as an even more limiting way).

6. Hi-res audio doesn't increase fidelity within the ranges of human hearing, because it's already theoretically 100 % with 16 bit, 44.1 kHz. Hi-res digital audio increases fidelity outside human hearing, if even that. Potentially Hi-res digital audio can even reduce fidelity if ultrasonic signal cause distortion on audible frequencies because some audio gear isn't designed for those frequencies.

7. Streaming services have an business incentive to milk audiophools offering "hi-res sound" at higher price. Placebo effect together with theoretical ignorance takes care of making these audiophools believe they are getting something extra for their money. My incentives to educate people here are the feel of moral responsibility to share understanding of something I believe I have (less selfish reason) and also to feel my life and existence matters (more selfish reason).


----------



## skhan007

The above is absolutely spot-on. While I'm not a physicist, audiologist, or otherwise an expert in sound science, I have read the research referenced by Nyquist's work and Lavry's white paper. This and the post above have been summarized quite succinctly in this article:

https://sonicscoop.com/2016/02/19/t...rates-when-higher-is-better-and-when-it-isnt/

I've been doing my own A/B testing listening to a song in 16/44.1 vs higher sampling rates (24/96, 24/192, DSD, etc.) and to my ears, hear only volume change or possibly artifacts from remastering, but no appreciable improvement from FLAC/ALAC to hi-res. It's eye-opening and I absolutely appreciate the avoidance of the rabbit hole and potential endless spending that could ensue otherwise chasing sonic attributes that are not detectable by the human ear.

This quote: "In theory, rates around 44.1kHz or 48kHz should be a near-perfect for recording and playing back music. Unless the Nyquist Theorem is ever disproved, it stands that any increase in sample rates _cannot_ increase fidelity within the audible spectrum. At all. Extra data points yield _no_ improvement."

Also, this quote: "...mathematical proof that showed any sound wave could be _perfectly_ re-created so long as it was limited in bandwidth and sampled at a rate more than twice its own frequency."

So, if the upper limit of the human ear's perception is 20 kHz, arriving at a sample rate of 44.1 kHz is more than adequate. Then there's the topic of intermodulation distortion that can happen with unnecessarily high sample rates. While I haven't heard this distortion with my own ears, the science makes sense.


----------



## bigshot

There is an article in my sig file called CD sound is all you need that is pretty useful.


----------



## theaudiologist1

skhan007 said:


> I cannot thank you enough for such a thorough and thoughtful response. I greatly appreciate this guidance and believe that I've potentially circumvented a rabbit hole of purchasing streaming plans, re-purchasing my favorite recordings, and otherwise believing the hype and placebo effect. I do understand that if I find a remaster of a favorite album, I could give it a go and see if this yields a better listening experience, but overall, re-buying the recordings shouldn't be necessary if I have them as ALAC 16/44.1 files.
> 
> I'm going to also try to read up and investigate the notion that FLAC is superior to ALAC. I've read in a few places that this is true, but again, I'm curious if there's an audible difference. I have not done any A/B testing of the same song in FLAC and ALAC, but I'm sure others have and I'm curious. As a Mac user, ALAC is the most convenient as I don't have to bother with another media player.


FLAC is superior because it's libre and not proprietary to apple.


----------



## theaudiologist1

skhan007 said:


> The above is absolutely spot-on. While I'm not a physicist, audiologist, or otherwise an expert in sound science, I have read the research referenced by Nyquist's work and Lavry's white paper. This and the post above have been summarized quite succinctly in this article:
> 
> https://sonicscoop.com/2016/02/19/t...rates-when-higher-is-better-and-when-it-isnt/
> 
> ...


Yeah 192kHz+ DOES cause distortion. 96kHz is already overkill.


----------



## Slaphead

theaudiologist1 said:


> FLAC is superior because it's libre and not proprietary to apple.


Maybe I'm just being pedantic here, but in the context of this thread FLAC and ALAC perform identically in terms of sonics - they are after all both lossless codecs.

Maybe from a technical aspect one or other of them might have a small advantage here and the other there, but being proprietary does not make a codec inferior in terms of sound quality.

In any event the argument is moot as Apple made ALAC open source and royalty free almost ten years ago.


----------



## Whazzzup




----------



## skhan007 (Mar 9, 2021)

So here's a question, that may have already been explored, but I'm curious: If certain members the audiophile community are fond of hi-res music that is reported to be sampled at rates higher than the human ear can perceive, can we see on a spectrum analyzer that there is no sound/data above 20 kHz? When I see threads that describe equipment capable of sampling and producing at higher levels, I'm perplexed. Unless I'm missing something (which is entirely possible, as I'm no audio engineer), a spectrum analyzer should be able to clearly indicate that no sound is produced or audible above a certain limit. Furthermore, what recording equipment is capturing these frequencies in the first place? I don't think studio mics are capable. If we were talking televisions, it would be ridiculous to see a product that produces light at frequencies beyond what is detectable with our eyes (e.g. infrared, ultraviolet, x-ray, etc.). Thoughts?


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## Slaphead (Mar 10, 2021)

skhan007 said:


> So here's a question, that may have already been explored, but I'm curious: If certain members the audiophile community are fond of hi-res music that is reported to be sampled at rates higher than the human ear can perceive, can we see on a spectrum analyzer that there is no sound/data above 20 kHz? When I see threads that describe equipment capable of sampling and producing at higher levels, I'm perplexed. Unless I'm missing something (which is entirely possible, as I'm no audio engineer), a spectrum analyzer should be able to clearly indicate that no sound is produced or audible above a certain limit. Furthermore, what recording equipment is capturing these frequencies in the first place? I don't think studio mics are capable. If we were talking televisions, it would be ridiculous to see a product that produces light at frequencies beyond what is detectable with our eyes (e.g. infrared, ultraviolet, x-ray, etc.). Thoughts?


Well analog magnetic tape is able to capture very high frequencies, well in excess of standard 16/44. And those frequencies can actually be transcribed onto vinyl except they don't last much longer than one or two playthroughs on that medium. The big advantage that digital has over analog mediums is dynamic range - the best studio reel to reel tape machines have around 13 bits of dynamic range, so around 78dB, and as for vinyl it tops out at around 10 bits of dynamic range (so 60dB) even for the best decks. Once you've got a medium that covers the complete human hearing frequency spectrum then dynamic range becomes king, but only to a point, going beyond around 90 dB or so is for the most part meaningless - standard redbook is 96dB without noise shaping.

And if you're talking televisions then all of them come with something that emits light beyond the visible spectrum - the remote control.


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## bigshot (Mar 10, 2021)

Here is a spectrum from a song from an SACD. Note the steep drop off at CD quality sound, a noise floor and spurious noise spikes... all of which are inaudible because of the volume as well as the frequency.



Slaphead said:


> Well analog magnetic tape is able to capture very high frequencies, well in excess of standard 16/44.



That is not true. At 15ips/24 track/2 inch, the format most late period analogue recordings were made on, most commercial analogue recordings just barely come up to 20kHz. They weren't designed to accurately record beyond that. http://www.endino.com/graphs/ None of them match the specs of 16/44.1 for response, noise and distortion. https://en.wikipedia.org/wiki/Comparison_of_analog_and_digital_recording

It was standard practice to apply a low pass filter to remove all frequencies above 20kHz. They weren't wanted.


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## old tech

bigshot said:


> Here is a spectrum from a song from an SACD. Note the steep drop off at CD quality sound, a noise floor and spurious noise spikes... all of which are inaudible because of the volume as well as the frequency.
> 
> 
> 
> ...


It also depends on the recorder.  For example, the 8 track recorder the Beatles used for the White Album had a frequency response of 30 - 15 khz +/- 2db.  That was a cutting edge recorder back in 1968. Yet some Beatles fans worry about frequency extension beyond 20khz, go figure that one out.

As for dynamic range, while LP records have a range equivalent to around 10 - 13 bits digital, it is not that important for most pop/rock music as these recordings rarely have a range of more than 50db.  It is mainly with classical music, particularly full orchestral productions, that the 96 db + advantages of CDs and SACDs become more apparent.


----------



## bigshot

All correct. The most dynamic LPs don't have more that about 45-50 dB dynamic range. And if they get up that high, they get very hard to listen to. You keep having to jump up to adjust the volume.

People tend to think "more is better". That just isn't true. Enough is better.


----------



## theaudiologist1

bigshot said:


> All correct. The most dynamic LPs don't have more that about 45-50 dB dynamic range. And if they get up that high, they get very hard to listen to. You keep having to jump up to adjust the volume.
> 
> People tend to think "more is better". That just isn't true. Enough is better.


Agreed. It's annoying to adjust the volumes on my classical collection.


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## bigshot

I've found BIS is the worst label for excessive dynamics.


----------



## Slaphead

bigshot said:


> That is not true. At 15ips/24 track/2 inch, the format most late period analogue recordings were made on, most commercial analogue recordings just barely come up to 20kHz. They weren't designed to accurately record beyond that. http://www.endino.com/graphs/ None of them match the specs of 16/44.1 for response, noise and distortion. https://en.wikipedia.org/wiki/Comparison_of_analog_and_digital_recording
> 
> It was standard practice to apply a low pass filter to remove all frequencies above 20kHz. They weren't wanted.


Ok, yes you're right for standard pro audio recording equipment at the time, However I was more thinking of what we were doing back in university in the 80's when we needed to record some ultrasonic frequencies, think 50 - 60 Khz for some other department that was studying aerodynamic flutter on a micro scale. We eventually used a jury rigged VHS VCR with a bit of a turbo on the tape speed - I think around 8x the normal video recording speed, maybe less, and could get a solid 50Khz, and higher with a bit of fiddling.

I often wonder why analog audio recording never adopted the helical scanning technique used in video recorders and then later for DAT (as well as a backup medium for computers) as it provided for a far greater dynamic range and frequency response.


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## theaudiologist1 (Mar 11, 2021)

Is DSD a horrible outdated meme format, considering it only has 2 quantization levels and 6dB of dynamic range? People claim DSD64 is garbage and noisy and that DSD is only good at 128fs+, but I couldn't tell a difference.


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## chef8489

theaudiologist1 said:


> Is DSD a horrible outdated meme format, considering it only has 2 quantization levels and 6dB of dynamic range? People claim DSD64 is garbage and noisy and that DSD is only good at 128fs+


There pretty much is no such thing as pure dsd. Just about every dsd album was converted to pcn at one stage or multiple stages.


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## theaudiologist1 (Mar 11, 2021)

chef8489 said:


> There pretty much is no such thing as pure dsd. Just about every dsd album was converted to pcn at one stage or multiple stages.


Pretty much. That's why it's a meme format. In theory I liked DSD more than PCM, but what's the point if DSD doesn't even exist? It's like trying to promote a video format that's 32K but it has to go through millions of processes of being converted to 4K or 1080p and THEN upsmapled to 32K again for mastering since there is no way no master 32K video.


----------



## sander99

Slaphead said:


> I often wonder why analog audio recording never adopted the helical scanning technique used in video recorders and then later for DAT (as well as a backup medium for computers) as it provided for a far greater dynamic range and frequency response.


"HiFi Video" recorders did. They recorded frequency modulated audio with rotating heads. But at that time digital audio was already upcoming.


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## bigshot

16/44.1 is all human ears can hear. Beyond that you’re recording for bats.


----------



## theaudiologist1

bigshot said:


> 16/44.1 is all human ears can hear. Beyond that you’re recording for bats.


I know DSD doen't sound better, but I'm actually wondering if it sounds worse due to the noise in the higher frequencies.


----------



## bigshot

I did a listening test comparing direct DSD to 16/44.1 and couldn’t discern any differences. If you’d like to try that yourself, I can tell you how to set that up.


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## theaudiologist1 (Mar 11, 2021)

bigshot said:


> I did a listening test comparing direct DSD to 16/44.1 and couldn’t discern any differences. If you’d like to try that yourself, I can tell you how to set that up.


I already tried that. To me and my DAC, PCM and DSD sound different and DSD is always quieter, but not necessarily better. Both are equal quality-wise.


----------



## old tech

theaudiologist1 said:


> I already tried that. To me and my DAC, PCM and DSD sound different and DSD is always quieter, but not necessarily better. Both are equal quality-wise.


Do you try level matching, ie either raising the volume of the DSD or lowering the volume of PCM so they are equal and then blind testing them?  All the controlled tests I've seen don't find a difference in normal listening.


----------



## Don Hills

sander99 said:


> "HiFi Video" recorders did. They recorded frequency modulated audio with rotating heads. But at that time digital audio was already upcoming.



They also had to use a DBX style compression / noise reduction system, because of the noise introduced by the head switching at the end of each scan. It wasn't a problem for video because the head switch was hidden in the flyback for each frame. Most decks had a tracking control knob, if you maladjusted it you could often hear a "purring" noise as the signal dropped out of the FM limiting.


----------



## theaudiologist1 (Mar 11, 2021)

old tech said:


> Do you try level matching, ie either raising the volume of the DSD or lowering the volume of PCM so they are equal and then blind testing them?  All the controlled tests I've seen don't find a difference in normal listening.


I did. I increased the volume of my AMP so that the DSD files are the same volume as the PCM files. Keep in mind even PCM files upsampled to DSD have a lower volume (I gave it a -6dB gain). Overall the DSD sounds flatter and more "analog" while the PCM has more "oomph" and the instruments are a bit more "in your face". I did this test with Audirvana.


----------



## bigshot (Mar 11, 2021)

theaudiologist1 said:


> I already tried that. To me and my DAC, PCM and DSD sound different and DSD is always quieter, but not necessarily better. Both are equal quality-wise.



Equal quality is equal. You can always turn up the volume a hair. If you properly level match, all differences will disappear. High bitrate lossy, CD quality sound, HD Audio and DSD are all capable of being audibly transparent. To human ears, they should all sound the same unless you're doing something wrong.

There are four components to a controlled listening test. I did all three of these when I compared DSD to 16/44.1:

Level matching: Both samples were set to an identical level. Humans tend to perceive slightly louder sounds as sounding better, even when the louder sound is the exact same quality. Even differences as little as a half a dB can make a difference. You can't just ride the volume control. It has to be calibrated carefully.

Direct A/B switched: Human auditory memory is very short. For similar sounds it can be as short as a couple of seconds. If you want to compare similar sounds, they need to be racked up through a switch box, playing the exact same music in sync so you can just flip a switch back and forth and directly compare them.

Blind Testing: Expectation bias is real and it is unconscious. You can't will it away. The way to eliminate it is to focus entirely on the sound and remove any other identifying information about the samples.

Multiple Tests: There is always an element of dumb luck or chance involved in picking one sample or another. To eliminate that, you do multiple rounds of test and average them to see if it reaches the level where it can't be blind chance.

Do all four of these, and you will know to a high degree of certainty what the truth is. I've done this. I know. That is step one. Step two is making your results repeatable for other people. Take care of step one and then we can talk about step two.


----------



## 118900

bigshot said:


> Equal quality is equal. You can always turn up the volume a hair. If you properly level match, all differences will disappear. High bitrate lossy, CD quality sound, HD Audio and DSD are all capable of being audibly transparent. To human ears, they should all sound the same unless you're doing something wrong.
> 
> There are four components to a controlled listening test. I did all three of these when I compared DSD to 16/44.1:
> 
> ...


I saw an interesting video where using an app someone played two files on opposite phases. If they cancel each other out and you hear just silence or hiss they are the same. Can’t remember the name of the app but he used it to show that 24 and 16 bit recordings are identical


----------



## bigshot

That is called a null test. It's a great way to test to see if things are the same.


----------



## old tech

juansan said:


> I saw an interesting video where using an app someone played two files on opposite phases. If they cancel each other out and you hear just silence or hiss they are the same. Can’t remember the name of the app but he used it to show that 24 and 16 bit recordings are identical


This one you mean?


----------



## 118900

old tech said:


> This one you mean?



That’s it!


----------



## Sebasistan

juansan said:


> That’s it!


Is the wrong takeaway from this to say "well, he does prove that higher bit depths have the same "amount" of music in them, BUT he also proves higher bit depths ARE superior because they have less noise in comparison!"?


----------



## 118900 (Mar 12, 2021)

Sebasistan said:


> Is the wrong takeaway from this to say "well, he does prove that higher bit depths have the same "amount" of music in them, BUT he also proves higher bit depths ARE superior because they have less noise in comparison!"?


The only thing I took away was what I originally stated, ie that by cancelling each other out when played back in opposing phases proves that no additional info is there. Not sure what else he tried to imply.


----------



## Sebasistan

juansan said:


> The only thing I took away was what I originally stated, ie that by cancelling each other out when played back in opposing phases proves that no additional info is there. Not sure what else he tried to imply.


I also wonder if this can be applied to cd vs. vinyl vs. mp3.


----------



## VNandor

Sebasistan said:


> I also wonder if this can be applied to cd vs. vinyl vs. mp3.


It could be easily applied to cd (lossless) vs. mp3. However this test is meant to check if the compared signals are different and since pretty much everyone knows lossy has to be different compared to lossless (that's why it is called lossy in the first place) the test wouldn't reveal anything that we didn't know in the first place. CD vs. vinyl would be harder to do because it would require some actual hardware that can do the phase inversion, gain match, and summing in the analog domain.


----------



## bigshot

A null test wouldn't be good for testing lossy vs lossless. You know lossy is missing information, and that is going to show up. However due to masking and other psychoacoustic techniques, the material that is missing isn't audible. So even though it would show up in a null test, it probably wouldn't make any difference in normal listening.

CD vs vinyl would be difficult, if not impossible to null because vinyl has so much speed fluctuation.

Edit: Oops! I just found out that Vnandor said the exact same thing. Bread and butter!


----------



## Slaphead (Mar 12, 2021)

theaudiologist1 said:


> Is DSD a horrible outdated meme format, considering it only has 2 quantization levels and 6dB of dynamic range? People claim DSD64 is garbage and noisy and that DSD is only good at 128fs+, but I couldn't tell a difference.


I guess you missed the information theory class then - apart from R2R DACs, pretty much all DACs use a delta-sigma approach, which is to say that they take the PCM and convert it to what is essentially the underpinnings of DSD - an averaging of the 1s and 0s to produce the required amplitude in the signal. Oversampling is what it is.

Now I'm no fan of DSD, as I think it's been marketed as something it's not. However as a digital archive medium it's bloody wonderful. I could program something to pepper a DSD file with 10% of random flipped bits and you probably wouldn't notice the corruption in playback. Do that to a PCM file and it would be unlistenable.

There's nothing really wrong with DSD, it's just that what it is isn't really intended for playback despite what the purveyors of DSD would have you believe.


----------



## theaudiologist1

Slaphead said:


> I guess you missed the information theory class then - apart from R2R DACs, pretty much all DACs use a delta-sigma approach, which is to say that they take the PCM and convert it to what is essentially the underpinnings of DSD - an averaging of the 1s and 0s to produce the required amplitude in the signal. Oversampling is what it is.
> 
> Now I'm no fan of DSD, as I think it's been marketed as something it's not. However as a digital archive medium it's bloody wonderful. I could program something to pepper a DSD file with 10% of random flipped bits and you probably wouldn't notice the corruption in playback. Do that to a PCM file and it would be unlistenable.
> 
> There's nothing really wrong with DSD, it's just that what it is isn't really intended for playback despite what the purveyors of DSD would have you believe.


I actually knew the delta-sigma convertion and that all DAC's are basically DSD DAC's which is why I converted my SACDs to DSD and also upsample my PCM to DSD over the air so the DAC doesn't do it.

I heart DSD in SACDs was originally meant to be 4bits at 5.6MHz. If that happened, perhaps it would have been better.


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## bigshot (Mar 13, 2021)

There is no audible difference in sound quality between CDs and SACDs other than the fact that SACDs support multichannel. If you only listen to stereo music DSD is as pointless as teats on a bull hog. It is more complicated and takes up more space for no audible benefit at all.

There is no "better" than audibly transparent. You can pack more frequencies and ultra low volume detail into the sound, but your ears won't be able to hear it. Once you achieve transparency, you can quit. Nothing else you can do will improve the sound, except for listening to better music.

Today, formats have become irrelevant. CD, HD Audio, SACD, DSD, high bitrate MP3 or AAC... It all sounds exactly the same.


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## 71 dB

bigshot said:


> Today, formats have become irrelevant.


That's why I am a fan of good music*, good production, good mixing and good mastering. The relevant stuff...

* Meaning music I _personally_ like and enjoy, but might be crappy music for someone else.


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## ADUHF (Mar 14, 2021)

bigshot said:


> There is no "better" than audibly transparent. You can pack more frequencies and ultra low volume detail into the sound, but your ears won't be able to hear it.



There may be some extreme instances where you can hear the difference between 16 and 24-bit, with some training. Amir of Audio Science Review was apparently able to foil some of the 24-bit vs. 16-bit ABX tests by turning up the volume and carefully scrutinizing some of the quieter musical passages, for example. He explains how in this recent video...


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## ADUHF (Mar 14, 2021)

Another instance where I think it could possibly make a difference is when making adjustments to the content's dynamic range with digital controls. With the volume control on a computer, or using a software-based digital EQ, for example. Increasing the bit depth on my audio device from 16 to 24-bit seems to help with the resampling of the levels in these cases. YMMV though.

If you are not making any digital alterations to the content, and simply listening to it "as is", then there may be no benefit to this.

If you are listening to 24-bit content though, then you probably want to maintain it at that bit-depth, rather than downsampling to 16-bit, which could be potentially lossy.


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## ADUHF (Mar 14, 2021)

Someone was also asking earlier about whether the sample rates made any difference in the high frequency detail that you can hear. And I think the general consensus on that was that most recorded content does not generally go above 20k.

Some humans (ie not me) can hear above that frequency though. And maybe that's why video went for the somewhat wider range offered by 48 kHz.



> Under ideal laboratory conditions, humans can hear sound as low as 12 Hz[11] and as high as 28 kHz, though the threshold increases sharply at 15 kHz in adults, corresponding to the last auditory channel of the cochlea.[12]



https://en.wikipedia.org/wiki/Hearing_range#Humans

The highest note that has supposedly ever been sung is G10 at 25.1 kHz. Most music won't go quite that high though.  For comparison, a typical soprano part rarely goes above "soprano C", which is C6 at a mere 1.0 kHz! Falsetto and coloratura singers can go higher than this though.

Most of the info above a certain frequency in the treble range will generally be overtones and timbral info, I believe, as opposed to actual "notes".


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## bigshot (Mar 14, 2021)

I don't believe that training can allow you to hear frequencies above 20kHz nor noise floors below -90dB. "Training" is just a way to try to get people to not trust their own ears and take your word for it. It's a power play that has nothing to do with perceptual ability.



ADUHF said:


> There may be some extreme instances where you can hear the difference between 16 and 24-bit, with some training. Amir of Audio Science Review was apparently able to foil some of the 24-bit vs. 16-bit ABX tests by turning up the volume and carefully scrutinizing some of the quieter musical passages, for example. He explains how in this recent video...



I try not to comment on this guy, but when he made the claim to me that he could hear the difference between 16 and 24 bit, I called him on goosing the volume and he refused to admit to me that he was doing gain riding. Cranking the volume on quiet parts isn't some unique skill that takes training. Sure you can hear something down below -90dB if you pump up the volume way up on a fade out. That doesn't take training. That is just plain cheating. No one listens to music riding the volume level to pull up all the quiet parts. Parlor tricks.



ADUHF said:


> Some humans (ie not me) can hear above that frequency though. And maybe that's why video went for the somewhat wider range offered by 48 kHz.



I believe the person who could hear the highest was a child who could hear up to 23-24kHz. That isn't even one whole note on the musical scale above 20kHz- a tiny sliver of sound. By the time the kid was a teenager, she couldn't hear that high any more. No one can hear frequencies as musical pitch much above 10kHz or so. It becomes an undifferentiated squeal and then you just feel sound pressure. If you crank a 30kHz tone loud enough, you can feel it. That isn't hearing.

All this stuff about super audible frequencies is bologna. 16/44.1 is already overkill. The only reason higher data rates are needed is for mixing and mastering. None of the exceptions you mention have any relevance to listening to music in the home. The difference is inaudible in the real world outside the heads of audio "experts".

Video chose 48kHz because it divided evenly with the frame rate of film. It made film to video conversions less difficult.


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## ADUHF (Mar 14, 2021)

bigshot said:


> I don't believe that training can allow you to hear frequencies above 20kHz nor noise floors below -90dB.



I tend to agree that this is probably the case in normal listening conditions. It would be interesting to see if there has been any testing on this though.



bigshot said:


> "Training" is just a way to try to get people to not trust their own ears and take your word for it. It's a power play that has nothing to do with perceptual ability.



Perhaps not surprisingly, I would disagree with this. At least when it comes to detecting things like audible noise and distortion, and some types of compression, resampling or aliasing artifacts. I can't give you any demonstrable proof of this though, beyond my own opinion. So my assertion is basically meaningless from a scientific standpoint. 



bigshot said:


> I try not to comment on this guy, but when he made the claim to me that he could hear the difference between 16 and 24 bit, I called him on goosing the volume and he refused to admit to me that he was doing gain riding. Cranking the volume on quiet parts isn't some unique skill that takes training. Sure you can hear something down below -90dB if you pump up the volume way up on a fade out. That doesn't take training. That is just plain cheating. No one listens to music riding the volume level to pull up all the quiet parts. Parlor tricks.



Fwiw, he acknowledges much of this in the above video. And it's why I gave it the label "extreme instances". Because this is certainly not the way most people listen to music.

If he were arguing the point with you though, then I think he might point out that "totally transparent in all listening conditions" and "transparent under normal listening conditions" may be two different things. And if you want the former rather than the latter, then maybe some of these smaller details may be worth paying attention to. It's just bits after all, so it's not like a few more is gonna hurt anything.  And maybe, in some rare or extreme circumstances, it might actually help a little.

If Amir can pick out the differences in quieter recordings at a higher volume, then maybe... just maybe, it's also a possibility for others to detect some noise or quantization errors in some lower volume 16-bit recordings as well. (?)



bigshot said:


> I believe the person who could hear the highest was a child who could hear up to 23kHz. That isn't even one whole note on the musical scale above 20kHz.



...Which is why I mentioned that sound in that range is mostly overtones and timbral in nature. I certainly can't hear that high though. It would be interesting to see if there have been more studies on this as well though. Perhaps there are certain demographic groups with slightly wider or higher ranges of hearing than others, for example.

Some also claim that they can "feel" the higher frequencies as well. Which is certainly the case with frequencies below 20 Hz. I dunno about the higher frequencies though.



bigshot said:


> All this stuff about super audible frequencies is bologna. 16/44.1 is already overkill. The only reason higher data rates are needed is for mixing and mastering. None of the exceptions you mention have any relevance to listening to music in the home. The difference is inaudible in the real world outside the heads of audio "experts".



I disagree somewhat with you on this as well. Mainly because of the proliferation of higher-quality tools, digital tools that is, for manipulating and altering audio content more to preference on the user's end. In days of yore, this kind of thing was done mostly in the analog domain.

If higher bit depths and sample rates are beneficial for mixing and mastering content. Then why not also for making similar kinds of adjustments in the digital domain on the user's end? That is my only point (I think).



bigshot said:


> Video chose 48kHz because it divided evenly with the frame rate of film. It made film to video conversions less difficult.



Yes! Now that you mention it. I seem to recall reading that somewhere as well. Thank you for the correction on that.


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## VNandor (Mar 14, 2021)

bigshot said:


> I try not to comment on this guy, but when he made the claim to me that he could hear the difference between 16 and 24 bit, I called him on it and he refused to admit to me that he was doing gain riding. Cranking the volume on quiet parts isn't some unique skill. Sure you can hear something down below -90dB if you pump up the volume way up on a fade out. That doesn't take training. That is just plain cheating. No one listens to music riding the volume level to pull up all the quiet parts. Parlor tricks.


In his video (around the 19:00-19:40 mark), he said that's exactly what he was doing to pass the test so I don't know why he wouldn't admit that to you.



ADUHF said:


> Another instance where I think it could possibly make a difference is when making adjustments to the content's dynamic range with digital controls. With the volume control on a computer, or using a software-based digital EQ, for example. Increasing the bit depth on my audio device from 16 to 24-bit seems to help with the resampling of the levels in these cases. YMMV though.


Only extremely naive implementations of DSP would make the signal lose bits (unless that was the job of the DSP in the first place). I'm not sure what's the deal with EQ APO but I don't think receiving 24bit samples instead of 16bit samples should help it.

Let's say all the processing that is done to a signal is first reducing the gain by 6dB and then increasing it right back by 6dB.
Reducing the gain by 6dB corresponds to dividing the samples by 2 (12dB would mean dividing by 4 and so on). Dividing binary numbers by 2 is very easy. You just have to shift all the 1-s and 0-s to the "right" (towards the least significant bit).

So for example dividing  some 8 bit number by two looks like this: 00110110 (54) divided by 2 is 00011011 (27). Everything just got shifted.
Multiplying is also easy, except the direction of the shift changes: 00110110 multplied by 2 is 01101100 (108).
With that in mind it can be seen how a big gain reduction could make the signal lose bits as the last bits are getting shifted out.

One solution could be just tacking on some zeros to the original number, for example if the input is a bunch of 16bit numbers just add eight 0s to the end of it and that will ensure that the bits won't get shifted out, they just get shifted into the last couple of zeros after the gain reduction. An extra 8bit could allow for 48dB reduction before the signal starts to lose bits. While that might sound good enough, I'm pretty sure most plugins do their calculations on either 32bit float or 64bit floating point numbers which work quite a bit differently but let's just say they'll have enough dynamic range for any kind of sensible processing. If EQ APO doesn't use floats then I can wrap my head around why 24bit might be better for it than 16bits.


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## sander99 (Mar 14, 2021)

ADUHF said:


> If higher bit depths and sample rates are beneficial for mixing and mastering content. Then why not also for making similar kinds of adjustments in the digital domain on the users end? That is my only point (I think).


Yes it can be beneficial to use more bits during digital signal processing. But for most processing that users would want to do it would be no problem to start with a 16 bit signal, and do the processing in more bits. (That way accumulation of rounding errors that could faul up a few of the least significant bits won't touch the 16 most significant bits.)
[Edit: I didn't see @VNandor 's last post, but he basically says the same in other words.]


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## ADUHF (Mar 14, 2021)

VNandor said:


> In his video (around the 19:00-19:40 mark), he said that's exactly what he was doing to pass the test so I don't know why he wouldn't admit that to you.
> 
> 
> Only extremely naive implementations of DSP would make the signal lose bits (unless that was the job of the DSP in the first place). I'm not sure what's the deal with EQ APO but I don't think receiving 24bit samples instead of 16bit samples should help it.
> ...



I think I followed some of that, VNandor. 

As far as EQ APO is concerned, some of it (maybe most of it) may be due more to user error on my part. Because I didn't seem to have my device configured to the same bit/sample rate as the content to begin with. And was likely down-converting everything from 24/48 to 16/44.1, as well as making the volume adjustments on top of that. So there may have been a combination of factors that made the difference between the audio device settings more noticeable.

I'm doing this on a somewhat older PC as well, that may not have the most up-to-date algorithms and what have you as well. So that could also be a factor.

There did seem to be a slight (but noticeable imho) difference though in the clarity and smoothness of the audio after switching from 16/44.1 to 24/48. Maybe more than slight, in fact.

I don't know whether EQ APO is using floating or fixed point integers for resampling. But I believe the coding is open source, if you want to have a look. I am not a programmer. But if I were coding the app, I think I might use whichever method had the least latency, which would probably be the latter.


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## ADUHF (Mar 14, 2021)

sander99 said:


> Yes it can be beneficial to use more bits during digital signal processing. But for most processing that users would want to do it would be no problem to start with a 16 bit signal, and do the processing in more bits. (That way accumulation of rounding errors that could faul up a few of the least significant bits won't touch the 16 most significant bits.)
> [Edit: I didn't see @VNandor 's last post, but he basically says the same in other words.]



So basically as long as you're going from 16 to 24, and not the other way round, you should be ok then?

I wonder also about PC utilities for expanding dynamic range by undoing the compression used to boost loudness. And whether those could also produce some noticeable artifacts when going from 16 to 16 bits. If you're starting with content that has been brick-walled into an extremely narrow dynamic range, and expanding/resampling that out into a much wider dynamic range, then it would seem as though more bits might be beneficial for something like. I don't really know though.

Although I have not actually tried it, EQ APO also includes a filter for doing this kind of thing (as mentioned recently in another topic here)...


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## sander99

ADUHF said:


> So basically as long as you're going from 16 to 24, and not the other way round, you should be ok then?


It also wouldn't be a problem to downconvert the end result back to 16 bits as a last step of the processing.


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## bigshot (Mar 15, 2021)

VNandor said:


> In his video (around the 19:00-19:40 mark), he said that's exactly what he was doing to pass the test so I don't know why he wouldn't admit that to you.



You tell me. He was arguing here that because of his training, he could hear the difference between 16 and 24. He posted Foobar results to prove it. I asked him if he had changed the volume levels as he listened and he talked around my question, so I got more pointed and asked him if he gain rode the fade outs. He dodged it over and over and then got mad and went back to his own forum where he could delete the pesky questions like mine.

I think he was trying to impress us with his superhuman ears and when I pointed out that a noise floor below -90dB wasn't a noise floor under -90dB if you turn the volume up by +50dB, he didn't want to be challenged on it and just refused to answer. That guy has an agenda.

ADUHF, I said that the only purpose for 24 bit is to provide a deeper noise floor for sound processing in mixing and mastering. When you sit in your living room listening to Beethoven on a commercially recorded CD, there is absolutely no reason for it. Consumer sound processing doesn't get anywhere near making any difference either. 12 bit is sufficient for listening to music in the home. 16 bit is overkill. There is plenty of headroom there.

The only way to understand what is important and what isn't is to take some music tracks and run them through different kinds of degradation and see the effect in real world applications. Ethan Winer does that in the videos in my sig file. He takes a horrible buzzing noise and mixes it into music and drops it -10dB at a time. Take a guess where you can't hear it any more under the music... I think you will be very surprised. You can download his files and listen to them for yourself. 

Without actually listening, specs are just abstract numbers on a page. Better numbers are better sound, right? Not always. To understand them with perspective, you need to translate those numbers to actual sound in a real world application. Then you know what -1dB sounds like as opposed to -10dB or -100dB. Numbers represent sound, but not always in an intuitive way. More is not always better. There is such a thing as good enough for human ears.

Dynamic expansion doesn't produce artifacts because of bit depth changes. It creates artifacts because there are many ways for the sound engineer who mixed the track to compress music. You have multiple variables, different ways to compress and different elements in the mix that can be compressed individually. Uncompressing it is like using a key to unlock a lock. If you don't know the exact kind of compression that was applied in the exact amount on the exact track at the exact point in the timeline, you can never uncompress it properly. You can only take a stab at it in one dimension across the whole track. The more you expand, the more artifacts you are going to get.


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## 71 dB

ADUHF said:


> The highest note that has supposedly ever been sung is G10 at 25.1 kHz. Most music won't go quite that high though.  For comparison, a typical soprano part rarely goes above "soprano C", which is C6 at a mere 1.0 kHz! Falsetto and coloratura singers can go higher than this though.
> 
> Most of the info above a certain frequency in the treble range will generally be overtones and timbral info, I believe, as opposed to actual "notes".


You must mean Geogia Brown? 



I recorded her high singing and checked it out in Audacity. Her singing goes up to about 3.8 kHz, which is impressive (not the kind of "brown note" we usually talk about), but far from this ridiculous G10 claim. Maybe they mean significant harmonic overtones go up to 25.1 kHz? I used linear frequency scale, because here it works nicely.


----------



## sonitus mirus

bigshot said:


> You tell me. He was arguing here that because of his training, he could hear the difference between 16 and 24. He posted Foobar results to prove it. I asked him if he had changed the volume levels as he listened and he talked around my question, so I got more pointed and asked him if he gain rode the fade outs. He dodged it over and over and then got mad and went back to his own forum where he could delete the pesky questions like mine.
> 
> I think he was trying to impress us with his superhuman ears and when I pointed out that a noise floor below -90dB wasn't a noise floor under -90dB if you turn the volume up by +50dB, he didn't want to be challenged on it and just refused to answer. That guy has an agenda.
> 
> ...



I don't think Amir's website was nearly as popular back then as it is today.  He was still attempting to draw in an audience and memberships.  I do recall parts of these discussions here and also over at Hydrogen Audio.  The testing method was pathological in that the volume level was so high on a quiet part of the music that if the music were to be played in its entirety, there would be temporary hearing loss or audible threshold shift that would make it impossible to hear without resting for a short period of time in a quiet environment.  

It wasn't only that the volume was increased significantly on a quiet section of some music, but that the test was only being done on a small section of the music.  His logs were showing rapid switching back and forth over a 1-2 second part or the music.   Once it was admitted that it would probably not be possible to identify any difference under normal listening conditions, the rest was just the typical back and forth exchanges where something was conflated and applied to something unrelated.


----------



## 71 dB

sonitus mirus said:


> The testing method was pathological in that the volume level was so high on a quiet part of the music that if the music were to be played in its entirety, there would be temporary hearing loss or audible threshold shift that would make it impossible to hear without resting for a short period of time in a quiet environment.
> 
> It wasn't only that the volume was increased significantly on a quiet section of some music, but that the test was only being done on a small section of the music.  His logs were showing rapid switching back and forth over a 1-2 second part or the music.   Once it was admitted that it would probably not be possible to identify any difference under normal listening conditions, the rest was just the typical back and forth exchanges where something was conflated and applied to something unrelated.


If you use a powerful magnifying glass on 8K video picture you can clearly see the individual pixels and incorrectly conclude 16K video format is needed in normal watching. The size of the magnifying glass you need tells you how much overkill/safety margin you have. In that sense these extreme tests are insightful, if you can interpret them correctly.

Why can't people just enjoy music instead of coming up with ways contradicting all practical listening scenarios to reveal "weaknesses" of 44.1 kHz/16 bit? Well, I know why. Money. Money becomes a monster if we allow it to become one.


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## bigshot

sonitus mirus said:


> Once it was admitted that it would probably not be possible to identify any difference under normal listening conditions, the rest was just the typical back and forth exchanges where something was conflated and applied to something unrelated.



He never admitted that to me. I specifically asked him if he was looping quiet sections and gain riding and he refused to admit it, saying his “training” was responsible for his ability to hear the difference.


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## Davesrose

71 dB said:


> If you use a powerful magnifying glass on 8K video picture you can clearly see the individual pixels and incorrectly conclude 16K video format is needed in normal watching. The size of the magnifying glass you need tells you how much overkill/safety margin you have. In that sense these extreme tests are insightful, if you can interpret them correctly.
> 
> Why can't people just enjoy music instead of coming up with ways contradicting all practical listening scenarios to reveal "weaknesses" of 44.1 kHz/16 bit? Well, I know why. Money. Money becomes a monster if we allow it to become one.


Video producer/nerd here....so I feel obligated to pull in realities of video standards.  4K is going to be the only video source for awhile.  Most all movies have been done 35mm (restored in 4K in the last 15 years with popular movies) and only a few 70mm (restored 8K).  The jump in 4K/UHD wasn't just resolution, but better color space/dynamic range.  For me, this is the greatest jump.  Dolby Vision grading is really a step up for a TV that supports it.  There is no consumer 8K video standard (there are 8K out displays for games), and I'm really dubious that it's going to be a standard for so many years (it's only in the last couple years that Hollywood has started producing movies with a 4K digital intermediate).  Also, when it comes to detail with a TV, a lot has to do with pixels.  I have seen comparison videos of 4K OLED vs 8K QLED, and the OLED still wins for perceived detail because each pixel is more defined. (also, maybe there's no native 8K with a distributable format).


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## bigshot

Photo analogies don’t always relate well to sound. Neither do cars and wine.


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## 118900

bigshot said:


> Photo analogies don’t always relate well to sound. Neither do cars and wine.


Even worse, people that equate cars to women. Unbelievable.


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## 71 dB

bigshot said:


> Photo analogies don’t always relate well to sound. Neither do cars and wine.


Yeah, they don't, but I used it to have magnifying glass as a visual analogy to upping volume at quiet parts...

--------------------------------------

About 8K video: 2K Blu-rays (when done decently) are good enough for me. I don't have 4K gear. For me the problem is availability of movies on physical media, not the resolution or color spaces.


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## ADUHF (Mar 18, 2021)

Davesrose said:


> Video producer/nerd here....so I feel obligated to pull in realities of video standards.  4K is going to be the only video source for awhile.  Most all movies have been done 35mm (restored in 4K in the last 15 years with popular movies) and only a few 70mm (restored 8K).  The jump in 4K/UHD wasn't just resolution, but better color space/dynamic range.  For me, this is the greatest jump.  Dolby Vision grading is really a step up for a TV that supports it.  There is no consumer 8K video standard (there are 8K out displays for games), and I'm really dubious that it's going to be a standard for so many years (it's only in the last couple years that Hollywood has started producing movies with a 4K digital intermediate).  Also, when it comes to detail with a TV, a lot has to do with pixels.  I have seen comparison videos of 4K OLED vs 8K QLED, and the OLED still wins for perceived detail because each pixel is more defined. (also, maybe there's no native 8K with a distributable format).



8k is included in the Rec. 2020 UHD spec. And there are already streaming services which offer some 8k content, including YouTube. Physical media seems a bit unlikely though for the near term. (If it picks up overseas though, who knows.)

4k UHD hit its peak for player sales a few years ago. And has been declining since then. And better compression would also be needed to squeeze the 8k files onto current UHD discs.



Davesrose said:


> Also, when it comes to detail with a TV, a lot has to do with pixels.  I have seen comparison videos of 4K OLED vs 8K QLED, and the OLED still wins for perceived detail because each pixel is more defined.



Folks said the same about plasma vs. LCD. OLED, which is emissive like plasma, has superior contrast ratio and angle of view to most LCD techs that I've seen though. So it would be my choice for those reasons. I believe burn-in and uneven wear is still a little more of an issue with OLED than with LCD though. So that might be another factor to consider. I'm not sure which has the superior color gamut, QLED or OLED. Contrast is usually king though when it comes to these types of video display tests.

QLED is a form of LCD, btw, for those who don't know.


----------



## ADUHF

71 dB said:


> You must mean Geogia Brown?
> 
> 
> 
> I recorded her high singing and checked it out in Audacity. Her singing goes up to about 3.8 kHz, which is impressive (not the kind of "brown note" we usually talk about), but far from this ridiculous G10 claim. Maybe they mean significant harmonic overtones go up to 25.1 kHz? I used linear frequency scale, because here it works nicely.




^Exactly what I was referring to.

That is an amazing image btw of the recording!


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## ADUHF (Mar 18, 2021)

bigshot said:


> ADUHF, I said that the only purpose for 24 bit is to provide a deeper noise floor for sound processing in mixing and mastering. When you sit in your living room listening to Beethoven on a commercially recorded CD, there is absolutely no reason for it. Consumer sound processing doesn't get anywhere near making any difference either. 12 bit is sufficient for listening to music in the home. 16 bit is overkill. There is plenty of headroom there.
> 
> The only way to understand what is important and what isn't is to take some music tracks and run them through different kinds of degradation and see the effect in real world applications. Ethan Winer does that in the videos in my sig file. He takes a horrible buzzing noise and mixes it into music and drops it -10dB at a time. Take a guess where you can't hear it any more under the music... I think you will be very surprised. You can download his files and listen to them for yourself.
> 
> ...



Hmm...

I think I agree with the basic thrust of what you're saying here. But I'm not sure it really addresses my question re bit depth, and changes to volume or dynamic range.

I obviously don't have the technical knowledge to debate the question intelligently though. So I suggest we call it a draw. 

2k is also good enough for me on video, since that's all my little Samsung and BD player currently support.


----------



## 71 dB

ADUHF said:


> ^Exactly what I was referring to.
> 
> That is an amazing image btw of the recording!


It is called a spectrogram.


----------



## bigshot

ADUHF said:


> I think I agree with the basic thrust of what you're saying here. But I'm not sure it really addresses my question re bit depth, and changes to volume or dynamic range. I obviously don't have the technical knowledge to debate the question intelligently though. So I suggest we call it a draw.



There is a good article linked in my sig about high data rate audio. It's called CD Sound Is All You Need. It's got a lot of useful info.


----------



## Davesrose

ADUHF said:


> 8k is included in the Rec. 2020 UHD spec. And there are already streaming services which offer some 8k content, including YouTube. Physical media seems a bit unlikely though for the near term. (If it picks up overseas though, who knows.)
> 
> 4k UHD hit its peak for player sales a few years ago. And has been declining since then. And better compression would also be needed to squeeze the 8k files onto current UHD discs.
> 
> ...


You're missing my point that content is not 8K.  I know video codecs have protocols for 8K, and there are digital cameras that can record 8K video.  But my point was that cinema movies have only recently been edited in 4K (you have to factor all the needed computer power for rendering so many layers of video cuts and VFX composites).  Gaming seems like the main category that can easily go 8K.  There may be some on YouTube who are recording in 8K and trying to post....I have seen other YouTubers try 4K and say it's not worth the extra effort for their content, and stay 2K.  YouTube and Vimeo are the main sources that allow 8K standards: all other services that have their own content (commercial TV or movies) are 2K and/or 4K to have optimal bandwidth.  Cinema content is not tied to physical discs: it starts with the studio digital intermediate then going to distribution to cinemas and home media (with home media, 4K streaming standards are becoming more popular...HBO finally getting on board with some new content being 4K Dolby Vision/Dolby Atmos).

When it comes to color gamut of display, one shouldn't just generalize by display type....as it really depends on the quality of panel and processor.  However, Tom's Hardware seems to indicate that Sony's Master Series OLEDs have very high gamuts.  OLED is my main display since I bought it for movies.  I've since gotten a PS5....I don't believe I'll have issues with burn-in though, as manufacturers have done enough to prevent it (save if you were watching a news network 24/7 with the same graphics on screen).  There's pixel shift for having a hud on screen.  RTings also has long term burn in tests with OLEDs, and it is the TV networks that have continual graphics (with them having it on non stop) that start to show significant burn in.


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## bigshot (Mar 19, 2021)

4K is perfectly capable of producing an image as good as film in theatrical projection. It's primarily a format for theatrical distribution and digitizing and archiving films for preservation purposes. 8K exists, but it is intended for films shot in large formats like 70mm, Cinerama, etc.

For the home, if you sit at the recommended seating distance, there really isn't any need for anything over 1080p or 2K (which are basically the same). The only benefit of 4K in the home is being able to get up and look closely at the screen.

And like audio, the resolution isn't the real determiner of image quality. I have DVDs that are mastered better than blu-rays.


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## Davesrose (Mar 19, 2021)

bigshot said:


> 4K is perfectly capable of producing an image as good as film in theatrical projection. It's primarily a format for theatrical distribution and digitizing and archiving films for preservation purposes. 8K exists, but it is intended for films shot in large formats like 70mm, Cinerama, etc.
> 
> For the home, if you sit at the recommended seating distance, there really isn't any need for anything over 1080p or 2K (which are basically the same). The only benefit of 4K in the home is being able to get up and look closely at the screen.
> 
> And like audio, the resolution isn't the real determiner of image quality. I have DVDs that are mastered better than blu-rays.


For me, the bigger jump with 4K home formats isn't so much resolution (though it helps when screens are getting larger), but also standards with HDR.  Whereas audible dynamic range was already reached with audio formats with early digital audio, the human eye can see 10-14 stops of light (and traditional TV formats carried 8 stops of DR).  Image formats can get up to 32 stops of light for simulation with 3D rendering or post processing of an environment that has bright light and deep shadow (say a picture taken indoors and you want to include a window with bright daylight).  Now 4K displays can start comfortably show 10 stops of DR.  Cinema cameras and 2K digital intermediates (that are in higher DCI resolutions for widescreen format than 1080P) have also had dynamic ranges exceeding 8bits (currently the best RAW video codecs are getting around 16bits).

I did learn traditional photography in the darkroom and then transitioned to digital.  My impression of my first digital cameras were that it had better low light performance than high speed film, but it was also obvious its dynamic range wasn't as good as film (I could "burn" or "dodge" areas to get more contrast out in areas...vs digital bringing in noise).  That quickly changed with more generations of digital sensors....now I don't see any advantage with film as a spec.  Seeing restored films in 4K, it's also easier to see the limitation of the film source (where a movie title can have different film stocks and ISO for given shots)...as well as lens and focusing technologies.


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## bigshot (Mar 20, 2021)

Turning out the lights in your viewing room and calibrating your black point can help a lot with getting better dynamics in video too. I have a projector, and those generally don't have the best contrast ratios, but with the lights off and on the big screen it isn't much of a problem. A lot of people have fancy 4K monitors, but they watch it like a TV set with the lights on in the room and don't get any of the benefits of the increased dynamics and improved color.

It's similar with audio... Dynamics extend downward. If you have a high ambient noise floor, broad dynamics aren't perceptible below a certain point. Wide dynamics aren't a positive thing on a train or plane or car or in the street.


----------



## 71 dB

I have watched movies from VHS tapes on a 14" standard definition TV. Eventually I moved to 32" standard definition TV (the EISA awarded Panasonic with 100 Hz picture and flat screen - it was the best money can buy) and DVD. The increase in picture quality was great! A decade later I moved to 32" HD flat TV and Blu-ray. Massive improvement in picture quality and suddenly movies looked like movies. They even play at the correct speed because there is no 4 % PAL speed-up we Europeans have suffered from in the standard definition era (because of 25 frames per second PAL system). 

I fell in love with Blu-ray. It is the first physical video format in my life that feels "good enough", so good that I don't need anything better. Given the reality that the screens I will be watching my movies from will remain rather small, I don't see any reason to go 4K. I have not seen HDR-picture, but to my eyes well-done Blu-rays look awesome. It's not about whether I would see the difference. I'm sure I would. It's about whether I "need" the improvements to ENJOY movies, which is the whole point for me.  

I have been upgrading my DVD-collection to Blu-rays, but it is not easy because they released much more stuff on DVD than Blu-ray. Blu-ray releases are also much more "localized" so that a French release of a Japanese movie might have only French subtitles when I need _at least_ English subtitles or preferably Finnish subtitles on non-English movies. So, if a Japanese movie is released only in Japan and France, I am probably out of luck. DVD-releases were often (but not always) much broader: You might have 20+ subtitle languages! Now that Brexit happened and ordering stuff from UK is becoming problematic, I have been ordering Blu-rays from Germany, which is one of the largest markets for Blu-rays. I have to be careful about the subtitles if the movie itself isn't English.

My taste in movies is such that what I would like to collect isn't released widely. I am not into Marvel superhero blockbusters with $300 million budgets. They are boring audiovisual noise to me. If I am having hard time collecting movies on Blu-ray, collecting them on 4K must be really frustrating! That's why I am not jumping on the 4K train anytime soon if ever.

Just watched the movie *Firestarter* (1984) on Blu-ray. It has got a soundtrack by Tangerine Dream. The picture quality of this German release is far from the best I have seen on Blu-ray, but it is decent. However, the real problem for me with this presentation is the original mono (!) sound which is quite crappy for a 80's movie. 4K would not help that at all.

My point is at this point of my life I am hardly at all interested of chasing more bits and resolution. I am more interested of getting my hands on stuff that I like and enjoy. Give me a decent Blu-ray of an obscure movie that for some reason resonates with my weird mind and I am happy. How about Michael Haneke's _"The Piano Teacher"_ on Blu-ray with Finnish subtitles? That's what I want, not HDR.


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## Davesrose (Mar 22, 2021)

71 dB said:


> I have watched movies from VHS tapes on a 14" standard definition TV. Eventually I moved to 32" standard definition TV (the EISA awarded Panasonic with 100 Hz picture and flat screen - it was the best money can buy) and DVD. The increase in picture quality was great! A decade later I moved to 32" HD flat TV and Blu-ray. Massive improvement in picture quality and suddenly movies looked like movies. They even play at the correct speed because there is no 4 % PAL speed-up we Europeans have suffered from in the standard definition era (because of 25 frames per second PAL system).



I'm a child of the 80s and have fond memories of spending time with my dad (a cinephile as well) by renting all sorts of movies on VHS from what was the supplier that had the largest and cheapest selection of VHS movies for rent: Pharmor.  It's funny that compared to my dad, I was the first to embrace the new media (until 4K, in which he was first to get an OLED display).  In college, I got a DVD drive for my computer, and tried having a mish mash of loudspeakers for fronts and computer speakers in back for a Dolby Surround 5.1 experience with a 19" display.  My dad had a 32" Sony CRT and Dolby surround with VHS tape.  He has a sister in Vienna....so back then it was considered we wouldn't do much exchanging.

By the time I had more disposable income, HDTV plasmas were becoming the premium TV format (there were also cheaper EDTVs that carried DV resolutions).  My 42" Panasonic plasma didn't have Full HD (that was unheard of then), but it's interesting that it still looks better with 1080i/p content (it's native resolution is inbetween, and it must have a better de-interlacer/de-scaler vs upscaler).  When my parents first viewed my HDTV (at that point, before HD discs and just a few HDTV channels)....my dad claimed no difference compared to a good DVD.  He had started his addiction with DVD collecting (he probably collected over 5,000 total).  A few more years, and with the sales of HD-DVD when it was starting to fade, HD-DVD was still something good for spending a few bucks to see what true HD media was like (by that point, blu-ray players were still expensive).  I had gotten a Toshiba HD-DVD player and several titles: Warner was good about having a cheaper exchange program of getting the same blu-ray titles.  Just like DVD, I first got a blu-ray drive for my computer.  I also updated my audio system to HDMI with full 7.1 speakers (including main towers).  What's nice about audio systems....I'm still using those 7 main speakers with my current 7.1.4 system.

The good thing about HD standards and being in a family that has members in Europe.....is that since the resolution is the same, many brands do make equipment that is the same model and has the variable refresh rates of PAL vs NTSC.  Unfortunately, blu-ray still kept region encoding....so I still have a modded Oppo region free player (and it's arduous with my current Denon 3D audio receiver in that I have to disable enhanced HDMI for it to get it to display an image: which is different than any other source).  When I got a VPN, I was also able to order European blu-rays with digital copies and still redeem the digital copies.  4K media is better with sharing between regions since it is region free (though physical media is dying and my dad does more internet rips of North American content for my aunt in Austria).

We're in interesting times if you're just a cinema purist looking for the most esoteric title.  I did try plugging a VHS player into my HDTV, and I was just appalled with the quality.  From time to time, I still watch SD analog video content on my big screen for its content, but I have to disassociate as to not having a cohesive image on such a large screen.  DVD can be hit or miss depending on how it was mastered.....but it is a medium in which there are a lot of out of print titles.  I'm not sure the difference in number of titles between blu-ray and HD streaming, but some titles will always just be physical and never re-encoded for online consumption.  However, there's also old movies that were scanned and then restored in 4K or 8K....and I've found they can be quite impressive in their new itteration.  Certainly not many 4:3 mono B&W titles....but even early Alfred Hitchcock movies have had some uptick (and when I compare a title that was HD vs 4K with HDR, I usually can pick up a difference with deeper contrast range).  There's also a number of movies 1980s+ that have been remastered in Atmos/DTS:X.  I have found some have been impressive.

I'm closer to my aunt in Vienna thanks to the internet: it's really nice that we can FaceTime vs just speaking over the phone.  We do more exchanging of music, American TV shows, and movies.  There's also not a concern about PAL vs NTSC with our digital exchanges.

I write this as I just finished watching a documentary about BlockBuster video (and how there's one independent outlet still identified as BlockBuster).  Many contributors were going on about physical media and how there's just something about getting that VHS case and opening it (and now amazing young kids about how a movie is stored on something like this).  I actually don't have such nostalgia....videotape was good for back then, but it's just terrible from a quality standpoint.  4K to date is the best for a home cinema experience.


----------



## magicscreen

GIYF

You can type into google:
24*192000
16*44100

4608000 > 705600 
The Hi-Res is better 6.53 times .

This is pure mathematics.


----------



## 118900

magicscreen said:


> GIYF
> 
> You can type into google:
> 24*192000
> ...


Have you actually read and, more crucially, understood the Nyquist-Shannon theorem?

typing arbitrary mathematical functions doesn’t  really help you.

that is pure logic (unless the aim of your comment was just to try and trigger people and troll the board)


----------



## castleofargh

After 400 pages, someone finally noticed that more bits and more samples mean more stuff.

Case closed...


----------



## SoundAndMotion

magicscreen said:


> GIYF
> 
> You can type into google:
> 24*192000
> ...


TFTT  I'd never heard of google before your post. 

2 Tylenol is good for a headache.
13 Tylenol is better 6.5 times.

Award-winning chili recipe says 3 tbsp salt.
20 tbsp salt is better 6.67 times.

This is pure mathematics... but one wonders whether mathematically-demonstrated "more is better" answers all questions...


----------



## 118900

SoundAndMotion said:


> TFTT  I'd never heard of google before your post.
> 
> 2 Tylenol is good for a headache.
> 13 Tylenol is better 6.5 times.
> ...


The sadness is that the only mathematical proof provided was that more is more, not that it’s better.


----------



## 71 dB

magicscreen said:


> GIYF
> 
> You can type into google:
> 24*192000
> ...


Yes, you can use Google to calculate these. Or you can use WolframAlpha. Or you can do what I use, the good old SHARP EL-5030 Scientific calculator from the late 80's.

*Bigger number is a bigger number. Sometimes it is "better." Sometimes it is "worse." Sometimes it doesn't make a difference.*

Digital audio is a bit tricky, because it is not always so intuitive. You can use Google also to study sampling theorem and human hearing. The sampling theorem (it is pure mathematics!) says how many samples per second we need to take to achieve the bandwidth we want. Human hearing covers 20 kHz of bandwith and about 130 dB of long term dynamic range, but only about 70 dB off short term dynamic range relevant to music listening. So, to have some safety margin lets have 80 dB of dynamic range. That's 20 dB more than vinyl gives at best. Okay, how many bits is 80 dB? 6 dB/bit => 80 dB / (6 dB/bit) is about 13 bits. Use of shaped dither can increase perceptual dynamic range 10 - 20 dB! So, 13 bits should be enough. If 13 bits is enough, so is 16! That's 3 bits worth of _overkill_. Young children hear up to 20 kHz and older people less than than. What is your own limit? 17 kHz? 16 kHz? 15 kHz? So, if sampling theory says we need to have a sampling frequency at least twice the highest frequency in our signal we need to go to 44100 Hz or so, but do not need 192 kHz because we are not bats.

*16 bit/44.1 kHz covers human hearing for the purpose of music listening. In any situation of practicality and sanity.*

Sampling theory also tells us the correctly bandlimited signals can be constructed 100 % accurately. We have one problem however, _quantization_. It creates distortion and noise. The less bit depth, the stronger the distortion and noise is. Now enters the miraculous savior named _dither_ I mentioned already above. Dither makes an offer you can't resist: It promises to remove distortion caused by quantization entirely for the price of increasing the noise level just a little bit, but the noise doesn't sound as bad as the granulating distortion! Okay, you need 13 bits and you have 16 bits... ...that is 18 dB of safety margin and dither would raise your noise floor by a few decibels. Hmm, surely you can accept that and you get rid of ALL quantization distortion! Even quiet signals are precise and you don't even hear the masking effect of the noise floor, because it is below the short term dynamic range of your hearing. Great deal! Isn't this dither so generous and nice?

*When you understand digital audio and you know the facts you realize 16 bit/44.1 kHz already gives you everything you need for consumer audio.*

More is better if you need more. It is not better if you don't need more. Human hearing dictates how much we need from digital audio, where the limit is. More can be worse. More bits requires more storage space. More bandwidth means more possibilities for unwanted interferences. Why suffer from such problems if the are no benefits? 24 bits is beneficial in audio _production_ for technical reasons and more bandwidth can be useful in some production situations, but thats it. For consumers it all can be downsampled to 16 bit/44.1 kHz without loss of audio quality. Don't be fooled by Hi-res sellers. They make you pay for your ignorance, your lack of understanding of digital audio and human hearing. Instead listen to people who do understand digital audio and are not after your money. I have no incentive to lie or deceive. I'm just telling how it is.


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## peskypesky (Apr 25, 2021)

SoundAndMotion said:


> TFTT  I'd never heard of google before your post.
> 
> 2 Tylenol is good for a headache.
> 13 Tylenol is better 6.5 times.
> ...


I smashed a cockroach and killed him.
I then smashed him 6 more times, so he's 6 times more dead.

I washed my clothes and then put them in the dryer. When the cycle was done, the clothing felt dry to the touch, but I dried them 6 more times, so they are 6 times drier.

This is simple mathematics.


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## 118900

peskypesky said:


> I smashed a cockroach and killed him.
> I then smashed him 6 more times, so he's 6 times more dead.
> 
> I washed my clothes and then put them in the dryer. When the cycle was done, the clothing felt dry to the touch, but I dried them 6 more times, so they are 6 times drier.
> ...


But you didn’t kill/dry them *6.53* times better, just 6. No good 🤣🤣🤣


----------



## peskypesky

71 dB said:


> Yes, you can use Google to calculate these. Or you can use WolframAlpha. Or you can do what I use, the good old SHARP EL-5030 Scientific calculator from the late 80's.
> 
> *Bigger number is a bigger number. Sometimes it is "better." Sometimes it is "worse." Sometimes it doesn't make a difference.*
> 
> ...



Unfortunately, you waste your time trying to explain complicated things to people who can only understand basic math like multiplication and addition.


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## peskypesky (Apr 25, 2021)

juansan said:


> But you didn’t kill/dry them *6.53* times better, just 6. No good 🤣🤣🤣


next time I will smash him 7 times more.
and dry my clothes 7 times more.

7 > 6.53


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## 71 dB

peskypesky said:


> Unfortunately, you waste your time trying to explain complicated things to people who can only understand basic math like multiplication and addition.


I think I have fantasies about someone who happens to be on the brink of understanding these things reading my posts and having his/her Heureka moment...


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## peskypesky

I went to the eye doctor and got some new contact lenses. But I don't just wear one contact lens on each eye. I wear 7.

Because 7 is 7x more than 1.

You can Google it.


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## peskypesky (Apr 25, 2021)

71 dB said:


> I think I have fantasies about someone who happens to be on the brink of understanding these things reading my posts and having his/her Heureka moment...



Sadly, people believe what they want to believe. And many of them cannot trust in science when it shows them something that seems counter-intuitive, or that seems to defy their personal observations. For example, even now, in the 21st Century, there are people who have trouble accepting that the earth is round.

Even scientists can have trouble accepting science and math. My older brother is a pyschiatrist and geneticist, and I've tried to explain to him many times why buying Hi-Res digital files is a waste of money, especially for his 60-year old ears. But he just can't wrap his head around it. I've challenged him many times to do ABX testing of high-res AAC files versus FLACs, 16bit or 24bit, and he won't do it. Why?  Because he's afraid that he won't be able to tell the difference, and then he will feel remorse for all the money he has wasted, and he will feel humiliation. He does not want to feel those unpleasant things, so it is easier for his ego to go on buying the high-res files.

I still remember the first time I did an ABX test and discovered I could not tell the difference between 320kb mp3 and FLAC. It was hard to believe. The FLAC's are so much BIGGER!  But, I've done probably a hundred ABX tests since then, and only a couple of times could tell FLAC from high bitrate lossy files.


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## bigshot (Apr 25, 2021)

magicscreen said:


> This is pure mathematics.



To follow along with this train of thought...

Sound waves travel at about 340 meters a second. Light travels at about 300,000 kilometers a second. I wish I could figure out how many times better light is than sound, but I'm not good at math. I'll google it for myself.

I like this thread. It's like haiku for dummies.


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## 118900

peskypesky said:


> Sadly, people believe what they want to believe. And many of them cannot trust in science when it shows them something that seems counter-intuitive, or that seems to defy our personal observations. For example, even now, in the 21st Century, there are people who have trouble accepting that the earth is round.
> 
> Even scientists can have trouble accepting science and math. My older brother is a pyschiatrist and geneticist, and I've tried to explain to him many times why buying Hi-Res digital files is a waste of time, especially for his 60-year old ears. But he just can't wrap his head around it. I've challenged him many times to do ABX testing of high-res AAC files versus FLACs, 16bit or 24bit, and he won't do it. Why?  Because he's afraid that he won't be able to tell the difference, and then he will feel remorse for all the money he has wasted, and he will feel humiliation. He does not want to feel those unpleasant things, so it is easier for his ego to go on buying the high-res files.


Tell him hi-res files aren’t any good unless they cost 7 times moar than 16/44.1.....

joking aside though I can appreciate why some people feel more comfortable doing that. So long as they don’t try and tell others why their view is better without being able to explain the science behind their misguided views (like our google loving friend who started this thing rolling).


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## bigshot

Wise man once said, "Too much is never enough."


----------



## peskypesky

juansan said:


> Tell him hi-res files aren’t any good unless they cost 7 times moar than 16/44.1.....
> 
> joking aside though I can appreciate why some people feel more comfortable doing that. So long as they don’t try and tell others why their view is better without being able to explain the science behind their misguided views (like our google loving friend who started this thing rolling).


let's just be glad he didn't make the same post 6.5 times to prove his point 6.5 times more.


----------



## KeithPhantom

I don’t know how people keep posting about 96 dB theoretical DR (undithered) vs. 144 dB theoretical DR (undithered). For music listening, it won’t make a difference, and by the way, in music production dither is a standard practice even encouraged, so expect better effective DR. Finally, the SINAD of any source is not even close to the DR of the formats we currently use, being nonlinearities a greater issue than DR.


----------



## 71 dB

peskypesky said:


> Sadly, people believe what they want to believe. And many of them cannot trust in science when it shows them something that seems counter-intuitive, or that seems to defy their personal observations. For example, even now, in the 21st Century, there are people who have trouble accepting that the earth is round.
> 
> Even scientists can have trouble accepting science and math. My older brother is a pyschiatrist and geneticist, and I've tried to explain to him many times why buying Hi-Res digital files is a waste of money, especially for his 60-year old ears. But he just can't wrap his head around it. I've challenged him many times to do ABX testing of high-res AAC files versus FLACs, 16bit or 24bit, and he won't do it. Why?  Because he's afraid that he won't be able to tell the difference, and then he will feel remorse for all the money he has wasted, and he will feel humiliation. He does not want to feel those unpleasant things, so it is easier for his ego to go on buying the high-res files.
> 
> I still remember the first time I did an ABX test and discovered I could not tell the difference between 320kb mp3 and FLAC. It was hard to believe. The FLAC's are so much BIGGER!  But, I've done probably a hundred ABX tests since then, and only a couple of times could tell FLAC from high bitrate lossy files.


Life has humiliated me so many times so badly that the humiliation of being proven wrong is nothing. I feel like my life has been a decades long journey of painful lessons teaching me slowly to not be wrong about everything. People who fear such humiliation have really lived inside a protective bubble where the life hasn't kicked them to the head...


----------



## 71 dB

KeithPhantom said:


> I don’t know how people keep posting about 96 dB theoretical DR (undithered) vs. 144 dB theoretical DR (undithered). For music listening, it won’t make a difference, and by the way, in music production dither is a standard practice even encouraged, so expect better effective DR. Finally, the SINAD of any source is not even close to the DR of the formats we currently use, being nonlinearities a greater issue than DR.



Not to mention, that unless your music creations are 100 % computer generated sounds, achieving 144 dB of dynamic range is pretty much impossible. Good luck recording the drums with 144 dB of DR! People don't understand the huge difference of a 0 dB sound and 144 dB sound. 0 dB is the noise floor in a good anechoic chamber, a place most people never experience in their lives while 144 dB is the sound of a hand gun!


----------



## Steve999

71 dB said:


> Life has humiliated me so many times so badly that the humiliation of being proven wrong is nothing. I feel like my life has been a decades long journey of painful lessons teaching me slowly to not be wrong about everything. People who fear such humiliation have really lived inside a protective bubble where the life hasn't kicked them to the head...



“A man who gives a good account of himself is probably lying, since any life when viewed from the inside is simply a series of defeats.”

-George Orwell


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## 71 dB

Steve999 said:


> “A man who gives a good account of himself is probably lying, since any life when viewed from the inside is simply a series of defeats.”
> 
> -George Orwell


I have tons of defeats to show for! How about the ninth grade Math competition in school? I was THIRD! That's not bad you might say, but I was THE Math nerd in my class! That was humiliating, but nothing compared to the struggles I have experienced in work life. I have a lot of frustration, issues with low self-esteem and anger in me and sometimes it shows in my posts. In friendly environments I can show a happier side of myself.


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## 118900 (Apr 26, 2021)

71 dB said:


> Not to mention, that unless your music creations are 100 % computer generated sounds, achieving 144 dB of dynamic range is pretty much impossible. Good luck recording the drums with 144 dB of DR! People don't understand the huge difference of a 0 dB sound and 144 dB sound. 0 dB is the noise floor in a good anechoic chamber, a place most people never experience in their lives while 144 dB is the sound of a hand gun!


Apparently 144 dB (if speakers were physically capable of reproducing such a loudness level) is enough to severely injure the listener, if not worse


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## bigshot

The most dynamic commercially recorded music has a dynamic range of about 50dB, so redbook is well into overkill.


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## 71 dB

bigshot said:


> The most dynamic commercially recorded music has a dynamic range of about 50dB, so redbook is well into overkill.


Redbook is overkill and Hi-rez is overbill...


----------



## Roland P

If redbook is overkill, then what sampling-rate/bit-depth would be the sweet spot?


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## 71 dB

Roland P said:


> If redbook is overkill, then what sampling-rate/bit-depth would be the sweet spot?


The overkill offered by redbook doesn't harm much, but 13 bits of dynamic range is enough.


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## Slaphead

Roland P said:


> If redbook is overkill, then what sampling-rate/bit-depth would be the sweet spot?


Redbook is probably the sweet spot as it encompasses slightly more than we can hear, but as @71 dB says 13-14 bits of dynamic range would be more than adequate.

As for the 44.1 KHz sampling rate it's pretty much bang on for human hearing, although some would say that 48Khz allows for a less aggressive AA filter.


----------



## bigshot

Not that there's really all that much to be heard above 14 or 15kHz...


----------



## 71 dB

Slaphead said:


> Redbook is probably the sweet spot as it encompasses slightly more than we can hear, but as @71 dB says 13-14 bits of dynamic range would be more than adequate.
> 
> As for the 44.1 KHz sampling rate it's pretty much bang on for human hearing, although some would say that 48Khz allows for a less aggressive AA filter.


Yes, little higher sampling rate would give the benefit of easier anti-alias/reconstruction filtering, but it is a very very small benefit. 
13 bit / 54 kHz would have almost the same bitrate as 16 bit / 44.1 kHz.


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## ADUHF (Apr 29, 2021)

This topic is badly in need of an update imo. Because alot of people are still thinking in terms of technology that is now close to 30 years old, which has largely fallen into disuse.

While it's probably true that most humans don't have "bat ears" that can hear above the 22.05 kHz Nyquist frequency of 44.1 kHz content, alot of content now uses the 48 kHz video standard for audio instead. Including YouTube clips encoded with the Opus codec (which is what most browsers now use).

I sold off the majority of my CD collection ages ago. So the only time I really think about audio bit depths and sample rates is when using my PC for viewing/listening to online content. And since I don't have the ability to always use the native rates of the content, and am often manipulating the amplitude in the digital domain via EQs and what have you, I usually keep the bit depths and sample rates on my PC at the highest setting my audio devices will allow, which is currently 24-bit / 48 kHz. Because this seems to work best for me.


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## ADUHF (Apr 29, 2021)

Removed by ADUHF.


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## sander99

ADUHF said:


> This topic is badly in need of an update imo. Because alot of people are still thinking in terms of technology that is now close to 30 years old, which has largely fallen into disuse.
> 
> While it's probably true that most humans don't have "bat ears" that can hear above the 22.05 kHz Nyquist frequency of 44.1 kHz content, alot of content now uses the 48 kHz video standard for audio instead. Including YouTube clips encoded with the Opus codec (which is what most browsers now use).
> 
> I sold off the majority of my CD collection ages ago. So the only time I really think about audio bit depths and sample rates is when using my PC for viewing/listening to online content. And since I don't have the ability to always use the native rates of the content, and am often manipulating the amplitude of the content in the digital domain via EQs and what have you, I usually keep the bit depths and sample rates on my PC at the highest setting my audio devices will allow, which is currently 24-bit / 48 kHz.


This topic is about what is enough for distribution of music. Actually this thread is specifically about bit depth and not sampling rate, but never mind that.

I don't think anyone here wants to argue against using 24 bits in your computer/equipment for digital volume control and other digital processing.
48 kHz indeed has its place for practical reasons in relation to video. The 44.1 vs 48 kHz difference is not so important in the discussions about sampling rates, it is more about 44.1 vs 88.2/176.4/... and 48 vs 96/192/.... Again mainly for distribution: I don't think anyone here wants to argue against oversampling in your DAC (or delta sigma converters).


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## ADUHF (Apr 29, 2021)

sander99 said:


> This topic is about what is enough for distribution of music. Actually this thread is specifically about bit depth and not sampling rate, but never mind that.
> 
> I don't think anyone here wants to argue against using 24 bits in your computer/equipment for digital volume control and other digital processing.
> 48 kHz indeed has its place for practical reasons in relation to video. The 44.1 vs 48 kHz difference is not so important in the discussions about sampling rates, it is more about 44.1 vs 88.2/176.4/... and 48 vs 96/192/.... Again mainly for distribution: I don't think anyone here wants to argue against oversampling in your DAC (or delta sigma converters).



Thank you for the reply, sander99. Fwiw, I wouldn't disagree with any of this. But I fear some of these subtleties may be lost on some of the newbies to this topic, like myself.

And I think alot of folks are now getting most of their music online (rather than via physical media), where a good deal of the content seems to be shifting over to the 48 kHz video standard.


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## bigshot (Apr 29, 2021)

Sanders99, my understanding is that 48 divides more evenly. (48, 24, 12, 6, 3) That is the same reason film runs at 24. (24, 12, 6, 3) It isn't for higher frequencies as much as it is number crunching and dividing for technical reasons. Is that correct?


----------



## sander99

bigshot said:


> Sanders99, my understanding is that 48 divides more evenly. (48, 24, 12, 6, 3) That is the same reason film runs at 24. (24, 12, 6, 3) It isn't for higher frequencies as much as it is number crunching and dividing for technical reasons. Is that correct?


Actually I don't know for sure, but probably yes. Not much of an answer, I know, but I didn't want to ignore your question.


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## castleofargh

https://www.tvtechnology.com/opinions/digital-audio-sample-rates-the-48-khz-question

Once again, audio choices were based on sight instead of sound. ^_^'


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## Roland P (Apr 30, 2021)

> Many in the U.S. television industry liked *60 kHz* as a standard sample rate because it was free of leap frames and split frequencies, and it synchronized readily with all timing signals used in 60 Hz and 50 Hz television systems, 24 Hz film and the 13.5 MHz component digital video sample rate. *The professional audio industry, however, considered it wastefully high*, and there was a quantity of 48 kHz software extant in Europe.


60kHz was considered wastefully high 🙂.

Quite amazing that this stuff has been sorted out in the late 1970's already.


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## 71 dB

Roland P said:


> 60kHz was considered wastefully high 🙂.
> 
> Quite amazing that this stuff has been sorted out in the late 1970's already.


(1) At that time "hi-rez" was extremely demanding if not impossible technologically and also REALLY expensive.
(2) There was no "hi-rez" snake oil bs to confuse people => opinions based on science.

No wonder they had rational views of the sensible sample rates.


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## bigshot

Too much is never enough!


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## Brahmsian (May 6, 2021)

peskypesky said:


> Sadly, people believe what they want to believe. And many of them cannot trust in science when it shows them something that seems counter-intuitive, or that seems to defy their personal observations. For example, even now, in the 21st Century, there are people who have trouble accepting that the earth is round.
> 
> Even scientists can have trouble accepting science and math. My older brother is a pyschiatrist and geneticist, and I've tried to explain to him many times why buying Hi-Res digital files is a waste of money, especially for his 60-year old ears. But he just can't wrap his head around it. I've challenged him many times to do ABX testing of high-res AAC files versus FLACs, 16bit or 24bit, and he won't do it. Why?  Because he's afraid that he won't be able to tell the difference, and then he will feel remorse for all the money he has wasted, and he will feel humiliation. He does not want to feel those unpleasant things, so it is easier for his ego to go on buying the high-res files.
> 
> I still remember the first time I did an ABX test and discovered I could not tell the difference between 320kb mp3 and FLAC. It was hard to believe. The FLAC's are so much BIGGER!  But, I've done probably a hundred ABX tests since then, and only a couple of times could tell FLAC from high bitrate lossy files.


What‘s so sad about it if he’s enjoying his music? I buy pure DSD 256 files. They sound fantastic. It’s just my imagination? Fine. If my imagination is enhancing my enjoyment, it’s enhancing my enjoyment.  I don’t bother A/Bing because, first, I couldn‘t care less; I want to listen to music not waste my time comparing files. And second my memory of the sonics vanishes almost instantly so that I probably wouldn’t get it right even if I could hear a difference. Or 320kb mp3 might sound indistinguishable in the moment but be more fatiguing in the long run. Or I might find it fatiguing just because I have a bias against it. At the end of the day, labels like Channel and Reference Recordings are making superb records in DSD and DXD. Whether it’s the DSD/DXD or just better engineering, these are some seriously spectacular classical music recordings and anybody who cares about good sound should support them, hi res or not. RR’s Shostakovich 5th with Honeck conducting is truly reference-class, a contender for THE best recording ever.


----------



## chef8489

Brahmsian said:


> What‘s so sad about it if he’s enjoying his music? I buy pure DSD 256 files. They sound fantastic. It’s just my imagination? Fine. If my imagination is enhancing my enjoyment, it’s enhancing my enjoyment.  I don’t bother A/Bing because, first, I couldn‘t care less; I want to listen to music not waste my time comparing files. And second my memory of the sonics vanishes almost instantly so that I probably wouldn’t get it right even if I could hear a difference. Or 320kb mp3 might sound indistinguishable in the moment but be more fatiguing in the long run. Or I might find it fatiguing just because I have a bias against it. At the end of the day, labels like Channel and Reference Recordings are making superb records in DSD and DXD. Whether it’s the DSD/DXD or just better engineering, these are some seriously spectacular classical music recordings and anybody who cares about good sound should support them, hi res or not. RR’s Shostakovich 5th with Honeck conducting is truly reference-class, a contender for THE best recording ever.


You do realize there are very few real dsd/dxd  music. The majority was pcm at one point so they can mix and edit it.


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## bigshot (May 6, 2021)

Using your own bias to make yourself happy is perfectly fine if you just keep it to yourself. The problem comes when you try to recommend your particular brand of placebo to other people. Head Fi is full of people doing this. It doesn't help. It hurts by muddying the water for people who actually want to use rational means of improving the sound quality of their system, not self hypnosis.

There are ways of actually improving sound quality. Spending money to build validation bias or being attracted to sales pitch full of glittering generalities and pseudo science to build expectation bias aren't ways to do that.



chef8489 said:


> You do realize there are very few real dsd/dxd  music. The majority was pcm at one point so they can mix and edit it.



I'm guessing he doesn't care and has no interest in knowing how digital audio works or how to improve fidelity in objective ways.


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## 71 dB

Brahmsian said:


> I buy pure DSD 256 files. They sound fantastic. It’s just my imagination?  At the end of the day, labels like Channel and Reference Recordings are making superb records in DSD and DXD. Whether it’s the DSD/DXD or just better engineering, these are some seriously spectacular classical music recordings and anybody who cares about good sound should support them, hi res or not. RR’s Shostakovich 5th with Honeck conducting is truly reference-class, a contender for THE best recording ever.


You kind of answered your own question here. Nobody says your DSD 256 files don't sound fantastic. I am sure they do, but I also know that if you make 16 bit/44.1 kHz PCM versions of them they wills till sound just as fantastic, at least in blind tests without placebo effects. Well composed, played, produced and mixed music tends to sound fantastic. Pretty much all my SACDs sound fantastic too, because those are really well made recordings, but the CD-layer sounds just as good if you forget it is not multichannel (SACD stereo layer sounds the same as the CD layer).


----------



## bigshot

I've found that not all redbook layers are the same mastering as the SACD layer. I've even found some that were completely different mixes. This sort of monkey business is especially prevalent in legacy album rock releases.


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## peskypesky (May 6, 2021)

Brahmsian said:


> What‘s so sad about it if he’s enjoying his music? I buy pure DSD 256 files. They sound fantastic. It’s just my imagination? Fine. If my imagination is enhancing my enjoyment, it’s enhancing my enjoyment.  I don’t bother A/Bing because, first, I couldn‘t care less; I want to listen to music not waste my time comparing files. And second my memory of the sonics vanishes almost instantly so that I probably wouldn’t get it right even if I could hear a difference. Or 320kb mp3 might sound indistinguishable in the moment but be more fatiguing in the long run. Or I might find it fatiguing just because I have a bias against it. At the end of the day, labels like Channel and Reference Recordings are making superb records in DSD and DXD. Whether it’s the DSD/DXD or just better engineering, these are some seriously spectacular classical music recordings and anybody who cares about good sound should support them, hi res or not. RR’s Shostakovich 5th with Honeck conducting is truly reference-class, a contender for THE best recording ever.


I find it sad when people are fooled into wasting money. You don't find it sad. So...we're different.

I also find it sad when some people in the 21st century still are unable to believe in science and math. You don't. So...we're different.


----------



## peskypesky

Brahmsian said:


> What‘s so sad about it if he’s enjoying his music? I buy pure DSD 256 files. They sound fantastic. It’s just my imagination? Fine. If my imagination is enhancing my enjoyment, it’s enhancing my enjoyment.  I don’t bother A/Bing because, first, I couldn‘t care less; I want to listen to music not waste my time comparing files. And second my memory of the sonics vanishes almost instantly so that I probably wouldn’t get it right even if I could hear a difference. Or 320kb mp3 might sound indistinguishable in the moment but be more fatiguing in the long run. Or I might find it fatiguing just because I have a bias against it. At the end of the day, labels like Channel and Reference Recordings are making superb records in DSD and DXD. Whether it’s the DSD/DXD or just better engineering, these are some seriously spectacular classical music recordings and anybody who cares about good sound should support them, hi res or not. RR’s Shostakovich 5th with Honeck conducting is truly reference-class, a contender for THE best recording ever.


some day, you'll realize how bad those DSD 256 files sound when they come out with super duper DSD 1024 files.


----------



## peskypesky

bigshot said:


> Not that there's really all that much to be heard above 14 or 15kHz...


At the age of 55, my hearing peaks at about 11.5kHz.


----------



## bigshot

I read a study that asked people to compare two samples: one full range response and the other everything removed from the top octave above 10khz. They asked people if they could hear a difference; and if they could, which one sounded better to them. The majority of people said they couldn't hear a difference and they sounded the same. A percentage (if I remember correctly, it was about 20-25%) said they could hear a difference; but almost all of them said that even though they could hear a slight difference, they didn't think one sounded any better than the other. It came out to something like 99% of people saying that the top octave made no impact on sound quality.


----------



## 71 dB

bigshot said:


> I've found that not all redbook layers are the same mastering as the SACD layer. I've even found some that were completely different mixes. This sort of monkey business is especially prevalent in legacy album rock releases.


I can believe this being the case with old rock albums, but my SACDs are classical music mostly from BIS label and also CPO + some discs from other labels.


----------



## bigshot

Classical usually doesn’t do that.


----------



## 71 dB

bigshot said:


> I read a study that asked people to compare two samples: one full range response and the other everything removed from the top octave above 10khz. They asked people if they could hear a difference; and if they could, which one sounded better to them. The majority of people said they couldn't hear a difference and they sounded the same. A percentage (if I remember correctly, it was about 20-25%) said they could hear a difference; but almost all of them said that even though they could hear a slight difference, they didn't think one sounded any better than the other. It came out to something like 99% of people saying that the top octave made no impact on sound quality.


Yeah, I remember that study. It is quite shocking how little frequencies above 10 kHz matter to sound quality.


----------



## peskypesky

71 dB said:


> Yeah, I remember that study. It is quite shocking how little frequencies above 10 kHz matter to sound quality.


this is probably why in over a hundred ABX tests, I haven't been able to tell the difference between FLACs and high-bitrate lossy files (mp3, aac, ogg, opus).  The high frequencies that are "lost" don't make a difference to my ears. Especially not at my age.


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## Brahmsian (May 6, 2021)

chef8489 said:


> You do realize there are very few real dsd/dxd  music. The majority was pcm at one point so they can mix and edit it.


Yes, indeed. There are inly a handful of labels that do pure DSD.


bigshot said:


> Using your own bias to make yourself happy is perfectly fine if you just keep it to yourself. The problem comes when you try to recommend your particular brand of placebo to other people. Head Fi is full of people doing this. It doesn't help. It hurts by muddying the water for people who actually want to use rational means of improving the sound quality of their system, not self hypnosis.
> 
> There are ways of actually improving sound quality. Spending money to build validation bias or being attracted to sales pitch full of glittering generalities and pseudo science to build expectation bias aren't ways to do that.
> 
> ...


I’m not recommending it to other people. Just the same, other people might already be enhancing their enjoyment that way.


71 dB said:


> You kind of answered your own question here. Nobody says your DSD 256 files don't sound fantastic. I am sure they do, but I also know that if you make 16 bit/44.1 kHz PCM versions of them they wills till sound just as fantastic, at least in blind tests without placebo effects. Well composed, played, produced and mixed music tends to sound fantastic. Pretty much all my SACDs sound fantastic too, because those are really well made recordings, but the CD-layer sounds just as good if you forget it is not multichannel (SACD stereo layer sounds the same as the CD layer).


I tend to agree with you. I was just reacting to the belief that it is so, so sad that some of us buy hi res music.


peskypesky said:


> I find it sad when people are fooled into wasting money. You don't find it sad. So...we're different.
> 
> I also find it sad when some people in the 21st century still are unable to believe in science and math. You don't. So...we're different.


OK. You keep being sad about it, and I’ll keep enjoying my music.


peskypesky said:


> some day, you'll realize how bad those DSD 256 files sound when they come out with super duper DSD 1024 files.


They already have DSD 512, but those files are too big even for me.


bigshot said:


> I read a study that asked people to compare two samples: one full range response and the other everything removed from the top octave above 10khz. They asked people if they could hear a difference; and if they could, which one sounded better to them. The majority of people said they couldn't hear a difference and they sounded the same. A percentage (if I remember correctly, it was about 20-25%) said they could hear a difference; but almost all of them said that even though they could hear a slight difference, they didn't think one sounded any better than the other. It came out to something like 99% of people saying that the top octave made no impact on sound quality.


I posted a meta-analysis that found that people could hear a difference. It was dismissed over supposedly bad methodology. I posted a Consumer Reports article claiming that in their audio lab they did indeed hear a difference. I was accused of cherry picking because they *also* said that the average consumer shouldn’t bother with high res files since the difference is so minuscule and, moreover, most people don’t have that kind of audio equipment. There was another study where subjects were ABXd and two sound engineers, trained tonmeisters, scored significantly above average, and the researchers wanted to have them back for further evaluation. But that was also dismissed out of hand, even by the researchers themselves, who basically discounted it as an anomaly. They just placed a big question mark over the sound engineers who scored so well. Not that I am convinced by the meta-analysis or by Consumer Reports, but it is clear that people have made up their minds and that having even a sliver of your thinking open on the subject is not allowed. The matter has been settled and anything that goes against it clearly has to be wrong.

My last post wasn’t an argument against the settled science but a reaction to peskypesky’s characterization of his brother as a poor dupe who is afraid to ABX because he might find that he couldn’t tell the difference and would then be filled with remorse because he paid extra for hi res files. I don’t know all the specifics there, but I wonder whether his brother would agree with that characterization and what he would say in his defense. As I see it, it’s more like buying extra warranty coverage for products that will probably never require it, i.e., peace of mind. For instance, if a label records in DSD 256 and doesn’t convert the file to DXD for processing and sends a copy of the DSD 256 file to NativeDSD, that’s the one I’ll choose because it’s basically an exact copy of the master. That file might not sound any better to me than the 320kb file were I to take the time to actually compare them, but since I don’t care to take the time to compare them, and since I know that the DSD file received directly from the label is as close to the source as you can get, that’s the one I choose. I don’t have to worry about it being converted, degraded, or manipulated in some way. If you don’t see the point in that, fine. But why call somebody else a dupe when possibly all they want is peace of mind that they’re getting the best product available?


----------



## bigshot (May 7, 2021)

You know. if you want to preserve your placebo illusion, it would be best to not post in Sound Science. Every time you post you hold up your balloon and hand us a dozen pins. You've been down this road before and got nowhere and left with your tail between your legs. Why do you keep coming back for more? Do you have some irresistible urge to be debunked? I'm just warning you that this isn't the best place for your peace of mind.


----------



## peskypesky

bigshot said:


> You know. if you want to preserve your placebo illusion, it would be best to not post in Sound Science. Every time you post you hold up your balloon and hand us a dozen pins. You've been down this road before and got nowhere and left with your tail between your legs. Why do you keep coming back for more? Do you have some irresistible urge to be debunked? I'm just warning you that this isn't the best place for your peace of mind.


Maybe masochism?


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## bigshot

You said the word I wouldn't say out loud. I think internet forums have all types of characters. Most people are quiet and just lurk and gather the info they need. Others talk. There are various reasons why people do that. Attracting attention- whatever kind of attention- is one of those.


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## chef8489

What's sad is I know better about 16bit vs 24bit and I still struggle at times picking a 24bit version on my qobuz subscription to listen to. I don't purchase them, just stream. Almost all my music I have purchased were 16/44.1 and usually ripped from cds.


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## bigshot

16/44.1 is all you need. https://web.archive.org/web/20200426202431/https://people.xiph.org/~xiphmont/demo/neil-young.html


----------



## chef8489

bigshot said:


> 16/44.1 is all you need. https://web.archive.org/web/20200426202431/https://people.xiph.org/~xiphmont/demo/neil-young.html


Yep I'm quite aware lol. Not sure why the mind still does the more is better. Ik they don't sound any different and not sure why if a 25bit is available I click that. I guess since I'm not paying for them it doesn't matter if qobuz plays a 16 vs 24.


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## bigshot

I'm posting that for the benefit of the lurkers!


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## chef8489

bigshot said:


> I'm posting that for the benefit of the lurkers!


Gotcha.


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## peskypesky (May 7, 2021)

bigshot said:


> 16/44.1 is all you need. https://web.archive.org/web/20200426202431/https://people.xiph.org/~xiphmont/demo/neil-young.html


Great article.

My favorite line: "Misinformation and superstition only serve charlatans.".

My second favorite:  "In any test where a listener can tell two choices apart via any means apart from listening, the results will usually be what the listener expected in advance; this is called confirmation bias and it's similar to the placebo effect. It means people 'hear' differences because of subconscious cues and preferences that have nothing to do with the audio, like preferring a more expensive (or more attractive) amplifier over a cheaper option."

And third:  "It's true enough that a properly encoded Ogg file (or MP3, or AAC file) will be indistinguishable from the original at a moderate bitrate."

This article explains why in many dozens of ABX tests, I haven't been able to tell the difference between high bitrate lossy files and 24/192 FLACS.


----------



## 71 dB

Brahmsian said:


> I tend to agree with you. I was just reacting to the belief that it is so, so sad that some of us buy hi res music.



It is good we have some sort of agreement. Over the years I have become "softer" and I try to understand emotional and even irrational aspects of audio experience. I try to understand people into vinyl, tubes, etc. but there has to be limits so that hard science has a say on things also. 

If you are ok with spending more money on hi-rez files in order to have peace of mind then that is your money and your peace of mind. It is clear peace of mind can be based on emotional and irrational aspects. For me science has given peace of mind. I am _convinced_ I don't need hi-rez. The purpose of this sound science section is to help people to learn and get convinced of what science teaches us and have immunity against being fooled by snake oil sellers.


----------



## castleofargh

Brahmsian said:


> I posted a meta-analysis that found that people could hear a difference. It was dismissed over supposedly bad methodology. I posted a Consumer Reports article claiming that in their audio lab they did indeed hear a difference. I was accused of cherry picking because they *also* said that the average consumer shouldn’t bother with high res files since the difference is so minuscule and, moreover, most people don’t have that kind of audio equipment. There was another study where subjects were ABXd and two sound engineers, trained tonmeisters, scored significantly above average, and the researchers wanted to have them back for further evaluation. But that was also dismissed out of hand, even by the researchers themselves, who basically discounted it as an anomaly. They just placed a big question mark over the sound engineers who scored so well. Not that I am convinced by the meta-analysis or by Consumer Reports, but it is clear that people have made up their minds and that having even a sliver of your thinking open on the subject is not allowed. The matter has been settled and anything that goes against it clearly has to be wrong.
> 
> My last post wasn’t an argument against the settled science but a reaction to peskypesky’s characterization of his brother as a poor dupe who is afraid to ABX because he might find that he couldn’t tell the difference and would then be filled with remorse because he paid extra for hi res files. I don’t know all the specifics there, but I wonder whether his brother would agree with that characterization and what he would say in his defense. As I see it, it’s more like buying extra warranty coverage for products that will probably never require it, i.e., peace of mind. For instance, if a label records in DSD 256 and doesn’t convert the file to DXD for processing and sends a copy of the DSD 256 file to NativeDSD, that’s the one I’ll choose because it’s basically an exact copy of the master. That file might not sound any better to me than the 320kb file were I to take the time to actually compare them, but since I don’t care to take the time to compare them, and since I know that the DSD file received directly from the label is as close to the source as you can get, that’s the one I choose. I don’t have to worry about it being converted, degraded, or manipulated in some way. If you don’t see the point in that, fine. But why call somebody else a dupe when possibly all they want is peace of mind that they’re getting the best product available


I might fuel an unnecessary fire but I prefer this type of post, to claims of definitive and universal inaudibility(or audibility).
Most of the regulars know about the references you mentioned, they exist. To me, it's a reasonable argument, so I thought I should support the effort, even if I’m not a big supporter of hires or DSD(aarrrrghhhhh) myself.


On the other hand it's a topic about bit depth, not ultrasounds. Seems like I’ll never manage to keep them apart for more than a few pages.


----------



## bigshot

More is not always better. Sometimes it's just more.


----------



## Brahmsian (May 7, 2021)

I don't have a balloon that can be popped, just as I don't have a tail to tuck between my legs. However, it's true that I've left here disgusted at the hostility of some on this thread. They're like chimps trying to rip off the face of anyone who enters their territory. But some of those same people have been very nice on other occasions, so I don't draw any conclusions.

As for the placebo illusion, people often imaginatively enhance their experiences. When you read a novel, your imagination fleshes out the picture beyond the words on the page. Children imagine that their action figures can kick some serious butt even while knowing that the figures aren't alive; the toy is just something their imagination can act upon. The power of the imagination to enhance one's experience is real, and the pleasure derived from it is real, which is one reason why people don't want to give up their illusions. I remain agnostic on the matter. But being agnostic is not good enough; you're expected to either toe the line or not make a peep. There are totalitarian states that have more freedom of speech than this thread.

Especially in the classical music realm, labels are recording and issuing their music in DXD and DSD 256, and the end product often sounds superb. Whether recording in DXD and DSD 256 is incidental to the great sound or not, the fact that their recordings are outstanding is something audio enthusiasts should celebrate. Forget about the so-called golden age of recording in the past; _this_ is the Golden Age. If you like classical music, check out some of the work that Reference Recordings has been doing. Simply superb!

Carry on.


----------



## bigshot

Man! So many words to say that you don’t want to be here. When you start off with chimps eating faces, I’m not going to bother reading any further. I dismiss that kind of thing with a wave of my hand.


----------



## 71 dB

Brahmsian said:


> I don't have a balloon that can be popped, just as I don't have a tail to tuck between my legs. However, it's true that I've left here disgusted at the hostility of some on this thread. They're like chimps trying to rip off the face of anyone who enters their territory. But some of those same people have been very nice on other occasions, so I don't draw any conclusions.
> 
> As for the placebo illusion, people often imaginatively enhance their experiences. When you read a novel, your imagination fleshes out the picture beyond the words on the page. Children imagine that their action figures can kick some serious butt even while knowing that the figures aren't alive; the toy is just something their imagination can act upon. The power of the imagination to enhance one's experience is real, and the pleasure derived from it is real, which is one reason why people don't want to give up their illusions. I remain agnostic on the matter. But being agnostic is not good enough; you're expected to either toe the line or not make a peep. There are totalitarian states that have more freedom of speech than this thread.
> 
> ...


Respecting other people is always a good thing, but respect doesn't mean respecting poor _opinions_. As you say yourself, people here have been nice to you on other occacions. That shows that people do respect others here, but not every single opinion of every single person is respected. People may not show respect to your opinions about hi-rez (because they are not very '_suitable_' for this section of this discussion board), but the same people may respect your opinions about classical music for example. Perhaps you are the undeniable Brahms-guru around? We just aren't talking about Brahms' symphonies here so that hasn't come up.

If you want imagination enhance your experience that's fine, but this is not really the place for doing that. This is the place were you get to know if the tooth fairy really exists so children who want to keep believing in the tooth fairy better stay away! Your freedom of speech isn't being violated unless the moderators have removed or edited your posts. Even if they have done that, it is totally normal that discussion boards have their specific rules about what is allowed and what is not. We are not allowed to discuss politics here for example. Freedom of speech doesn't mean freedom from being criticized or being given counter-arguments. You get to present your views and others get to comment on them. Freedom of speech doesn't mean others have to just accept and agree with any opinion no matter how factually wrong it is.

I do agree about that modern recordings can be stunning, but it isn't because of extreme bitrates or bitstream technology. You can make stunning recordings at 44.1 kHz PCM. 24 bits gives flexibility and safety margin in the music _production_, but for consumers of music the final product can be easily fitted into 16 bit format, because about 13 bits is enough.


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## castleofargh (May 9, 2021)

Brahmsian said:


> I don't have a balloon that can be popped, just as I don't have a tail to tuck between my legs. However, it's true that I've left here disgusted at the hostility of some on this thread. They're like chimps trying to rip off the face of anyone who enters their territory. But some of those same people have been very nice on other occasions, so I don't draw any conclusions.
> 
> As for the placebo illusion, people often imaginatively enhance their experiences. When you read a novel, your imagination fleshes out the picture beyond the words on the page. Children imagine that their action figures can kick some serious butt even while knowing that the figures aren't alive; the toy is just something their imagination can act upon. The power of the imagination to enhance one's experience is real, and the pleasure derived from it is real, which is one reason why people don't want to give up their illusions. I remain agnostic on the matter. But being agnostic is not good enough; you're expected to either toe the line or not make a peep. There are totalitarian states that have more freedom of speech than this thread.
> 
> ...


There is, or at least there should be a clear line between statements of objective facts and opinions/impressions. 
That line when it comes to objective facts should indeed remain strictly defined and what can be said about those facts should clearly be stated as opinions or hypotheses in the absence of clear demonstration. I believe that anybody and any place doing it differently is wrong and potentially dangerous to somebody(so... everywhere...).
Now, many people will write claims while not really thinking about it, or while having very peculiar ideas about what constitutes proof of an event. Like most people thinking, ”I heard it, what more proof can I need?”
And yes we also have it happen the other way around, when reacting to an opinion as if it was an objective claim. Your last posts were carefully written but were no doubt read by some as claims about hires or whatever else. It will happen, we're humans and we all interpret massively. Plus, English isn't everybody's native language. I hope that I can use that as an excuse next time I jump on somebody for no reason.^_^

 I’m completely fine with statements that hires makes someone feel better for reasons mostly unknown. I still do enjoy my overall audio experience more when I use the prettier lod cable in my DAC. I just pay attention to never let people think that the sound itself is better. But it sure is easy to slip or get misunderstood. Because after all, I do feel like the sound is better. But I can't demonstrate that I hear a difference in a blind test, and measurements actually favor another short cable I made myself(so of course it's ugly). I then decide that those are supporting evidence of me making up ”audio” changes based on what I see. Another guy will face the same data and decide that he can't be wrong about his impression so blind tests are flawed and we can't measure everything there is to hear in music. Most likely some silly excuse seeking self validation. But ultimately it's all a matter of how much confidence we put in a given piece of information. And both of us will happily use the things that make us feel the best, regardless of who's right about the objective facts. That's fine by me. But I may rip your face off if you make unlikely claims you can't support. That is where I put my line, or try to at least.

Masters are masters, when a good master is only available in a specific(overpriced?) hires format, TBH it just makes me hate that format and the entire industry behind it. That's partly how I came to hate dsd even though it's perfectly capable of carrying quality audio.
In conclusion, happiness and truth won't always walk hand in hand. This section can discuss both but should not let people assume that they're automatically one and the same.


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## bigshot

All opinions are not created equal. Some are built on a foundation of critical thinking and are supported by facts, and others are subjective impressions based on emotions, perceptual error or bias. I know in internet forums the idea that everyone's opinion is valid is the general rule, but that's why a lot of opinions in internet forums aren't worth the time it takes to type or read them.

People can believe things that aren't true and spend money on things that don't buy them any better sound. That is perfectly fine to me. You have the God given right to be wasteful and foolish. Go for it. Just don't try to convince other people that your personal foolishness is valid. That crosses the line between foolishness and deception, even if the intent is just to self-validate.


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## peskypesky

71 dB said:


> The purpose of this sound science section is to help people to learn and get convinced of what science teaches us and have immunity against being fooled by snake oil sellers.



Exactly.


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## peskypesky

bigshot said:


> More is not always better. Sometimes it's just more.


Especially when the more is frequency data that lies beyond the range of human perception.


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## 71 dB

peskypesky said:


> Especially when the more is frequency data that lies beyond the range of human perception.


This discussion about sampling frequencies is off topic in this thread, but it is important nevertheless. Content relevant in this thread marked blue.

There seems to be four main claims justifying "benefits" of high ( more than 48 kHz ) sampling frequencies:

*(1) The claim that ultrasonic content alters the way we hear/experience sounds on the frequency range of our hearing.* Sure, non-linearities in our hearing can _theoretically_ produce difference frequencies to be heard. However, if we analyse high sample rate recordings of acoustic instruments we notice the ultrasonic content is typically very weak, because that's how natural sounds are build: The amplitudes of the harmonic components get smaller and smaller with frequency. Our hearing system is an effective low pass filter mechanically and also the non-linearities aren't very sharp (no digital clipping). This means that any difference frequencies caused by non-linearities of our hearing will be extremely weak and probably totally masked by the music itself. Also, intermodulation distortion is very non-musical in nature because it works completely against the harmonic structure of music. So, even if we heard these products caused by non-linearities, we might want to get rid of them filtering ultrasonics away for a _cleaner _more musical sound. The ultrasonic content in hi-rez music tends to be very uncontrolled: It contents noise and other interferences that have nothing to do with the music itself. This is not a mystery: Ultrasonic content is very difficult to control, because we don't hear it!

*(2) The claim that higher sample rates give more accurate (less staircase-like) signal.* This is the "easiest" claim to debunk by referring to the sampling theory, but it is unfortunately difficult to explain to someone who doesn't understand digital signals well. It is unfortunate, that digital audio signals are illustrated graphically using _staircase-signals_. Those mental images are hard to unlearn. It is important for people to understand that digital audio is based on band-limited signals required by the sampling theory. Band-limited signals don't know about stair-cases! This claim is also used to justify (wrongly again) bigger bit depth. Dither is used to make sure the samples represent 100 % accurately the signal. We just have to accept the noise floor determined by the bit depth. We simply use as many bits per sample as we need to achieve low enough noise floor.  

*(3) The claim that higher sample rates increased temporal accuracy of the music including more accurate stereo image/soundstage.* Again, this is due to lack of understanding digital signals. Sampling frequency doesn't limit temporal accuracy at all. It only limits the bandwidth of allowed frequencies. We don't need to move signals in time by steps of samples. Because the signals are band-limited, we can "slide" the signal like a snake any amount of time we want. In fact digital filters do this all the time causing frequency dependent non-linear phase shifts (unless we use linear phase filter). Theoretically _bit depth_ limits temporal accuracy, but in practice so little the effect is meaningless when talking about the accuracy of human hearing. Also, if we make a 16 bit version of a 24 bit original version using proper dithering, we have the temporal accuracy of the 24 bit version in the 16 bit version. All we have to pay is raised noise floor, and as I keep telling people no more than 13 bits is what is needed in music listening so having the noise floor set by 16 bits is not an issue at all for music consumers. 

*(4) The claim that higher sample rates allow more relaxed anti-alias filters.* This is perhaps the most valid point, but it doesn't mean we need to use very high sampling frequencies. About 60 kHz is enough to exhaust all the benefits of "relaxed" anti-alias filters. So, this claim doesn't really justify even the sampling frequencies of 88.2 kHz and 96 kHz. Even if this claim has _some_ validity in it, it is a very tiny issue (based on listening tests) and it justifies sampling frequencies around 60 kHz rather than 96 kHz and higher.


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## Arjey

Can freely convert flac to wav and wav to flac without any consequences? I mean, both are lossless, the only difference is that flac is compressed (still lossless), which means it takes less storage space, but at the same time uses more processing power to play it.

The thing is, I have an mp3 player.. that heats up whenever I play heavy (24bit or higher than 44.1khz) files, it even heats up a bit when I play normal flac 16/44.1.. and when it heats up the microcontroller starts clicking the "right" button on its own 😅 so I was thinking of converting everything to wav (16/44.1) so it wouldn't heat up.

Also, can I convert mp3 to flac/wav without losing quality? I know that I shouldn't convert it back to lossy, or lossy->lossy, but just lossy to lossless should be fine, right?


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## Vamp898

You can convert from lossless to lossless as often as you want. You never lose something (thats why it's lossless), so it doesn't matter how often as long its the same Format.

If you convert 96 KHz 24bit into 44.1 KHz 16bit --> you loose something because you're not just compressing, you're also converting.

AAC (or as people from the 90s say, MP3) is lossy compressed. _Every_ compression looses quality.

Lossy --> Lossless doesn't loose quality, but only has disadvantages, i'd not recommend it.


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## 71 dB

Arjey said:


> Can freely convert flac to wav and wav to flac without any consequences?


The only consequences are wasted time and energy.



Arjey said:


> I mean, both are lossless, the only difference is that flac is compressed (still lossless), which means it takes less storage space, but at the same time uses more processing power to play it.


Yes. Flac just has less redundant information. 



Arjey said:


> The thing is, I have an mp3 player.. that heats up whenever I play heavy (24bit or higher than 44.1khz) files, it even heats up a bit when I play normal flac 16/44.1.. and when it heats up the microcontroller starts clicking the "right" button on its own 😅 so I was thinking of converting everything to wav (16/44.1) so it wouldn't heat up.


Well, that is worth trying. Before converting everything convert say one album and test with it if it solves the heat problem.



Arjey said:


> Also, can I convert mp3 to flac/wav without losing quality? I know that I shouldn't convert it back to lossy, or lossy->lossy, but just lossy to lossless should be fine, right?


You never lose quality when converting into a lossless format unless there is also change in sampling frequency.


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## Slaphead

Vamp898 said:


> If you convert 96 KHz 24bit into 44.1 KHz 16bit --> you loose something because you're not just compressing, you're also converting.


You're not technically compressing, you're downsampling. Basically you're just dumping data, some of which may potentially be relevant. The idea of compression is reduce the file size in one of two ways, either not to lose data (FLAC, ALAC, APE, etc), or to only lose data that's judged by the compression algorithm to not be relevant (MP3, AAC, SBC, etc)


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## old tech

71 dB said:


> This discussion about sampling frequencies is off topic in this thread, but it is important nevertheless. Content relevant in this thread marked blue.
> 
> There seems to be four main claims justifying "benefits" of high ( more than 48 kHz ) sampling frequencies:
> 
> *(1) The claim that ultrasonic content alters the way we hear/experience sounds on the frequency range of our hearing.* Sure, non-linearities in our hearing can _theoretically_ produce difference frequencies to be heard. However, if we analyse high sample rate recordings of acoustic instruments we notice the ultrasonic content is typically very weak, because that's how natural sounds are build: The amplitudes of the harmonic components get smaller and smaller with frequency. Our hearing system is an effective low pass filter mechanically and also the non-linearities aren't very sharp (no digital clipping). This means that any difference frequencies caused by non-linearities of our hearing will be extremely weak and probably totally masked by the music itself. Also, intermodulation distortion is very non-musical in nature because it works completely against the harmonic structure of music. So, even if we heard these products caused by non-linearities, we might want to get rid of them filtering ultrasonics away for a _cleaner _more musical sound. The ultrasonic content in hi-rez music tends to be very uncontrolled: It contents noise and other interferences that have nothing to do with the music itself. This is not a mystery: Ultrasonic content is very difficult to control, because we don't hear it!


All good points but with point (1) while I agree with what you say, the claim can be dismissed for any recordings even before considering the other aspects which nullify that claim. If the ultrasonic does alter the sound we hear in some way then surely this is only relevant to a live acoustic event? As (with this claim) the altered sound is altering the sound we hear then by definition the altered sound is within the human range of hearing so any recording of that sound would have the altered sound baked into the recording. In fact, if ultrasonics do actually effect the sound we hear then we really should be filtering them out so it doesn't double down on altering the sound (ie the already baked in part plus the addition of play backing the ultrasonic content).


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## sander99

old tech said:


> All good points but with point (1) while I agree with what you say, the claim can be dismissed for any recordings even before considering the other aspects which nullify that claim. If the ultrasonic does alter the sound we hear in some way then surely this is only relevant to a live acoustic event? As (with this claim) the altered sound is altering the sound we hear then by definition the altered sound is within the human range of hearing so any recording of that sound would have the altered sound baked into the recording.


Not when the non-linearities producing difference frequencies are somewhere within the human hearing system itself, somewhere in our ears. For example in the "mechanics" of ear drums, malleus, incus, and stapes. So this hasn't happened yet to the sound that is on a recording.
I assume this is what @71 dB meant?

However, shouldn't this phenomenon have showed in controlled listening tests if it really existed?


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## Vamp898

Slaphead said:


> You're not technically compressing, you're downsampling. Basically you're just dumping data, some of which may potentially be relevant. The idea of compression is reduce the file size in one of two ways, either not to lose data (FLAC, ALAC, APE, etc), or to only lose data that's judged by the compression algorithm to not be relevant (MP3, AAC, SBC, etc)


He goes from a 96KHz 24bit WAV to an 44.1KHz 16bit FLAC.

So he is compressing and downsampling at the same time (where the compression is lossless, but the downsampling isn't)


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## Vamp898

old tech said:


> All good points but with point (1) while I agree with what you say, the claim can be dismissed for any recordings even before considering the other aspects which nullify that claim. If the ultrasonic does alter the sound we hear in some way then surely this is only relevant to a live acoustic event? As (with this claim) the altered sound is altering the sound we hear then by definition the altered sound is within the human range of hearing so any recording of that sound would have the altered sound baked into the recording. In fact, if ultrasonics do actually effect the sound we hear then we really should be filtering them out so it doesn't double down on altering the sound (ie the already baked in part plus the addition of play backing the ultrasonic content).


Yes and no.

Its our ears job to filter that out. You want to hear the recording exactly the way it is.

Its the same with cameras. Our eyes are most sensitive for green and less for red and blue. Thats why Bayer Sensors have 50% Green, 25% Red and 25% Blue.

Foveon Sensors showed, that this is nonsense because you're double filtering. You're filtering on the sensor and then your eyes filters again and that causes a visible loss in overall image color and quality.

You have to record 100% Green, 100% Red and 100% Blue because the real world looks like that and then let the eye filter.

The same with Sound. If the real sound in the concert hall contained ultrasonic sounds that are altering the original sound, and you want to hear the sound exactly like it was in the concert hall, you have to record and playback them.

But its not just ultra sonic sounds, the sample rate does have two effects and the second is much more important. The Resolution.

The Higher the sample rate, the closer the digital Signal is to the original analog signal.

If you playback with 96KHz, you are able to play sounds up to 48 KHz, but you are also able to play a sound in the 10 KHz region with double the accuracy.

My personal experience with some songs (but i don't know the cause) is that the CD-Version has worse layering. At Qobuz for example you have the choice between CD Audio Quality and Hi-Res Audio sometimes and there are (extremely minor) differences in the layering in busy songs.

But that could have other causes as well and may or may not be related to 96KHz at all. Sometimes the Hi-Res Audio Version is just mastered and mixed less pop-like because they are targeting a different audience.

The producer 松任谷正隆 (who produced albums for Nora Jones and Steely Dan) said that Hi-Res Audio is one of the best ideas because he always felt pain when downsampling albums to CD Audio Quality because they loose their layering (and that was in 2000/2002)

So producers in the 2000s already were sure there are differences and he also pointed out samples with exact time stamps where you could hear a difference and what the difference is.

I am pretty sure that 99,9% of people will never be able to tell any difference at all and i personally think, it just doesn't matter at all.

But if i have the chance, i always go for Hi-Res Audio. In most cases its the identical price (Bandcamp for examples charges nothing for Hi-Res Audio) and we live in a Time where TB SD-Cards exist. So why risk in the first place. There are samples that proof that there can be a difference and even if i will never be able to hear it, there is this peace of calm like "You did your best, if this is not good enough, there is nothing you can do, just enjoy it".


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## gregorio

Vamp898 said:


> [1] But its not just ultra sonic sounds, the sample rate does have two effects and the second is much more important. The Resolution.
> The Higher the sample rate, the closer the digital Signal is to the original analog signal.
> If you playback with 96KHz, you are able to play sounds up to 48 KHz, but you are also able to play a sound in the 10 KHz region with double the accuracy.
> 
> ...


1. The sample rate does NOT affect the resolution, the reconstructed digital signal is NOT closer to the original input (analogue) signal and sounds "in the 10kHz region" do NOT have "double the accuracy", they have exactly the same accuracy/resolution regardless of whether a 96kHz, 48kHz or 44.1kHz sample rate is used. Please read the original post; for your statements to be true, the proven Nyquist-Shannon Sampling Theorem would have to be false and digital audio would not exist!

2. No, producers in the 2000's were NOT "sure there are differences" or more accurately, there were some who didn't understand the basics of how digital audio works (and there still are some) and thought they heard differences where there weren't any differences. There were also some specific circumstances in the 2000s where higher than 48kHz sample rates could make an audible difference during certain mixing/processes (but not in the final master).

3. As there is no difference, then obviously no one can "tell any difference". Although of course some people can experience a difference (where there isn't one) due to some placebo/expectation bias.



Vamp898 said:


> If the real sound in the concert hall contained ultrasonic sounds that are altering the original sound, and you want to hear the sound exactly like it was in the concert hall, you have to record and playback them.



No, you don't. For addition frequencies to make a difference to what you hear, the additional frequencies have to be audible. By definition, ultrasonic frequencies are inaudible and therefore do not make a difference to what can be heard. This is very easy to test and has been tested countless times for many decades. The only exception to this is in the case of IMD (inter-modulation distortion) but the product/result of IMD must again be within the audible spectrum, which again, would be captured at 44.1/48kHz sample rates.



Vamp898 said:


> At Qobuz for example you have the choice between CD Audio Quality and Hi-Res Audio sometimes and there are (extremely minor) differences in the layering in busy songs.
> But that could have other causes as well and may or may not be related to 96KHz at all. Sometimes the Hi-Res Audio Version is just mastered and mixed less pop-like because they are targeting a different audience.



Exactly, Hi-Res and CD versions are sometimes mastered differently, in which case, you are comparing different masters which have nothing to do with the sample rate. In other words, you could take the Hi-Res version, downsample it to CD format (16/44.1) and the result would be audibly identical. Again, this has been tested countless times over several decades.

G


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## castleofargh (Aug 13, 2021)

Vamp898 said:


> Yes and no.
> 
> Its our ears job to filter that out. You want to hear the recording exactly the way it is.
> 
> ...


The sensor argument makes it look like the result was twice as green ^_^. Not being able to capture stuff as accurately as they wanted, they gave more of the sensor’s surface to capture what mattered more to human eyes. If you just plain dismiss tech limitations such as sensitivity of the sensors, then sure enough your ideals are correct.

To stay on camera analogy, do you wish for 100% of the original UV light too?
With cameras, UV light has ruined more shots than we can count in a very obvious way. So it wasn't hard to decide on what to do. In audio, for many reasons, there is very little, if any impact from having the low amplitude ultrasounds that usually remain. So not only wasn't it obvious that ultrasounds were bad, people actually started to replace the lack of actual evidence with ideals like, ”More is better”, and the points you made.
The typical lack of significance for our ears is ironically a main reason why audiophiles can keep on making a big deal out of hires. That, and the tradition of listening with our eyes. And also decades of marketing efforts to convince us that the bigger box sounded night and day better, despite decades of failing to prove it.
IMO, it matters so little in practice, that it's alright to use whatever format and resolutions we like.


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## 71 dB

old tech said:


> All good points but with point (1) while I agree with what you say, the claim can be dismissed for any recordings even before considering the other aspects which nullify that claim. If the ultrasonic does alter the sound we hear in some way then surely this is only relevant to a live acoustic event? As (with this claim) the altered sound is altering the sound we hear then by definition the altered sound is within the human range of hearing so any recording of that sound would have the altered sound baked into the recording. In fact, if ultrasonics do actually effect the sound we hear then we really should be filtering them out so it doesn't double down on altering the sound (ie the already baked in part plus the addition of play backing the ultrasonic content).


We must be careful of what we are talking about.

When listening to live sound, ultrasonics exists, but are typically quite weak and ultrasonic sounds attenuate fast in air *. So by the time the ultrasonics reach the listeners in a concert hall they are even weaker. Theoretically they can create stimulus in our ears at audible frequencies, but most probably all of that gets masked by auditory masking so that what we actually hear is not different compared to the situation where the ultrasonics didn't exist. That's why a recording say at 44.1 kHz lacking all the ultrasonics isn't going to give different (lesser) perceived sound. I believe you can demonstrate any kind of alteration caused by ultrasonic content to the perceived sound by using test signals that maximize the phenomenon and minimize masking. Music is not like that. 

Our hearing has developping so that frequencies above 20 kHz simple do not matter. If we needed to hear higher frequencies our hearing would be different and we would hear say up to 50 kHz (but our ability to hear bass frequencies would probably be worse). In fact I'd say the _relevant_ range of human hearing ends at around 10 kHz and the 10 kHz - 20 kHz octave is just transition away from the relevant range. That explains why our hearing loses its ability in this range as we age and why filtering out all frequencies above 10 kHz reduces perceived  sound quality so little. 

* At 20°C/air humidity 50 % the absorption is:

10 kHz: 0.16 dB/m
20 kHz: 0.52 dB/m
30 kHz: 0.94 dB/m
40 kHz: 1.32 dB/m
50 kHz: 1.66 dB/m


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## Davesrose (Aug 13, 2021)

Vamp898 said:


> Yes and no.
> 
> Its our ears job to filter that out. You want to hear the recording exactly the way it is.
> 
> ...


First of all, not all light is %100 R and G and B (or for that matter, the rest of ROYGBIV: and it can also have other spectrums).

Secondly, it's not just a Bayer pattern that has one channel carrying more resolution then other channels (component video is another example).  It can be more efficient to set one channel as a luminance channel (which can be weighted for the aspect of an image that requires more bandwidth: luminance or dynamic range) and then have others be chroma.  This was especially important with early sensors, which had limited dynamic range and resolution.  They also had dyes for RGB (where Bayer originally wanted CMY: which a few new sensors are now able to have CMY for improved light absorption/quantum efficiency).

A foveon sensor works on the principles of light.  It has the same number of color pixels, each layered on top of one another.  It is layered in a manner similar to spectrum of light: B->G->R.  One of the main advantages of the foveon sensor is that it may not exhibit demosaicing artifacts that a Bayer sensor could impart (but given that cameras now have high resolutions, this is fairly mitigated).


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## Vamp898

Davesrose said:


> First of all, not all light is %100 R and G and B (or for that matter, the rest of ROYGBIV: and it can also have other spectrums).
> 
> Secondly, it's not just a Bayer pattern that has one channel carrying more resolution then other channels (component video is another example).  It can be more efficient to set one channel as a luminance channel (which can be weighted for the aspect of an image that requires more bandwidth: luminance or dynamic range) and then have others be chroma.  This was especially important with early sensors, which had limited dynamic range and resolution.  They also had dyes for RGB (where Bayer originally wanted CMY: which a few new sensors are now able to have CMY for improved light absorption/quantum efficiency).
> 
> A foveon sensor works on the principles of light.  It has the same number of color pixels, each layered on top of one another.  It is layered in a manner similar to spectrum of light: B->G->R.  One of the main advantages of the foveon sensor is that it may not exhibit demosaicing artifacts that a Bayer sensor could impart (but given that cameras now have high resolutions, this is fairly mitigated).


The FOVEON sensor captures all wavelengths (to different amounts) on all three layers (that's why the Quattro design works)


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## Vamp898 (Aug 13, 2021)

gregorio said:


> 1. The sample rate does NOT affect the resolution, the reconstructed digital signal is NOT closer to the original input (analogue) signal and sounds "in the 10kHz region" do NOT have "double the accuracy", they have exactly the same accuracy/resolution regardless of whether a 96kHz, 48kHz or 44.1kHz sample rate is used. Please read the original post; for your statements to be true, the proven Nyquist-Shannon Sampling Theorem would have to be false and digital audio would not exist!
> 
> 2. No, producers in the 2000's were NOT "sure there are differences" or more accurately, there were some who didn't understand the basics of how digital audio works (and there still are some) and thought they heard differences where there weren't any differences. There were also some specific circumstances in the 2000s where higher than 48kHz sample rates could make an audible difference during certain mixing/processes (but not in the final master).
> 
> ...


I just checked that by recording a bass and played in the 100-200 Hz Reagion.

I recorded with 8 KHz 16bit and i recorded with 44.1 KHz 16bit

The 8 KHz sounds like crap, total garbage. Like someone is playing bass over an old telephone. The 44.1 KHz recording sounds excellent.

By your explanation, there should be no hearable difference below the 4000 KHz region, why do 100-200 Hz sounds, sound like crap when recorded with 8 KHz which should be way more than needed.

Please explain.


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## sonitus mirus

Vamp898 said:


> I just checked that by recording a bass and played in the 100-200 Hz Reagion.
> 
> I recorded with 8 KHz 16bit and i recorded with 44.1 KHz 16bit
> 
> ...



All anyone has to do is refer to the well-established Nyquist-Shannon sampling theorem.  You are the person that needs to explain a bit more about what is being done and to provide some evidence.  Are you certain only 100-200 Hz frequencies are being recorded and played back?  Do you have the ability to show us visual examples or to include the audio files being tested?


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## Vamp898

Vamp898 said:


> I just checked that by recording a bass and played in the 100-200 Hz Reagion.
> 
> I recorded with 8 KHz 16bit and i recorded with 44.1 KHz 16bit
> 
> ...


I answered my own question by looking up wikipedia.


sonitus mirus said:


> All anyone has to do is refer to the well-established Nyquist-Shannon sampling theorem.  You are the person that needs to explain a bit more about what is being done and to provide some evidence.  Are you certain only 100-200 Hz frequencies are being recorded and played back?  Do you have the ability to show us visual examples or to include the audio files being tested?


Ah i looked it up, now i understand.

Its just a theorem, it doesn't exist in the real world. It exists in math, if it would exist in the real world, it would not output any sound as it would have an infinite delay.


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## KeithPhantom

Vamp898 said:


> Its just a theorem


That is the best part. It is not open to discussion. Theorems are mathematically proven, also physically. If *you *don't like it, *you provide the evidence. You have the burden of proof.*


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## sander99

Vamp898 said:


> I just checked that by recording a bass and played in the 100-200 Hz Reagion.
> 
> I recorded with 8 KHz 16bit and i recorded with 44.1 KHz 16bit
> 
> ...


Very simple, I assume the sound of the bass has harmonics exceeding 4 kHz. At 44.1 kHz sampling rate all audible harmonics are captured and reproduced. At 8 kHz sampling rate they are not.
I guess you didn't read this:
https://www.head-fi.org/threads/dsd64-noise-issue.958698/page-6#post-16478626


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## Vamp898

KeithPhantom said:


> That is the best part. It is not open to discussion. Theorems are mathematically proven, also physically. If *you *don't like it, *you provide the evidence. You have the burden of proof.*


I'll take some screenshots in audacity later.

I already did a half year ago (in this very same thread afair) which proved that there is a difference from 96 KHz to 44.1 KHz.

It was inaudible, but it was there.

I never said it makes sense to keep the information and I made clear back then, that there difference is inaudible, but it's there.

Also is not my point to proof it.

Wikipedia says the theorem works only with an ideal filter and Wikipedia says an ideal filter does not exist because it would have infinite delay and so not produce any sound at all.

Quote: Real-time filters can only approximate this ideal, since an ideal sinc filter (a.k.a. rectangular filter) is non-causal and has an infinite delay, but it is commonly found in conceptual demonstrations or proofs, such as the sampling theorem and the Whittaker–Shannon interpolation formula.

I'd say it's rather your job to proof Wikipedia wrong than my to show they are right.


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## Vamp898

sander99 said:


> Very simple, I assume the sound of the bass has harmonics exceeding 4 kHz. At 44.1 kHz sampling rate all audible harmonics are captured and reproduced. At 8 kHz sampling rate they are not.
> I guess you didn't read this:
> https://www.head-fi.org/threads/dsd64-noise-issue.958698/page-6#post-16478626


I'm talking about PCM, not DSD. It's this thread still relevant?


----------



## 71 dB

Vamp898 said:


> I just checked that by recording a bass and played in the 100-200 Hz Reagion.
> 
> I recorded with 8 KHz 16bit and i recorded with 44.1 KHz 16bit
> 
> ...


Well, 8 kHz sampling rate IS telephone quality! No wonder it sounds like telephone. The problem here is probably mostly the initial transients of each note. Those attacks must contain a lot of frequencies above 4 kHz (considering the anti-alias filter, the usable bandwidth is about 3.5 kHz). When those important parts are filtered that way, the resulting sound is really lame and bad. Of course the harmonics contain frequencies above 200 Hz, but after the initial attack not so much to make huge difference.


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## Davesrose

Vamp898 said:


> The FOVEON sensor captures all wavelengths (to different amounts) on all three layers (that's why the Quattro design works)


No, you're still ignoring that camera sensors are designed to the capture visible spectrums of light (and since light is not always white, saturation point doesn't always reach 100% with B and G and R).  The main reason why the foveon sensor is layered B->G->R is their respective wavelengths (and to record one color on a pixel, instead of Bayer sensor which needs to demosiac sub-pixels).  If the light is full spectrum, B: 440 - 490 nm, G: 490 - 570 nm, R: 620 - 780 nm (because there's still a step to full 400-780, there's still processing involved with determining color values).


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## sander99

Vamp898 said:


> I'm talking about PCM, not DSD. It's this thread still relevant?


I was talking about PCM in the post that I linked. And sound in general. What I wanted to explain is that also every "note" on the bass is composed of multiple frequencies. The fundamental plus harmonics. The harmonics are what makes different instruments sound different even if they play the same note in the same octave. You have to capture all audible harmonics to get good sound, which is what you do at 44.1 kHz sampling rate, but not at 8 kHz sampling rate. And you don't need to capture harmonics that are not audible, so you don't need 96 kHz sampling rate.


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## gregorio

Vamp898 said:


> By your explanation, there should be no hearable difference below the 4000 KHz region



It's not my explanation, it's the proven facts, both mathematically and in practice!



Vamp898 said:


> I just checked that by recording a bass and played in the 100-200 Hz Reagion.
> 
> why do 100-200 Hz sounds, sound like crap when recorded with 8 KHz which should be way more than needed.
> 
> Please explain.



Sure: As others have said, you did NOT play in the 100-200Hz region! Only the fundamental pitch was in the 100-200Hz range, not the harmonics and assuming it's an electric bass, there's also considerable noise and distortion produced by the amp/cab that extends well beyond 4kHz and is part of the "sound". Removing all that sound, that's not only well within the audible spectrum but right next to the most sensitive part of it, will obviously sound significantly different!

Therefore your statement is incorrect, 100-200Hz sounds do NOT "sound like crap" when recorded with 8kFs/S, they sound exactly the same.



Vamp898 said:


> Wikipedia says the theorem works only with an ideal filter ...



No it doesn't, please provide the quote which states that. The wiki quote you've already provided does NOT say the theorem "works only with an ideal filter"! What Wikipedia actually states is: "_Instead, some type of approximation of the sinc functions, finite in length, is used. The imperfections attributable to the approximation are known as interpolation error._"- So the question becomes how much interpolation error is introduced and is it enough to be audible? The demonstrated answer (again, both mathematically and in practice) is an insignificant amount and not even close. This was true even of fairly early ADCs and DACs, let alone modern/current ones!

G


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## old tech

71 dB said:


> We must be careful of what we are talking about.
> 
> When listening to live sound, ultrasonics exists, but are typically quite weak and ultrasonic sounds attenuate fast in air *. So by the time the ultrasonics reach the listeners in a concert hall they are even weaker. Theoretically they can create stimulus in our ears at audible frequencies, but most probably all of that gets masked by auditory masking so that what we actually hear is not different compared to the situation where the ultrasonics didn't exist. That's why a recording say at 44.1 kHz lacking all the ultrasonics isn't going to give different (lesser) perceived sound. I believe you can demonstrate any kind of alteration caused by ultrasonic content to the perceived sound by using test signals that maximize the phenomenon and minimize masking. Music is not like that.
> 
> ...


I think you may have misunderstood my post. I have no reason to doubt any of your points on the highly unlikelyhood of ultrasounds affecting sound we hear in the audible range. I was just pointing out that if it did, a recording would have captured that effect and therefore the actual untrasonics would be unecessary for playback. To be clear though, i certainly don't accept ulrasonic content in music affects frequencies we can hear (unless it is causing the playback gear to distort).


----------



## PhonoPhi

gregorio said:


> 1. The sample rate does NOT affect the resolution, the reconstructed digital signal is NOT closer to the original input (analogue) signal and sounds "in the 10kHz region" do NOT have "double the accuracy", they have exactly the same accuracy/resolution regardless of whether a 96kHz, 48kHz or 44.1kHz sample rate is used. Please read the original post; for your statements to be true, the proven Nyquist-Shannon Sampling Theorem would have to be false and digital audio would not exist!
> 
> 2. No, producers in the 2000's were NOT "sure there are differences" or more accurately, there were some who didn't understand the basics of how digital audio works (and there still are some) and thought they heard differences where there weren't any differences. There were also some specific circumstances in the 2000s where higher than 48kHz sample rates could make an audible difference during certain mixing/processes (but not in the final master).
> 
> ...


You are always so categorical, it is amazing.

What about an audio sample with the ultrasonic frequencies of 20.1 kHz, 40.2 kHz, 60.3 kHz, etc.
There should me the main tone reconstruction at 10.05 kHz that is perfectly audible - both from the famous experiments and the Fourier transform math. (Would 5.025 kHz be any perceptible is a good question for those knowledgable in the decay power of these series).

How then it is possible to claim that anything above 20 kHz does not matter for the audible range is not clear to me.


----------



## 71 dB

old tech said:


> I think you may have misunderstood my post. I have no reason to doubt any of your points on the highly unlikelyhood of ultrasounds affecting sound we hear in the audible range. I was just pointing out that if it did, a recording would have captured that effect and therefore the actual untrasonics would be unecessary for playback. To be clear though, i certainly don't accept ulrasonic content in music affects frequencies we can hear (unless it is causing the playback gear to distort).


Sorry about misunderstanding your post.

If the recording is done at high sample rate (88.2 kHz or higher) the ulrasonic content is likely to be on the recording if the microphones are able to pick them up. Nothing weird happens at audible range at this point. Only when the high-res file is played something can happen (audio gear or ear may distort the ultrasonics and generate stuff at audible frequencies).

If the recording is done at 44.1 kHz, all the ultrasonic stuff has to be filtered out before AD conversion. This means the ultrasonics never get to do anything. It is like they never existed.

Since we have established and we agree about that the theoretical effects of ultrasonic are unlikely at best, it doesn't matter whether they are on the recording or not. That's the reason why 44.1 kHz is enough in digital audio for consumers.


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## gregorio

71 dB said:


> When listening to live sound, ultrasonics exists, but are typically quite weak and ultrasonic sounds attenuate fast in air *. So by the time the ultrasonics reach the listeners in a concert hall they are even weaker. ...
> 
> * At 20°C/air humidity 50 % the absorption is:
> 
> ...



Careful here, the figures you've quoted aren't the signal attenuation in air, they are the ADDITIONAL attenuation due to high frequency air dissipation. Your quoted figures have to be added to the signal attenuation due to distance. 

For example, let's say we have an instrument producing a very loud 30kHz harmonic, say 50dBSPL at 1m. By the time that harmonic had reached an audience member 10 meters further away (11m in total) it would be attenuated by about 20.8dBSPL according to the inverse square law, plus the additional 9.4dB you've quoted for high frequency air dissipation, a total attenuation of about 30.2dB. Our loud 50dB 30kHz harmonic would be a far quieter 19.8dB by the time it reached our audience member. 

There's only a few instruments capable of producing a 50dB harmonic that high (metal perc instruments) and in practice, I wouldn't be surprised if it were below 10dB at our audience position because we haven't accounted for the adsorption caused by other factors, EG. Chairs, musicians, other audience members, etc. For the vast majority of instruments, any similarly high frequency harmonics are going to be much lower level to start with and therefore 0dB or lower at the audience position!

G


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## Slaphead

Vamp898 said:


> The 8 KHz sounds like crap, total garbage. Like someone is playing bass over an old telephone. The 44.1 KHz recording sounds excellent.


There's a good reason for that - 8 KHz was the sample rate chosen for the original GSM mobile phones (2G phones). The reason is that speech is well covered within the 4 KHz range. To use more would have stretched the networks of the time, and would have reduced the number of concurrent connections to each cell tower.


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## audiokangaroo (Aug 14, 2021)

sander99 said:


> I was talking about PCM in the post that I linked. And sound in general. What I wanted to explain is that also every "note" on the bass is composed of multiple frequencies. The fundamental plus harmonics. The harmonics are what makes different instruments sound different even if they play the same note in the same octave. You have to capture all audible harmonics to get good sound, which is what you do at 44.1 kHz sampling rate, but not at 8 kHz sampling rate. And you don't need to capture harmonics that are not audible, so you don't need 96 kHz sampling rate.


Capturing all audible frequencies is not sufficient if you want to achieve high fidelity. You have to capture also the information produced by the combination of these audible frequencies in the time domain, which is a mathematical issue. In fact, we have to capture the analogue waveform as accurately as possible. This problem is beyond the sampling theorem, which is only about capturing audible frequencies, without caring about the way these frequencies are set together inside the global signal.


----------



## KeithPhantom (Aug 14, 2021)

audiokangaroo said:


> This problem is beyond the sampling theorem, which is only about capturing audible frequencies, without caring about the way these frequencies are set together inside the global signal.


Eh, no.

First, frequency and time domain are only two different representations of the same thing, this is stated by the Fourier Series. Second, Nyquist-Shannon applies to *any* set of frequencies, not just in the audible range. It is a mathematical fact. Third, when recording a frequency range, we record everything in between that range and then low-pass when it isn’t needed. Even if you do not low pass at the recording and filter before Fs/2 (what any competent reconstruction filter would do, being this needed by the sampling theorem), you will have a perfect reconstruction of signals below Fs/2 plus quantization errors (due to the limited precision of digital values) that will average as noise, and can be noise shaped to increase SNR.


----------



## audiokangaroo (Aug 14, 2021)

KeithPhantom said:


> Eh, no.
> 
> First, frequency and time domain are only two different representations of the same thing, this is stated by the Fourier Series. Second, Nyquist-Shannon applies to *any* set of frequencies, not just in the audible range. It is a mathematical fact. Third, when recording a frequency range, we record everything in between that range and then low-pass when it isn’t needed. Even if you do not low pass at the recording and filter before Fs/2 (what any competent reconstruction filter would do, being this needed by the sampling theorem), you will have a perfect reconstruction of signals below Fs/2 plus quantization errors (due to the limited precision of digital values) that will average as noise, and can be noise shaped to increase SNR.


We go from the time domain to the frequency domain with the Fourier transform. What we get with the Fourier transform are mathematical frequencies. They are a different thing than the audible physical frequencies that we can find in the time domain and that are band limited for audibility reasons.
Although the analogue waveform is made up with band limited frequencies, the product of the Fourier transform is not necessarily band limited. This can explain for instance why  192 KHz PCM can sound more accurate and natural than 44.1 kHz.
Experience shows that only capturing the waveform with a very high level of accuracy can make digital sound like analogue.
The Nyquist-Shannon theorem is still valid, but we have then to target a sampling band that goes much Higher than 22 KHz. Our target isn't the audible frequencies any more, but the Fourier frequencies.


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## 71 dB

gregorio said:


> Careful here, the figures you've quoted aren't the signal attenuation in air, they are the ADDITIONAL attenuation due to high frequency air dissipation. Your quoted figures have to be added to the signal attenuation due to distance.


Yes, of course!



gregorio said:


> For example, let's say we have an instrument producing a very loud 30kHz harmonic, say 50dBSPL at 1m. By the time that harmonic had reached an audience member 10 meters further away (11m in total) it would be attenuated by about 20.8dBSPL according to the inverse square law


Correct.



gregorio said:


> , plus the additional 9.4dB you've quoted for high frequency air dissipation, a total attenuation of about 30.2dB.


So true my friend.



gregorio said:


> Our loud 50dB 30kHz harmonic would be a far quieter 19.8dB by the time it reached our audience member.


The sound pressure level might be just a little above that due to hall acoustics. The surfaces don't reflect much 30 kHz, but they probably reflect _something_. So, maybe 20 dB or 21 dB in the end? Very very quiet nevertheless!



gregorio said:


> There's only a few instruments capable of producing a 50dB harmonic that high (metal perc instruments) and in practice, I wouldn't be surprised if it were below 10dB at our audience position because we haven't accounted for the adsorption caused by other factors, EG. Chairs, musicians, other audience members, etc. For the vast majority of instruments, any similarly high frequency harmonics are going to be much lower level to start with and therefore 0dB or lower at the audience position!
> 
> G


No disagreements here!


----------



## 71 dB

audiokangaroo said:


> We go from the time domain to the frequency domain with the Fourier transform. What we get with the Fourier transform are mathematical frequencies. They are a different thing than the audible physical frequencies that we can find in the time domain and that are band limited for audibility reasons.
> Although the analogue waveform is made up with band limited frequencies, the product of the Fourier transform is not necessarily band limited. This can explain for instance why  192 KHz PCM can sound more accurate and natural than 44.1 kHz.
> Experience shows that only capturing the waveform with a very high level of accuracy can make digital sound like analogue.
> The Nyquist-Shannon theorem is still valid, but we have then to target a sampling band that goes much Higher than 22 KHz. Our target isn't the audible frequencies any more, but the Fourier frequencies.


Infinitely long Fourier Transformation of a bandlimited signal (say 110 Hz sinewave) gives finite spectrum. Signals of finite length will give infinite spectrum, because finite signals can be considered as windowed in time so that the spectrum of the corresponding infinitely long signal gets convoluted with the spectrum of the time window which means the spectrum gets smeared from negative infinity to positive infinity. This is probably what you are trying to say here?

This is all math. In audio we need to deal with _practicalities_. Infinitely good audio gear cost infinitely. It is impossible to reach. We need limits for performance that we can consider "good enough" in relation to the properties of our hearing and other practicalities. Spectral smearing of time-limited signals is EXTREMELY small. Math can point it out for us. Our ears can't. It is insignificant. It doesn't matter. Anyone who says it does is lying or doesn't know what they are talking about. That's why the claims of 192 kHz PCM being more accurate and natural than 44.1 kHz are silly.

Analog sound is prone to all kind of distortions and noise. It has its limits. The whole idea behind digital audio it that we can do things accurately enough to _surpass_ analog sound, avoid the distortions and noise, especially when making copies of a recording. The reason why analog gear still being used is to have those signature distortions in the sound, to make it less accurate, but"warmer" if you will. If analog was superior, it would be used in everything, but it isn't. That's why the World went digital, and if 192 kHz was superior to 44.1 kHz or 48 kHz, pretty much everyone working with audio would use high sample rates, but that's not the case.


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## audiokangaroo

71 dB said:


> Infinitely long Fourier Transformation of a bandlimited signal (say 110 Hz sinewave) gives finite spectrum. Signals of finite length will give infinite spectrum, because finite signals can be considered as windowed in time so that the spectrum of the corresponding infinitely long signal gets convoluted with the spectrum of the time window which means the spectrum gets smeared from negative infinity to positive infinity. This is probably what you are trying to say here?
> 
> This is all math. In audio we need to deal with _practicalities_. Infinitely good audio gear cost infinitely. It is impossible to reach. We need limits for performance that we can consider "good enough" in relation to the properties of our hearing and other practicalities. Spectral smearing of time-limited signals is EXTREMELY small. Math can point it out for us. Our ears can't. It is insignificant. It doesn't matter. Anyone who says it does is lying or doesn't know what they are talking about. That's why the claims of 192 kHz PCM being more accurate and natural than 44.1 kHz are silly.
> 
> Analog sound is prone to all kind of distortions and noise. It has its limits. The whole idea behind digital audio it that we can do things accurately enough to _surpass_ analog sound, avoid the distortions and noise, especially when making copies of a recording. The reason why analog gear still being used is to have those signature distortions in the sound, to make it less accurate, but"warmer" if you will. If analog was superior, it would be used in everything, but it isn't. That's why the World went digital, and if 192 kHz was superior to 44.1 kHz or 48 kHz, pretty much everyone working with audio would use high sample rates, but that's not the case.


The actual signal is made up with components that belong to a limited band, but can we say that the combination of those elementary signals is actually band limited ?
The combination process creates information, and this new information can be non band limited. If the waveform is not band limited, then its spectrum through the Fourier Transformation will be unlimited.
Some people don't seem to hear the difference of sound between 44.1 and 192. Then they can compare with DSD, which is equivalent to PCM above 800 KHz sample rate. It sounds so much more natural that we must admit that something is lacking in 44.1 KHz PCM.


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## 71 dB (Aug 14, 2021)

audiokangaroo said:


> The actual signal is made up with components that belong to a limited band, but can we say that the combination of those elementary signals is actually band limited ?
> The combination process creates information, and this new information can be non band limited. If the waveform is not band limited, then its spectrum through the Fourier Transformation will be unlimited.
> Some people don't seem to hear the difference of sound between 44.1 and 192. Then they can compare with DSD, which is equivalent to PCM above 800 KHz sample rate. It sounds so much more natural that we must admit that something is lacking in 44.1 KHz PCM.


Combination of frequencies within a limited band is also band-limited! If I put together 100 Hz, 1000 Hz and 10.000 Hz, the sum of these does NOT create anything beyond this range. The only issue is we need theoretically infinitely long signals, or their spectrums smear. I already told the smearing is insignificant and I am not going to argue about it. You are just a random guy online having some snake oil beliefs. Believe what you want. I am here for those who are willing to listen and learn.


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## audiokangaroo (Aug 14, 2021)

71 dB said:


> Combination of frequencies within a limited band is also band-limited! If I put together 100 Hz, 1000 Hz and 10.000 Hz, the sum of these DOES not create anything beyond this range. The only issue is we need theoretically infinitely long signals, or their spectrums smear. I already told the smearing is insignificant and I am not going to argue about it. You are just a random guy online having some snake oil beliefs. Believe what you want. I am here for those who are willing to listen and learn.


I don't know any combination process that doesn't produce any information. If some information is produced through the building of the waveform, this must be reflected into the frequency content. We hear the global waveform before it is analysed into frequencies by the inner ear and the reproduction of the actual waveform is important. Reproducing audible frequencies inside the waveform is probably not sufficient to achieve high fidelity. If 44.1 was sufficient, it wouldn't sound digital but it would sound analogue.
Yes, I'm a random guy online , just like you and anybody here.


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## old tech

audiokangaroo said:


> I don't know any combination process that doesn't produce any information. If some information is produced through the building of the waveform, this must be reflected into the frequency content. We hear the global waveform before it is analysed into frequencies by the inner ear and the reproduction of the actual waveform is important. Reproducing audible frequencies inside the waveform is probably not sufficient to achieve high fidelity. If 44.1 was sufficient, it wouldn't sound digital but it would sound analogue.
> Yes, I'm a random guy online , just like you and anybody here.


What do you mean by sounding digital? At 44.1 digital sounds like the original audio signal whereas analog is a degradation of the original signal. No need to guess which is higher fidelity and has made modern technology possible. You're coming across as one of those irrational digiphobes. As for your assertion that 192 sounds different than 44.1, yes i am sure you are special and an exception to technical impossibilites and all the credible controlled tests conducted over the past 20 odd years.


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## 71 dB (Aug 15, 2021)

audiokangaroo said:


> I don't know any combination process that doesn't produce any information. If some information is produced through the building of the waveform, this must be reflected into the frequency content. We hear the global waveform before it is analysed into frequencies by the inner ear and the reproduction of the actual waveform is important. Reproducing audible frequencies inside the waveform is probably not sufficient to achieve high fidelity. If 44.1 was sufficient, it wouldn't sound digital but it would sound analogue.
> Yes, I'm a random guy online , just like you and anybody here.


The frequency content is what the waveform is built of.
"Sufficient" audio format doesn't have a sound to it, it is neutral, "soundless". It doesn't do anything to the original source. digital is much closer to that than analog and this is the reason why some people like you like analog. You like the sound (distortions) it adds to the source material.
If 192 kHz and 44.1 kHz are different to you it is because of different masters or placebo effect or both. Your mind doesn't allow you to hear 44.1 kHz correctly, because you have false mental images of its insufficiency. I don't have high hopes for you, but if you someday understand that 44.1 kHz is sufficient, you start hearing the sound correctly.
Don't judge 44.1 kHz for some crappy overcompressed releases. That's not about 44.1 kHz. That's about overcompression. Listen to great 44.1 kHz stuff (classical music is often a safe pet) and you'll see that 44.1 kHz is sufficient and enough.


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## castleofargh

Can we all agree now that @audiokangaroo talks with too much confidence and does too much fishing, on subjects he obviously knows too little about?


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## 71 dB

castleofargh said:


> Can we all agree now that @audiokangaroo talks with too much confidence and does too much fishing, on subjects he obviously knows too little about?


Not knowing isn't the problem. Lack of willingness to learn is. Full understanding of digital audio takes a long time, at least it took for me. Heck, who knows if there's still some special nuggets of understanding about digital audio I don't have yet?

Images of staircase waveforms "teached" people the intuitive, but false idea that more resolution is always more fidelity. To understand digital audio one needs to unlearn that. Bad sounding CD releases "teached" people the false idea that 44.1 kHz/16 bit isn't sufficient, when the format simply faithfully reproduced the bad sound that was put on it. People have learned to like the distortions of analog sound and confuse this for better fidelity without realising you can have the exact same sound/distortions on digital format if you put them there. People are psychologically unable to admit they prefer _lesser_ fidelity (e.g. vinyl). Then we have hi-rez snake oil sellers with their convincing-looking pseudo-scientific claims and differing masters about the "superiority" of higher sample rates. That is a lot to de-program! 

Every forum has overconfident people who know less than they think. My patience struggles to communicate with such people and my MBTI personality type being INTJ (Introvert-iNtuitive-Thinking-Judging) doesn't help. Fortunately every now and then there are people who have come to this discussion board to _learn_. Those people put me on a good mood.


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## audiokangaroo

I don't think that I'm overconfident. I compared a lot of recordings corresponding to various PCM and DSD resolution and I try to explain the differences
in sound that I can hear between them.
I don't say that 44.1 KHz sounds bad. It sounds very clean and it can sound good, especially when it was produced from an analogue master. However, I can hear a lack of precision and realism with this format and I don't think that this difference can be totally explained through a possible  placebo effect.


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## 71 dB

audiokangaroo said:


> I don't think that I'm overconfident. I compared a lot of recordings corresponding to various PCM and DSD resolution and I try to explain the differences
> in sound that I can hear between them.
> I don't say that 44.1 KHz sounds bad. It sounds very clean and it can sound good, especially when it was produced from an analogue master. However, I can hear a lack of precision and realism with this format and I don't think that this difference can be totally explained through a possible  placebo effect.


Assume 90 % of the differences you hear in audio be due to placebo effect. Proper blind listening tests nullify the effects of placebo, and this is why in proper listening tests almost all differences typically disappear.


----------



## gregorio

PhonoPhi said:


> [1] You are always so categorical, it is amazing.
> [2] How then it is possible to claim that anything above 20 kHz does not matter for the audible range is not clear to me.



1. Thanks. When the scientific evidence is categorical and the only evidence against is audiophile marketing and "impressions" from those duped by it, then yes, it's entirely reasonable and rational to be "so categorical". In fact, not being "so categorical" would be unreasonable, irrational and/or ignorant!

2. Not only is it possible to claim that but it's THE ONLY reasonable and rational claim possible! The scientific/reliable evidence of what exists in music above 20kHz is categorical, the ability to hear that content is categorical and so is the lack of ability to detect if/when that content is removed.



audiokangaroo said:


> Capturing all audible frequencies is not sufficient if you want to achieve high fidelity.


True, unless of course you are talking about audible fidelity, in which case it's false.


audiokangaroo said:


> You have to capture also the information produced by the combination of these audible frequencies in the time domain, which is a mathematical issue.


True in a sense. Really it's an engineering issue, the conversion of the physical movement of mechanical sound waves into and from an electrical "analogue" signal. However, as these engineering issues are based on the mathematics of both physical motion and of electric signals, your statement is true in a sense.


audiokangaroo said:


> In fact, we have to capture the analogue waveform as accurately as possible. This problem is beyond the sampling theorem ...


Sorry, that makes absolutely no sense at all. How can that be "beyond" the sampling theorem when capturing an analogue waveform as accurately as possible is the entirety of what the Sampling Theorem was invented for and is? What "problem" do you think the Sampling Theorem was developed to addresses if not for capturing an analogue waveform as accurately as possible?


audiokangaroo said:


> We go from the time domain to the frequency domain with the Fourier transform. What we get with the Fourier transform are mathematical frequencies. They are a different thing than the audible physical frequencies that we can find in the time domain and that are band limited for audibility reasons.


What we get with a Fourier transform is a mathematical breakdown of all the frequencies that a physical waveform contains.


audiokangaroo said:


> Although the analogue waveform is made up with band limited frequencies, the product of the Fourier transform is not necessarily band limited. This can explain for instance why  192 KHz PCM can sound more accurate and natural than 44.1 kHz.


No! Again, a Fourier transform is a mathematical breakdown of all the frequencies in a signal. So if that signal is band limited and therefore contains no frequencies above a certain point then the Fourier transform will obviously reflect that fact. How does this explain why 192kHz can sound more accurate than 44.1kHz?


audiokangaroo said:


> Experience shows that only capturing the waveform with a very high level of accuracy can make digital sound like analogue.


Who's experience, the experience of those duped by marketing claims proven false? This is a rhetorical question by the way, because this is the Science Forum, not another audiophile impressions and experiences forum. In this forum, an "experience" (unsupported with reliable evidence) that contradicts the proven, demonstrated science is less than worthless!

Also, think about the logic of what you're claiming. The whole reason digital audio was invented in the first place was to avoid the inaccuracies and distortion inherent with transporting and storing analogue audio signals. Therefore, if digital audio were "like analogue" then it would have failed utterly, that's the very last thing digital audio wants to be "like"!

As far as digital audio is concerned, the opposite of your statement is therefore closer to the truth. A very LOW level of accuracy can make digital sound like analogue! Although this is only closer to the truth rather than the actual truth, because the distortions that would be introduced by such exceptionally poor digital accuracy are somewhat different from the distortions introduced by analogue.


audiokangaroo said:


> I don't know any combination process that doesn't produce any information. If some information is produced through the building of the waveform, this must be reflected into the frequency content.


True but then that will obviously be demonstrated in a Fourier transform and therefore be covered by the Sampling Theorem.


audiokangaroo said:


> We hear the global waveform before it is analysed into frequencies by the inner ear and the reproduction of the actual waveform is important.


Are you sure about that? In the signal chain inside your head, does your brain come before your inner ear or after?


audiokangaroo said:


> If 44.1 was sufficient, it wouldn't sound digital but it would sound analogue.


As per above, your assertion is effectively backwards. Who on earth would want a more accurate, superior format to sound like the less accurate, inferior format it was specifically invented to improve upon? What would be the point of that?


audiokangaroo said:


> Yes, I'm a random guy online , just like you and anybody here.


No, not just like castleofargh and anybody here! Many of us here know accept what Science has proven and demonstrated or at least are open to learning the scientific facts/evidence, instead of just blindly accepting and regurgitating the audiophile marketing nonsense already proven false.

G


----------



## gregorio

audiokangaroo said:


> I don't think that I'm overconfident. I compared a lot of recordings corresponding to various PCM and DSD resolution and I try to explain the differences
> in sound that I can hear between them.


Unfortunately, that's a contradiction. You "try to explain the differences" you hear but your explanation contradicts long proven and demonstrated scientific fact and therefore by definition, you are overconfident in your explanation.


audiokangaroo said:


> However, I can hear a lack of precision and realism with this format and I don't think that this difference can be totally explained through a possible placebo effect.


How do you know it's not placebo effect unless you actually eliminate that possibility? Even if you do eliminate that possibility how do you know that the lack of precision and realism you're perceiving isn't due to something other than the format, say a deliberately different mastering process for example? Unless you've eliminated the other possibilities then you're effectively just guessing.

Do you think it's wise to have any confidence at all in guesses that contradict the science or to express any confidence in them in a forum that's actually called the Science Forum?!!

G


----------



## audiokangaroo (Aug 15, 2021)

I think we have to solve the contradictions between established science and audition experiences.
I made a mistake when saying that high fidelity audio is beyond the sampling theorem. I should better say that it is beyond the usual interpretation of the waveform
and the way we traditionnaly apply the sampling theorem.
The fact that the waveform is made up from sinewaves that are band limited does not mean that this waveform is mathematically band limited. I think this is really the heart of the problem.
 This is a very abstract and difficult problem. The waveform is the product of a random combinatory process and this process does mathematically produce information. This information is audible and can be found in the frequency spectrum above 22 KHz. Not sampling this information will result in a distortion of the audible signals decoded by the inner ear. This is the reason why frequencies that we use to call ultrasonic are audible when they are part of the waveform, whereas they are not when they are taken separately.


----------



## gregorio

audiokangaroo said:


> I think we have to solve the contradictions between established science and audition experiences.


What contradictions between the established science and audition experiences? There aren't any!


audiokangaroo said:


> The fact that the waveform is made up from sinewaves that are band limited does not mean that this waveform is mathematically band limited.


Of course it does, in fact that's the very definition of a band limited waveform!


audiokangaroo said:


> I think this is really the heart of the problem.


Yep 


audiokangaroo said:


> This is a very abstract and difficult problem. The waveform is the product of a random combinatory process and this process does mathematically produce information.


No it's not! It is not abstract or random at all, it is entirely logical and ENTIRELY predictable, that is what Fourier discovered about 2 centuries ago. It's also what the Nyquist-Shannon Theorem proves and has been demonstrated in practice since the 1950's and not just in digital audio, analogue synthesis is entirely based on the predictable addition and subtraction of various frequencies to generate complex waveforms! If it were all abstract and random, synthesizers couldn't exist.


audiokangaroo said:


> This information is audible and can be found in the frequency spectrum above 22 KHz. Not sampling this information will result in a distortion of the audible signals decoded by the inner ear.


I'm sorry but you cannot just keep making up assertions, without ANY reliable supporting evidence, that contradicts decades or even centuries of proven science.


audiokangaroo said:


> This is the reason why frequencies that we use to call ultrasonic are audible when they are part of the waveform, whereas they are not when they are taken separately.


Again, not only wrong but actually exactly backwards. The lowest ultrasonic frequencies can be audible under certain circumstances, but ONLY if they are played SEPARATELY, at extremely high (almost damaging) levels and the listener is young with healthy ears. However, at safe listening levels and when part of a complex waveform containing higher level signals within the audible band (such as all music recordings) then they are NOT audible. Frequencies that we used to call ultrasonic we still call ultrasonic and we call them ultrasonic precisely because they are inaudible!

G


----------



## audiokangaroo (Aug 15, 2021)

The process is much more random than you think. The dispatching of the elementary sinewaves in the time domain in totally random. This is the reason why you can always have information between two samples, even though the elementary components are band limited.
What we sample is not sinewaves but the surface of the waveform. If you don't sample this surface with a very high precision, then the reproduction of the sinewaves that are inside is distorted.
We need to be aware that the AD converter doesnt have any access to the content of the waveform.
What is under the waveform is band limited, but the surface is not.


----------



## bfreedma

audiokangaroo said:


> The process is much more random than you think. The dispatching of the elementary sinewaves in the time domain in totally random. This is the reason why you can always have information between two samples, even though the elementary components are band limited.
> What we sample is not sinewaves but the surface of the waveform. If you don't sample this surface with a very high precision, then the reproduction of the sinewaves that are inside is distorted.
> We need to be aware that the AD converter doesnt have any access to the content of the waveform.
> What is under the waveform is band limited, but the surface is not.



I’m curious as to where this information is coming from.  My apologies, but reading through the thread, it feels to me like you’re constructing a model as you go in an effort to support your beliefs.  All the while, discounting and discrediting existing knowledge of the topic (most of which is peer reviewed and not that recent)

I’m happy to reconsider if you can post links to data or research that supports your vision.  Something peer reviewed would be best and, hopefully, not from a vendor.


----------



## 71 dB (Aug 15, 2021)

audiokangaroo said:


> The dispatching of the elementary sinewaves in the time domain in totally random. This is the reason why you can always have information between two samples, even though the elementary components are band limited.
> What we sample is not sinewaves but the surface of the waveform. If you don't sample this surface with a very high precision, then the reproduction of the sinewaves that are inside is distorted.
> We need to be aware that the AD converter doesnt have any access to the content of the waveform.
> What is under the waveform is band limited, but the surface is not.


Sure, there are "information" between samples, but as long as the signal is properly bandlimited so that no frequencies above the Nyquist frequency are present, it is all _redundant_ information. We already know _everything_ there is to know about the signal having the sample points. There is only one way the signal can go between sample points. Any other way would make the signal non-bandlimited. That is what the sampling theorem tells us. That is the whole foundation of digital audio.

If the original signal contains frequencies above the Nyquist frequency (22.05 kHz in this context) and we need to bandlimit it before sampling, the waveform will of course be different (to eyes/mind), but not for the ears. That's what ultrasonics not being audible means.

Bandlimiting the waveform removes all frequencies outside the band and makes the shape of the waveform such that sampling it captures all information. All the remaining sinewaves within the band will be sampled correctly. There is no distortion. How many times this has to be told to you?


----------



## audiokangaroo (Aug 15, 2021)

71 dB said:


> Sure, there are "information" between samples, but as long as the signal is properly bandlimited so that no frequencies above the Nyquist frequency are present, it is all _redundant_ information. We already know _everything_ there is to know about the signal having the sample points. There is only one way the signal can go between sample points. Any other way would make the signal non-bandlimited. That is what the sampling theorem tells us. That is the whole foundation of digital audio.
> 
> If the original signal contains frequencies above the Nyquist frequency (22.05 kHz in this context) and we need to bandlimit it before sampling, the waveform will of course be different (to eyes/mind), but not for the ears. That's what ultrasonics not being audible means.
> 
> Bandlimiting the waveform removes all frequencies outside the band and makes the shape of the waveform such that sampling it captures all information. All the remaining sinewaves within the band will be sampled correctly. There is no distortion. How many times this has to be told to you?


Could you explain why you think that the information between the samples is redundant ?
According to you, what do we need to sample ? The frequency content under the waveform or the frequency content of the surface ?
Do you think that the AD converter can access the frequency content under the waveform  ?


----------



## audiokangaroo

bfreedma said:


> I’m curious as to where this information is coming from.  My apologies, but reading through the thread, it feels to me like you’re constructing a model as you go in an effort to support your beliefs.  All the while, discounting and discrediting existing knowledge of the topic (most of which is peer reviewed and not that recent)
> 
> I’m happy to reconsider if you can post links to data or research that supports your vision.  Something peer reviewed would be best and, hopefully, not from a vendor.


What I said here is the product of my own reflexion and I cannot refer to any academic research to back it. However, I read an interview with a renowned mastering engineer in California who stated that high fidelity digital audio begins with 192 KHz PCM. This is not a scientific proof, of course, but the opinion of such golden ears professionals
 should not be ignored or neglected.
On the other side, another kind of experimentation is possible. We generally admit that people older than 30 cannot hear anyting above 16 KHz. Then we could sample a good analogue master on tape at a sample rate of 32 Khz and see if it sounds as good as 44.1 KHz. 
We can also try to interprete the sonic results achieved with DSD. This technique is very different from PCM and the sound depends a lot on the analogue filtering after the bitstream, but I think it can be compared to PCM with a sample rate above 500 KHz, depending on the corner and the slope of the filter.


----------



## Slaphead

Man this thread is rubbernecking paradise.


----------



## KeithPhantom

He is a troll, ignore him.


----------



## bfreedma

audiokangaroo said:


> What I said here is the product of my own reflexion and I cannot refer to any academic research to back it. However, I read an interview with a renowned mastering engineer in California who stated that high fidelity digital audio begins with 192 KHz PCM. This is not a scientific proof, of course, but the opinion of such golden ears professionals
> should not be ignored or neglected.
> On the other side, another kind of experimentation is possible. We generally admit that people older than 30 cannot hear anyting above 16 KHz. Then we could sample a good analogue master on tape at a sample rate of 32 Khz and see if it sounds as good as 44.1 KHz.
> We can also try to interprete the sonic results achieved with DSD. This technique is very different from PCM and the sound depends a lot on the analogue filtering after the bitstream, but I think it can be compared to PCM with a sample rate above 500 KHz, depending on the corner and the slope of the filter.



I really don’t think you’re going to change minds here regarding established audio science if all you have is an unattributed testimonial from a single anonymous person on the Internet (and your opinion).

Suggestion:  take time and conduct your experiments.  Once you have actual data, if you think it’s indicative of something, post it.  That would lead to a potentially constructive discussion.  I’d also read a book or two on psychoacoustics - Floyd Toole is an excellent starting place.


----------



## Slaphead

KeithPhantom said:


> He is a troll, ignore him.


Me, or him?

I'm on your side. 16/44 or 16/48 is far beyond that which is needed for end listener purposes.


----------



## KeithPhantom

Slaphead said:


> Me, or him?
> 
> I'm on your side. 16/44 or 16/48 is far beyond that which is needed for end listener purposes.


He obviously.


----------



## audiokangaroo

bfreedma said:


> I really don’t think you’re going to change minds here regarding established audio science if all you have is an unattributed testimonial from a single anonymous person on the Internet (and your opinion).
> 
> Suggestion:  take time and conduct your experiments.  Once you have actual data, if you think it’s indicative of something, post it.  That would lead to a potentially constructive discussion.  I’d also read a book or two on psychoacoustics - Floyd Toole is an excellent starting place.


Thank you for considering my explanations seriously. The main point a disagree with some people here is about the nature of the waveform. This is a mathematical problem and I would need the help of a skilled mathematician to elucidate it properly.
We need to understand that the waveform is the product of a mathematical function. The audible band is the input of the fonction and the actual waveform is the output.
I don't see any reason that the output should belong to the same frequency band as the input.
If we take the function called multiplication, for instance, with as input the integer numbers between 0 and 9, we can see that the output goes from 0 to 81. The oustput space is wider that the input space. This should be the same with waveforms and frequencies.


----------



## bfreedma

audiokangaroo said:


> Thank you for considering my explanations seriously. The main point a disagree with some people here is about the nature of the waveform. This is a mathematical problem and I would need the help of a skilled mathematician to elucidate it properly.
> We need to understand that the waveform is the product of a mathematical function. The audible band is the input of the fonction and the actual waveform is the output.
> I don't see any reason that the output should belong to the same frequency band as the input.
> If we take the function called multiplication, for instance, with as input the integer numbers between 0 and 9, we can see that the output goes from 0 to 81. The oustput space is wider that the input space. This should be the same with waveforms and frequencies.



Sorry to disappoint you, but you offered nothing that I could consider seriously.  I’m just trying to suggest a path that will provide a better background so that, hopefully, you gain some perspective and, again hopefully, realize that your theories are highly unlikely to be correct.


----------



## 71 dB (Aug 15, 2021)

audiokangaroo said:


> Could you explain why you think that the information between the samples is redundant ?
> According to you, what do we need to sample ? The frequency content under the waveform or the frequency content of the surface ?
> Do you think that the AD converter can access the frequency content under the waveform  ?


It is redundant, because the original signal can be completely reconstructed using the information in the sample points. If I tell you a line goes thru origo (0,0) and the point (2,4) you know everything about the line. You can easily calculate the line goes also thru the point (-9, -18) for example. Similarly the behavior of the signal in between the samples is known thanks to band-limiting. There is no additional data to be gain.

The waveform is the signal in time-space. The frequencies (spectrum) is the same signal in frequency space. They represent the same thing in different spaces. PCM samples the signal in time-space.

AD converters don't care about frequencies. They just take samples of the signals every 1/fs seconds, where fs is the sampling frequency. It happens in time-space. Of course when you calculate the Fourier Transformation to get the spectrum you'll see the frequencies.

You make this more complicated than it is.


----------



## audiokangaroo

bfreedma said:


> Sorry to disappoint you, but you offered nothing that I could consider seriously.  I’m just trying to suggest a path that will provide a better background so that, hopefully, you gain some perspective and, again hopefully, realize that your theories are highly unlikely to be correct.


Auditions do confirm my theory. People who have learnt the basics of traditional digital audio generally consider for certain that a sample rate above 48 KHz cannot benefit to sound reproduction and they are submited to a nocebo effect because hearing the difference would lead them to reconsider what they learnt and this often makes them uncomfortable.


----------



## 71 dB

audiokangaroo said:


> Thank you for considering my explanations seriously. The main point a disagree with some people here is about the nature of the waveform. This is a mathematical problem and I would need the help of a skilled mathematician to elucidate it properly.
> We need to understand that the waveform is the product of a mathematical function. The audible band is the input of the fonction and the actual waveform is the output.
> I don't see any reason that the output should belong to the same frequency band as the input.
> If we take the function called multiplication, for instance, with as input the integer numbers between 0 and 9, we can see that the output goes from 0 to 81. The oustput space is wider that the input space. This should be the same with waveforms and frequencies.


Frequencies don't multiply each other the way you think. If a signal consists of 100 Hz, 1000 Hz and 10.000 Hz, those sine waves are added together and no knew frequencies are created. It is what it is! You confuse maybe nonlinear distortion? That is another issue.


----------



## 71 dB

audiokangaroo said:


> However, I read an interview with a renowned mastering engineer in California who stated that high fidelity digital audio begins with 192 KHz PCM.


Being in California doesn't make anyone a God of sound. Those mastering engineer exists, but so do those who know better.


----------



## audiokangaroo

71 dB said:


> It is redundant, because the original signal can be completely reconstructed using the information in the sample points. If I tell you a line goes thru origo (0,0) and the point (2,4) you know everything about the line. You can easily calculate the line goes also thru the point (-9, -18) for example. Similarly the behavior of the signal in between the samples is known thanks to band-limiting. There is no additional data to be gain.
> 
> The waveform is the signal in time-space. The frequencies (spectrum) is the same signal in frequency space. They represent the same thing in different spaces. PCM samples the signal in time-space.
> 
> ...


You say that AD converters don't care about frequencies. This is absolutly right. They only care about the value of the signal at a given point in time. But then, how can we decide which sampling frequency is sufficient ? Can we demonstrate that the frequency content of the waveform is band limited ?
The waveform is not the same thing as the audio band. There is a mathematical function between them. Then we have to consider the input and the output of the function.


----------



## KeithPhantom

audiokangaroo said:


> Auditions do confirm my theory.


You hear better than an Audio Precision APx555. Sir, you may be awarded your prize...


----------



## bfreedma

audiokangaroo said:


> Auditions do confirm my theory. People who have learnt the basics of traditional digital audio generally consider for certain that a sample rate above 48 KHz cannot benefit to sound reproduction and they are submited to a nocebo effect because hearing the difference would lead them to reconsider what they learnt and this often makes them uncomfortable.



Well, I tried to be nice…

Bluntly, it’s abundantly clear that you don’t have a good grasp of either the topic at hand or the scientific method.  Sighted subjective listening sessions by one person (you) “confirms your theory”?  No, it most definitely does no such thing.  Go find somewhere else to troll.

Seriously - you just claimed that your singular experience disproves existing science and supersedes the accumulated knowledge of people who understand digital audio.  Awesome.  Thanks for perfectly capturing the year 2021, where actual experts are ignored and lack of knowledge is celebrated.


----------



## audiokangaroo

KeithPhantom said:


> You hear better than an Audio Precision APx555. Sir, you may be awarded your prize...


I don't know this device. What is It ?


----------



## audiokangaroo (Aug 15, 2021)

bfreedma said:


> Well, I tried to be nice…
> 
> Bluntly, it’s abundantly clear that you don’t have a good grasp of either the topic at hand or the scientific method.  Sighted subjective listening sessions by one person (you) “confirms your theory”?  No, it most definitely does no such thing.  Go find somewhere else to troll.
> 
> Seriously - you just claimed that your singular experience disproves existing science and supersedes the accumulated knowledge of people who understand digital audio.  Awesome.  Thanks for perfectly capturing the year 2021, where actual experts are ignored and lack of knowledge is celebrated.


I'm happy that you are trying to be nice. This is not only about my own experience, though I globally trust my ears. High resolutions audio formats are becoming more and more successful, with offerings from Apple, Qobuz, Amazon and Tidal. The people who are ready to pay a premium price for them are expected to hear the benefit they bring to the sound. Why would they pay more for they same sound quality ?


----------



## 71 dB

Here is a simple demonstration of sampling:





The waveform above is 5 kHz sinewave at 44.1 kHz. Since the frequency is pretty high, the sampling looks coarse, and the line segments in between sample points look ugly, but that is just a simple way to "connect the dots". It is not how it plays out.

The waveform below is the same 5 kHz sinewave originally at 44.1 kHz, but RESAMPLED to 384 kHz sample rate! Look how beautifully the resampling algorithm has been able to "invent" the new sample points. This works, because the 44.1 kHz version is band-limited. There is only one way all the the new sample points can be and this is the result. The same happens in DACs, but the output is analog signal. This is why 44.1 kHz is enough.


----------



## bfreedma

audiokangaroo said:


> High resolutions audio formats are becoming more and more successful, with offerings from Apple, Qobuz, Amazon and Tidal. The people who are ready to pay a premium price for them are expected to hear the benefit they bring to the sound.



Good luck finding a properly controlled test that demonstrates humans can tell the difference between the same master in 44.1 vs any of the high res formats.  Gregorio already explained this in detail, but you seem determined to ignore everyone and plow ahead.

Not that I understand how this red herring helps your argument in any way.  You’re kind of just thrashing now.  Time to give up?


----------



## audiokangaroo

71 dB said:


> Here is a simple demonstration of sampling:
> 
> 
> The waveform above is 5 kHz sinewave at 44.1 kHz. Since the frequency is pretty high, the sampling looks coarse, and the line segments in between sample points look ugly, but that is just a simple way to "connect the dots". It is not how it plays out.
> ...


What you are showing here works with a simple sinewave. However, it is not relevant because an actual musical waveform is much more complicated than a sinewave.
Fourier demonstrated that it can be represented with sinewaves, but this series of sinewaves must generally be infinite.


----------



## 71 dB (Aug 15, 2021)

audiokangaroo said:


> You say that AD converters don't care about frequencies. This is absolutly right. They only care about the value of the signal at a given point in time. But then, how can we decide which sampling frequency is sufficient ? Can we demonstrate that the frequency content of the waveform is band limited ?
> The waveform is not the same thing as the audio band. There is a mathematical function between them. Then we have to consider the input and the output of the function.


Sampling theorem tells us the sufficient sampling rate. The frequency content dictates what kind of shape the waveform can take. Band-limiting limits the allowed shapes so that sampling works. This really is the FOUNDATION of digital audio, why it works.

waveform is time-space, bands are frequency-space. Apples and oranges. You are a confused person!


----------



## 71 dB (Aug 15, 2021)

audiokangaroo said:


> What you are showing here works with a simple sinewave. However, it is not relevant because an actual musical waveform is much more complicated than a sinewave.
> Fourier demonstrated that it can be represented with sinewaves, but this series of sinewaves must generally be infinite.


Sinewaves are not magically simpler in this sense. Complex musical signals are just as combination of sinewaves so if something works for one sinewave, it works for a million sinewaves. To proof this I did the same for music, now in STEREO! Look and be amazed!

Band-limited signals have a finite/limited series of sinewaves.


----------



## gregorio

audiokangaroo said:


> The process is much more random than you think.


What I think is irrelevant, it's what the science proves/demonstrates that's important here and what it proves is that it is NOT random.


audiokangaroo said:


> The dispatching of the elementary sinewaves in the time domain in totally random. This is the reason why you can always have information between two samples, even though the elementary components are band limited.


I have no idea what you mean by the "dispatching of the  elementary sinewaves in the time-domain". There are no "elementary sinewaves in the time-domain" unless you're reproducing nothing but a single sine wave.


audiokangaroo said:


> What we sample is not sinewaves but the surface of the waveform.


No, what we sample is the amplitude of the waveform.


audiokangaroo said:


> If you don't sample this surface with a very high precision, then the reproduction of the sinewaves that are inside is distorted.


Now you are contradicting yourself again. In another thread you repeatedly state how good DSD is but DSD has just one bit resolution, the very lowest precision that's possible in digital audio! So which is it, do you need very high precision, in which case DSD must be absolutely terrible or is your statement above false?


audiokangaroo said:


> We need to be aware that the AD converter doesnt have any access to the content of the waveform.


Of course it has access to the content of the waveform. If it has no access to the content of the waveform then what do you think a converter is converting? Even by your own incorrect definition of "what we sample", isn't this "surface of the waveform" part of the content of the waveform?


audiokangaroo said:


> What is under the waveform is band limited, but the surface is not.


There is no under or surface, there's just the waveform and a waveform is either band limited or it's not, it cannot be both at the same time.


audiokangaroo said:


> We generally admit that people older than 30 cannot hear anyting above 16 KHz. Then we could sample a good analogue master on tape at a sample rate of 32 Khz and see if it sounds as good as 44.1 KHz.


Not sure what you mean by "we could" because science and engineers have already done that. In fact it was standard practice in some parts of the audio industry at one time, so it's probably been done millions of times! Although under controlled test conditions probably only a few thousand times or so.


audiokangaroo said:


> We can also try to interprete the sonic results achieved with DSD.


Why do we need to interpret the sonic results achieved with DSD when we can precisely measure the sonic results objectively.


audiokangaroo said:


> This technique is very different from PCM and the sound depends a lot on the analogue filtering after the bitstream


No, it's very similar to PCM and one of the main points of DSD was to reduce the quality requirements of the analogue filtering. So again, pretty much the exact opposite of what you're falsely stating!


audiokangaroo said:


> We need to understand that the waveform is the product of a mathematical function.


Why do we need to understand a statement that is false?  


audiokangaroo said:


> Auditions do confirm my theory.


No they don't. Controlled tests demonstrate no one can tell the difference, in some sighted tests the subjects can't tell the difference and in some they can but even these cases do NOT confirm your theory! What subsquent controlled testing confirms is the theory of placebo effect.


audiokangaroo said:


> People who have learnt the basics of traditional digital audio generally consider for certain that a sample rate above 48 KHz cannot benefit to sound reproduction and they are submited to a nocebo effect because hearing the difference would lead them to reconsider what they learnt and this often makes them uncomfortable.


Which is why we haven't only tested scientists and audio engineers but also the public, audiophiles and students who have no knowledge of the basics of digital audio. The results are the same!


audiokangaroo said:


> High resolutions audio formats are becoming more and more successful, with offerings from Apple, Qobuz, Amazon and Tidal. The people who are ready to pay a premium price for them are expected to hear the benefit they bring to the sound. Why would they pay more for they same sound quality ?


Isn't that obvious? If marketing can convince them that the sound quality is better (even though it isn't) people will pay more. A very large part of the audiophile world is entirely based on this fact! Did you not know that?

G


----------



## 71 dB (Aug 16, 2021)

gregorio said:


> Isn't that obvious? If marketing can convince them that the sound quality is better (even though it isn't) people will pay more. A very large part of the audiophile world is entirely based on this fact! Did you not know that?
> 
> G


Some people are predominantly _sensing_ -type meaning they absorb information "as it is" while some other people are _intuitive_ -type meaning they analyse information and seek for hidden meanings within it. Sensing- types are typically better at detail while Intuitive -types are often better at critical evaluation of the information. Sensing -types, especially those who are also _feeling_ -types rather than _thinking_ -types are easier to convince of unwarranted claims and are therefor potential clients for snake oil sellers.

People experience the World differently and process information differently. What is self-evident for some may not be self-evident for some others. Myself being an INTJ explains why my way of thinking about audio is "Is this good enough?", "Why pay more than X euros for headphones when X euros is about the price category giving the most bang for the buck?" etc. Given my personality type and my knowledge of digital audio, I am one of the last people on Earth to be convinced of the "benefits" of hi-rez audio. This gives me the insight that xSFx -personality types without knowledge of digital audio are probably massively more easily convinced of the "benefits" of hi-rez audio.

It would be interesting to see studies of the correlations between personality types and the way people do audiophile hobby. I did Googling, but didn't find anything good.


----------



## audiokangaroo (Aug 16, 2021)

Gregorio, I'm not sure that you have a very good understanding of Fourier analysis.
The signal is made of elementary sinewaves. The waveform is the surface of this phenomenon. It is the output of a mathematical fonction.
The input of the fonction consist in sinewaves from the audible band and the position of these elementary signals in the time domain.
If you don't agree, what is you explanation of the waveform ?


----------



## castleofargh

audiokangaroo said:


> Gregorio, I'm not sure that you have a very good understanding of Fourier analysis.
> The signal is made of elementary sinewaves. The waveform is the surface of this phenomenon. It is the output of a mathematical fonction.
> The input of the fonction consist in sinewaves from the audible band and the position of these elementary signals in the time domain.
> If you don't agree, what is you explanation of the waveform ?


I wish you could join us back into the real world at some point. 

You replied to my last comment that you didn't think you were overconfident. This might just be your biggest mistake. And I’m saying that right after quoting complete nonsense about the surface of an audio waveform. Alluding to what, audio fishes swimming below? 
Goggle how a microphone works before posting even more science fiction. You have mistaken almost everything that can be so far, and that while people were explaining it to you in details. Forget about Nyquist, Fourier, and sampling for now. start at the beginning. What's a mic? What can it do? What signal can it deliver? Spoiler: no fish.


----------



## Nickhasarrived

castleofargh said:


> Spoiler: no fish.


Darn I was hoping to combine two of my favorite hobbies, fishing and audio.


----------



## castleofargh

Nickhasarrived said:


> Darn I was hoping to combine two of my favorite hobbies, fishing and audio.


I was only talking about the electrical signal coming from a mic. But with a little Pono imagination, there is still hope.





Apparently, hornitophiles need 384kHz to grow wings and fly with the birdies.


----------



## audiokangaroo (Aug 16, 2021)

castleofargh said:


> I wish you could join us back into the real world at some point.
> 
> You replied to my last comment that you didn't think you were overconfident. This might just be your biggest mistake. And I’m saying that right after quoting complete nonsense about the surface of an audio waveform. Alluding to what, audio fishes swimming below?
> Goggle how a microphone works before posting even more science fiction. You have mistaken almost everything that can be so far, and that while people were explaining it to you in details. Forget about Nyquist, Fourier, and sampling for now. start at the beginning. What's a mic? What can it do? What signal can it deliver? Spoiler: no fish.


What do you think I didn't understand about the way a mic works ?
I'm open to learn new things if you have something to explain. However, the waveform problem is exactly the same at the voltage level and at the air pressure level.
The waveform is the result of a process. I don't want to be overconfident, but this process seems to have a combinatory dimension and a random dimension.
I don't think that explaining the production of the waveform with a mathematical function is a wrong idea.
You have the right to compare my theory with science fiction, but it would be more useful if you could explain precisely what is wrong among the things I tried to explain.
In my opinion, considering that the waveform has the same frequency content as the audio band is simplistic.
I can give you again the example of multiplication, which is a rather simple mathematical function. If you take as input the integer numbers from 0 to 9, you can see that the
output range goes from 0 to 81. The output range is wider than the input range. I dont see any reason it should be any different with the building of a waveform from elementary audio frequencies.
We must be aware that what makes audio frequences that are present in the waveform audible is the decoding work done by the inner ear. If the waveform is not reproduced with a very high level of accuracy, the decoding will be more difficult and as a result the audio message will sound less natural.


----------



## 71 dB

I'm too tired of decoding audiokangaroo's posts and trying to make heads or tails out of the bizarre claims. It is like a perfect example of backward thinking from the conclusion: Analog is best, then comes hi-rez digital audio and then comes CD audio. How can this belief be justified? Well invent some crazy theories about wavefrom surfaces and how band-limited signals aren't actually band-limited. Its so insane I feel bad for audiokangaroo...


----------



## Nickhasarrived

audiokangaroo said:


> What do you think I didn't understand about the way a mic works ?
> I'm open to learn new things if you have something to explain. However, the waveform problem is exactly the same at the voltage level and at the air pressure level.
> The waveform is the result of a process. I don't want to be overconfident, but this process seems to have a combinatory dimension and a random dimension.
> I don't think that explaining the production of the waveform with a mathematical function is a wrong idea.
> ...


I think what you are talking about matters more to the way a DAC decodes the information. The number of samples will matter less to the way a DAC actually works.


----------



## audiokangaroo (Aug 16, 2021)

Nickhasarrived said:


> I think what you are talking about matters more to the way a DAC decodes the information. The number of samples will matter less to the way a DAC actually works.


DAC technology is interesting, of course, but this is not what we are discussing here. The number of samples is directly related to the frequency band we want to capture from
the waveform.
Basically, we want to capture perfectly every audible frequency in order to be able to reproduce them. These frequencies are sinewaves and we have to make sure that they are reproduced as perfect sinewaves and not as distorted sinewaves, because the difference is audible. What is not easy to understand here is that the perfect reproduction of
 these sinewaves whose frequency is limited to 22 KHz, depends on the capture of other sinewaves whose frequency is much higher. These ultrasonic frequencies are inaudible as such, but they are for mathematical reasons, necessary to achieve a perfect capture of the waveform. As the perfect capture of the waveform is necessary to the perfect capture and reproduction of the audible sinewaves, these very high frequencies have to be captured in order to achieve high fidelity. If they are not present in the sampled content, frequencies under 22 KHz will be distorted and the output of the DAC will not sound natural.


----------



## 71 dB

audiokangaroo said:


> DAC technology is interesting, of course, but this is not what we are discussing here. The number of samples is directly related to the frequency band we want to capture from
> the waveform.
> Basically, we want to capture perfectly every audible frequency in order to be able to reproduce them. These frequencies are sinewaves and we have to make sure that they are reproduced as perfect sinewaves and not as distorted sinewaves, because the difference is audible. What is not easy to understand here is that the perfect reproduction of
> these sinewaves whose frequency is limited to 22 KHz, depends on the capture of other sinewaves whose frequency is much higher. These ultrasonic frequencies are inaudible as such, but they are for mathematical reasons, necessary to achieve a perfect capture of the waveform. As the perfect capture of the waveform is necessary to the perfect capture and reproduction of the audible sinewaves, these very high frequencies have to be captured in order to achieve high fidelity. If they are not present in the sampled content, frequencies under 22 KHz will be distorted and the output of the DAC will not sound natural.


Are you by any chance talking about _Gibbs phenomenon_?


----------



## audiokangaroo (Aug 16, 2021)

71 dB said:


> Are you by any chance talking about _Gibbs phenomenon_?


Gibbs phenomenon is a mathematical phenomenon. I guess it is due to the fact that a sinewave is a sinewave, which sets a limit to the achievement of a perfect square wave.
This is not what I was talking about here, however. My point about sampling, which has to be confirmed, of course, is that the reproduction that the form of the signals that we can hear depends on the perfect capture and reproduction of the global waveform (what I called the surface). I think that this waveform is made of audible and inaudible frequencies and that each of those signals has to be captured. If we don't capture them all, the reproduced waveform will be different from the original, and the sinewaves produced by the decoding job in the inner ear will not be perfect, which is potentially audible.
As you can see, I don't pretend that we can hear frequencies much above 20 KHz, like bats. All we have to do is to make sure that audibles sinewaves are perfectly reproduced and decoded.


----------



## 71 dB (Aug 16, 2021)

audiokangaroo said:


> Gibbs phenomenon is a mathematical phenomenon. I guess it is due to the fact that a sinewave is a sinewave, which sets a limit to the achievement of a perfect square wave.
> This is not what I was talking about here, however. My point about sampling, which has to be confirmed, of course, is that the reproduction that the form of the signals that we can hear depends on the perfect capture and reproduction of the global waveform (what I called the surface). I think that this waveform is made of audible and inaudible frequencies and that each of those signals has to be captured. If we don't capture them all, the reproduced waveform will be different from the original, and the sinewaves produced by the decoding job in the inner ear will not be perfect, which is potentially audible.
> As you can see, I don't pretend that we can hear frequencies much above 20 KHz, like bats. All we have to do is to make sure that audibles sinewaves are perfectly reproduced and decoded.


Well, if you don't believe 44.1 kHz can reproduced audible sinewaves perfectly that is your problem. Sampling theorem proves it. My resampling examples show strong evidence for it too not to mention all the listening tests. You keep ignoring the evidence that debunks your claims. That is why people get frustrated with you.


----------



## Dogmatrix

@audiokangaroo I see the point you are trying to make and I also think there may be something there 
Such exploration is true science whether it proves fruitless or genius is for the future but the endeavour is science
Unfortunately this thread in spite of its name has little to do with science it would be better named sound engineering
You will not find any help in your quest here , I see the personal attacks are already beginning
I wish you all the best in your search for the truth


----------



## audiokangaroo

Dogmatrix said:


> @audiokangaroo I see the point you are trying to make and I also think there may be something there
> Such exploration is true science whether it proves fruitless or genius is for the future but the endeavour is science
> Unfortunately this thread in spite of its name has little to do with science it would be better named sound engineering
> You will not find any help in your quest here , I see the personal attacks are already beginning
> I wish you all the best in your search for the truth


Thank you for your nice comment. 
I'm pleased to read something positive. However, in spite of all the negative reactions I've met here, this discussion helped me to shape my theory and I think I
have now a better understanding of the phenomenons I tried to explain.


----------



## KeithPhantom

audiokangaroo said:


> These ultrasonic frequencies are inaudible as such, but they are for mathematical reasons, necessary to achieve a perfect capture of the waveform.


I am getting tired of you. Please, if you think this is even right in the slightest, please go and show the whole world how the Nyquist-Shannon Sampling Theorem is incorrect. *Please, provide both mathematical and experimental evidence of your claims. *Please, enlighten us and show how this theorem is wrong and your ideas that go against everything digital audio is are the right path.


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## audiokangaroo (Aug 16, 2021)

KeithPhantom said:


> I am getting tired of you. Please, if you think this is even right in the slightest, please go and show the whole world how the Nyquist-Shannon Sampling Theorem is incorrect. *Please, provide both mathematical and experimental evidence of your claims. *Please, enlighten us and show how this theorem is wrong and your ideas that go against everything digital audio is are the right path.


You don't need to be rude. I'm free to explain my opinion, even though it doesn't match you dogma.
I never said that the Nyquist-Shannon theorem was wrong. However, it is applied in a context were people believe that the signal they want to sample is band limited. The theorem is mathematically correct, but its application is based on a wrong understanding of the concept of waveform. Please read what I wrote above where I try to explain why the waveform is not band limited, in spite of audible frequencies beeing band limited.
My point is that conflating the frequency input of the waveform  and the frequency content of the waveform structure in a mistake, because there is a mathematical function between them.


----------



## KeithPhantom

audiokangaroo said:


> they want to sample is band limited.


Anybody who knows how recording even works knows that you record analog signals, and these are not bandlimited since *they are analog and come straight out of the instruments.*


audiokangaroo said:


> why the waveform is not band limited, in spite of audible frequencies beeing band limited.


What do you mean by this? It just does not make any sense. It *does not* matter what is audible, we can *perfectly *(plus quantization noise that is at least -90 dBFS from the fundamental @ 16 bits) reconstruct *any *signal from DC to Fs/2. We do not need extra ultrasonics to reconstruct signals below and at 22.05 kHz, we just sample at double the highest frequency desired. It works like this in audio and in RF.


audiokangaroo said:


> My point is that conflating the frequency input of the waveform and the frequency content of the waveform structure in a mistake, because there is a mathematical function between them.


What 'function'? Samples will equal their analog counterparts since they are one and the *same* but in different domains. Maybe the 'mathematical function' that you are trying to describe is the Delta-Sigma modulation that allows encoding continuous analog signals into discrete digital samples.

Me highlighting does not mean I am rude at you, it is just a visual aid so you can see important points that you may be missing.


----------



## audiokangaroo

Vamp898 said:


> I just checked that by recording a bass and played in the 100-200 Hz Reagion.
> 
> I recorded with 8 KHz 16bit and i recorded with 44.1 KHz 16bit
> 
> ...


This is a very interesting and useful experiment. People should learnt from it.
If traditional digital audio theory was right, audible frequencies up to 4 KHz should have been reproduced properly.
But they were not.


----------



## KeithPhantom

audiokangaroo said:


> If traditional digital audio theory was right, audible frequencies up to 4 KHz should have been reproduced properly.


You are forgetting that even these instruments that produce these low frequencies have harmonics and overtones, and these go further than 4 kHz.


----------



## audiokangaroo

Keith Phantom, analogue  signals are not band limited, as you said. That's true, and then the signal received by the microphone and which is converted into a voltage isn't limited either ( inside the frequency response limits of the microphone, but we have to consider this response at a very low amplitude level, not only around -3dB, which is not relevant for our problem ).
Then the question is about if we need to sample all the frequencies that contribute to the waveform structure or only those corresponding to the audible band.
We are used to think that sampling the frequencies corresponding to the audible range is sufficient, as higher frequencies are inaudible.
However, I think that we are comparing here apples and oranges, because the frequencies produced by the instruments and the frequencies building the waveform at le microphone level have a different mathematical signification. There is a physical fusion between them, which can be described by a mathematical function with an input and an output. What the microphone can see is this output. 
The reverse function is operated in the inner ear where this output is decoded into sinewaves corresponding to the audible band.
The key idea here is that the audible frequency content and the frequency content builing the waveform are two completely different things, and looking for a correspondance beteween them doesn't really make sense.
The recording experiment made by Vamp898 (see above) should be considered as a proof for what I say.
I know that my explanations are far from being perfect, and as I have limited mathematical skills I will not be able to back this with formulas.


----------



## sonitus mirus

KeithPhantom said:


> You are forgetting that even these instruments that produce these low frequencies have harmonics and overtones, and these go further than 4 kHz.



Absolutely. You can clearly see by a single bass guitar string being played and captured on an inexpensive iPad app that a 4 kHz limit is going to be quite disruptive from an audible perspective.  

Watch from around 3:12 in this video.


----------



## audiokangaroo (Aug 16, 2021)

KeithPhantom said:


> You are forgetting that even these instruments that produce these low frequencies have harmonics and overtones, and these go further than 4 kHz.


Your argument makes sense, but I don't think that it is sufficient to explain the extremely poor sonic result achieved with 8 KHz sampling.
If you were right, the result would sound like deprived from clarity rather that totally distorted. We would need to experiment further, though, to make a better
interpretation.


----------



## sonitus mirus

Try 16 kHz and see how it sounds.  I think the harmonics and overtones would be too quiet to make an audible difference at that high of a frequency for a sub-100 Hz fundamental at rational volume levels.


----------



## audiokangaroo

sonitus mirus said:


> Absolutely. You can clearly see by a single bass guitar string being played and captured on an inexpensive iPad app that a 4 kHz limit is going to be quite disruptive from an audible perspective.
> 
> Watch from around 3:12 in this video.



Not sure I did really undertand. What does this guy try to demonstrate ?


----------



## audiokangaroo

sonitus mirus said:


> Try 16 kHz and see how it sounds.  I think the harmonics and overtones would be too quiet to make an audible difference at that high of a frequency for a sub-100 Hz fundamental at rational volume levels.


How should an acoustic instrument sampled at 16 KHz sound, according to you ? Lacking airiness, lacking clarity or very thick and distorted ?


----------



## sonitus mirus

audiokangaroo said:


> Not sure I did really undertand. What does this guy try to demonstrate ?


It was only to demonstrate frequencies over 4 kHz would be quite predominant despite playing a low bass note.  8 kHz sampling rate would easily miss audible frequencies and anyone would most likely notice a difference.


----------



## sonitus mirus

audiokangaroo said:


> How should an acoustic instrument sampled at 16 KHz sound, according to you ? Lacking airiness, lacking clarity or very thick and distorted ?


I’m referring to your original bass note claim. Your sampling rate was too low and was not capturing the necessary harmonics and overtones.


----------



## audiokangaroo

sonitus mirus said:


> I’m referring to your original bass note claim. Your sampling rate was too low and was not capturing the necessary harmonics and overtones.


My interpretation of low rate sampling is that it produces a global waveform distortion. As a consequence, it has an effect all along the audible band, and not only on the upper range that has been forgotten through the sampling process. The result should sound globally thick and distorted. The problem seems to go beyond the lack of overtones.


----------



## Davesrose

I think we shouldn't forget what Castle brought up: the actual signal coming from the mic.  It doesn't matter how perfectly modeled your reproduced sine wave is.  It starts with your input.  And your input is limited to SNR.  In most situations, it's the analog input of noise floor and maximum signal/saturation that's a dynamic range below the capabilities of current digital formats.  And when it comes to arguments of vinyl having frequency ranges beyond 20khz....you might be able to measure frequencies from a TT, but why assume it's from an original source and not random fluctuations?


----------



## sonitus mirus

Your interpretation seems a bit nonsensical to me, to put it bluntly.  If there is an audible difference, I’m quite certain it would be obvious and measurable if the examples could be provided for independent analysis


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## sander99 (Aug 17, 2021)

audiokangaroo said:


> The key idea here is that the audible frequency content and the frequency content builing the waveform are two completely different things, and looking for a correspondance beteween them doesn't really make sense.


It does make sense, they are exactly the same, except that the inaudible frequency content can be left out.
You do realise by the way that as part of the AD conversion the analog input signal is first low pass filtered to avoid aliasing?
What later is reconstructed is of course the waveform as it was after the low pass filtering. But the only difference between that and the original analog waveform is composed of inaudible frequencies.
(And using a shallow analog filter and oversampling and digital filtering on the AD side can mitigate problems caused by steep analog filtering before you start about that.)



audiokangaroo said:


> Your argument makes sense, but I don't think that it is sufficient to explain to extremely poor sonic result achieved with 8 KHz sampling.


Yes it does fully explain it. _[Edit: My apologies, mistakes in how the test was performed could have conributed to even further degradation of the result. But stil it is fully explainable by the established sound science.]_


audiokangaroo said:


> If you were right, the result would sound like deprived from clarity rather that totally distorted.


I know from personal experience that lack of overtones can give a strong subjective impression of distortion, probably because our brain senses something is wrong and interprets it that way. Besides, I don't recall @Vamp898 using the word distorted.


audiokangaroo said:


> My point is that conflating the frequency input of the waveform and the frequency content of the waveform structure in a mistake, because there is a mathematical function between them.


It is not a mistake. The total waveform is the summation of the sine waves. Only the audible sinewaves matter.


audiokangaroo said:


> My interpretation of low rate sampling is that it produces a global waveform distortion.


Of course the waveform without the inaudible frequency content looks different from the waveform with inaudible frequency content. But the difference is not audible. Because the difference is the inaudible frequency content.


audiokangaroo said:


> Basically, we want to capture perfectly every audible frequency in order to be able to reproduce them. These frequencies are sinewaves and we have to make sure that they are reproduced as perfect sinewaves and not as distorted sinewaves, because the difference is audible.


The bandlimited version of the analog input signal (after low pass filtering) can be perfectly reconstructed from the sampled points. That is what the proven sampling theorem states. We sure don't need frequencies above Fs/2 for that.


audiokangaroo said:


> the sinewaves produced by the decoding job in the inner ear will not be perfect


The inner ear doesn't produce sinewaves. The hairs in the inner ear just react to the frequencies they are tuned to. By the way: an "imperfect sinewave" is of course always a summation of multiple perfect sinewaves.


audiokangaroo said:


> The recording experiment made by Vamp898 (see above) should be considered as a proof for what I say.


No. It can be fully explained by the established science of audio. It does not prove that your wild, speculative, unsubstanciated, and totally illogical theories disprove the established science.


----------



## Vamp898

Hey guys, why argue? Measure

I used リライト from Asian Kung Fu Generation (2016 Version), 96 KHz 24 bit. I changed the bit rate to 44.1 KHz and changed it back to 96 KHz and let Audacity analyze the difference

Here is the result






The biggest peak/difference is right after 20KHz, but as we are all able to see, there is a difference in all other frequencies too.

There is a difference, audacity says so. It exists. There is nothing to argue if it exists or not, its there and its measurable.

You can argue if it does matter, but not if its there.


----------



## 71 dB

KeithPhantom said:


> You are forgetting that even these instruments that produce these low frequencies have harmonics and overtones, and these go further than 4 kHz.


Especially the initial attacks of notes may contain lots of high frequencies. These frequencies probably die out fast, but are important in the sonic character.

Also, If you play a 8 kHz sample rate file, it probably needs to be resampled in real time to a higher sample rate supported by the system you are using and this sample rate conversion can be rude and can sound ugly. Here is an example of how much sample rate conversion quality can matter: I generated 2.5 kHz sine at 8 kHz sample rate (top of the picture). Then I resampled it to 44.1 kHz using the lowest (LQ) and highest (HQ) quality options. The highest quality mode gives really good result, perfect sine while the lowest quality mode struggles a lot.


----------



## 71 dB

Vamp898 said:


> Hey guys, why argue? Measure
> 
> I used リライト from Asian Kung Fu Generation (2016 Version), 96 KHz 24 bit. I changed the bit rate to 44.1 KHz and changed it back to 96 KHz and let Audacity analyze the difference
> 
> ...


What quality settings did you use? I did similar test with white noise: I generated white noise at 96 kHz sample rate (24 bit). Then downsampled it to 44.1 kHz (24 bit) using the best quality setting, but without dither. Then I upsampled it back to 96 kHz. I got this for the difference signal spectrum:





I won't lose my sleep over that -140 dB difference below 15 kHz. It is most probably just quantization noise.


----------



## castleofargh

Dogmatrix said:


> @audiokangaroo I see the point you are trying to make and I also think there may be something there
> Such exploration is true science whether it proves fruitless or genius is for the future but the endeavour is science
> Unfortunately this thread in spite of its name has little to do with science it would be better named sound engineering
> You will not find any help in your quest here , I see the personal attacks are already beginning
> I wish you all the best in your search for the truth


Is it true science to question evidence of something very solid, well accepted, and oppose it with no actual evidence that it might be wrong? All because someone happened to fill big gaps in his knowledge with intuition and ego? I don’t think that is the role of science at all. We didn’t burn all the books and start all research from scratch after some fool decided that the planet was flat.

Of course it's nothing new, anybody trying to make sense of digital audio without learning a good deal about it, will end up with some varying amount of nonsense. And will have some typical intuition that it is super flawed in some way for some reason. But not every physical phenomenon works in an intuitive way! Why so many audiophiles think otherwise is beyond me.
 There are things we need to learn just so our brain can start thinking within that frame of reference, and that will allow to understand and learn some more. And at some point, maybe someone will get a brilliant idea about how reality works.
One cannot just make stuff up from nothing and magically grasp sampling theory, how waves behave, human hearing and psychoacoustic. Kanga badly jumped to conclusion about what was audible, then made up his own idea of why(kind of). He now relies on false axioms to spam more deducted falsehoods, and openly question well established, well demonstrated, well tested knowledge. All because he doesn’t know much of anything on the topic(he probably learned more on waves in the last days than in his entire life before that). And at no point in that already too long spam of blatant nonsense, has he given a hint of being able to admit when he's wrong.
 You seeing science within his posts is a little scary.

But you do have a point, this forum isn't a research lab, or even a place where scientists meet. There is no rule to stop someone from posting utter nonsense, clearly. So of course, a bunch of posts(no matter the supported position on a topic), will be gluten and fact free. The sheer number of claims posted in a page without supporting evidence, tells you that while science oriented, little of what's in it is science.
Doesn't mean that complete made up nonsense has the same value as fact based knowledge. Or that ignorant guesses are science.


----------



## Vamp898

71 dB said:


> What quality settings did you use? I did similar test with white noise: I generated white noise at 96 kHz sample rate (24 bit). Then downsampled it to 44.1 kHz (24 bit) using the best quality setting, but without dither. Then I upsampled it back to 96 kHz. I got this for the difference signal spectrum:
> 
> 
> 
> I won't lose my sleep over that -140 dB difference below 15 kHz. It is most probably just quantization noise.


Congratulations. With testing something different, you got different results.

Of course the sampling works much better with white noise or even a plain Sinus curve.

Thats why i used a complex song with erratic patterns. The more complex and erratic the signal is, the more is it prone to errors.

So yes, it works better with white noise. But i rarely listen to white noise, i listen to music.


----------



## 71 dB

Vamp898 said:


> Congratulations. With testing something different, you got different results.
> 
> Of course the sampling works much better with white noise or even a plain Sinus curve.
> 
> ...


Noise is maximally complex signal and using white noise the spectrum flat making it easy to see how the resampling works on different frequencies.
The result of your test is so bad it must be due to some lousy algorithm used in the resampling.


----------



## gregorio

audiokangaroo said:


> What do you think I didn't understand about the way a mic works ?


This: "_That's true, and then the signal received by the microphone and which is converted into a voltage isn't limited either ( inside the frequency response limits of the microphone, but we have to consider this response at a very low amplitude level, not only around -3dB, which is not relevant for our problem_ )."

First of all, the acoustic signal (sound pressure wave) is limited. It's limited by the laws of physics, the loss of amplitude over distance, even more so the higher the frequency, the noise floor of the environment (Brownian motion etc.) and that's before we even get to the microphone. A microphone capsule has mass and is therefore subject to the laws of motion of objects with mass; inertia, friction, etc. ALL microphones therefore MUST have a limit to the sound wave frequency to which they can respond. The response of many music recording microphones rolls off significantly above around 14kHz, most others at around 20kHz and only a handful of so go up to or beyond 40Khz. The analogue signal output by a mic is not band limited, HOWEVER, much beyond the mic's stated response, that frequency content is just noise/distortion generated by the mic's internal electronics and is unrelated to the acoustic signal hitting the mic's capsule.

So OF COURSE this is "relevant for our problem"! If there are no recordable frequencies above say 20kHz (besides unwanted noise/distortion), then the questions of recording, reproducing or being able to hear above 20kHz are all moot to start with.


audiokangaroo said:


> I'm open to learn new things if you have something to explain.


Clearly that's not true, you've had a number of things explained to but you have NOT learned from any of them and you just keep repeating the same self contradictory nonsense.


audiokangaroo said:


> The waveform is the result of a process. I don't want to be overconfident, but this process seems to have a combinatory dimension and a random dimension.


Again, you really haven't thought through your (false) explanation. If sine wave production were random as you've stated, then we wouldn't need musicians; you could put a violin on a stage and it would randomly start playing itself. Better still, the sine waves it was randomly producing would mean that instead of a violin, it could sound like a piano, the spice girls, a deer being hit by a truck or all three at the same time! How much confidence do you have in your explanation now? Wouldn't any amount of confidence be "overconfident"?


audiokangaroo said:


> I don't think that explaining the production of the waveform with a mathematical function is a wrong idea.


The production of all natural waveforms is obviously a mechanical process; the plucking or hitting of a string or some other material or the mechanical movement of air by the lungs through the voice box or a wind instrument. However, we can of course manufacture artificial waveforms, in which case it can be an electronic process, such as a signal generator or synthesizer and the only time it's a mathematical process is for certain artificial sounds generated purely in the digital domain. What maths is good for, is expressing the properties of sound waves and of course, sound waves and all the electronics used to record and reproduce them are governed by physical laws, which are again expressed with mathematics.


audiokangaroo said:


> You have the right to compare my theory with science fiction, but it would be more useful if you could explain precisely what is wrong among the things I tried to explain.


Again, how is it "more useful" if you just completely ignore it and continue on regardless?


audiokangaroo said:


> I can give you again the example of multiplication, which is a rather simple mathematical function. If you take as input the integer numbers from 0 to 9, you can see that the
> output range goes from 0 to 81. The output range is wider than the input range. I dont see any reason it should be any different with the building of a waveform from elementary audio frequencies.


Exactly, it's been explained to you but you have NOT "been open to learn new things from it" and you just repeat the same nonsense regardless! It has been explained to you that sound waves loose high/ultrasonic frequency over distance/time, high/ultrasonic frequencies are not added and they certainly are not multiplied. We also loose high/ultrasonic freqs with analogue signal recording/reproduction, except for the added unwanted electronic distortion and (Johnson) noise. Therefore, you could hardly have picked a worse analogy than simple multiplication, you'd have been far better using subtraction or division as an analogy but of course that would have contradicted your false explanations. However, I can't see why we need any sort of analogy, when we already have the exact, proven mathematical function/s!!


Dogmatrix said:


> @audiokangaroo I see the point you are trying to make and I also think there may be something there
> Such exploration is true science whether it proves fruitless or genius is for the future but the endeavour is science


What "exploration" or "endeavour"? Just making up nonsense explanations that contradicts actual science, without a shred of reliable supporting evidence, is not ANY sort of science, let alone "true science". In fact, it's pretty much the opposite of true science! 

In my day, they taught the basics of what science is and the scientific method to all children in middle school, when did they stop teaching it?

G


----------



## Vamp898

71 dB said:


> Noise is maximally complex signal and using white noise the spectrum flat making it easy to see how the resampling works on different frequencies.
> The result of your test is so bad it must be due to some lousy algorithm used in the resampling.


I clicked on "change sample rate" in audacity and changed it there + back.

You used the exact same method so we're using the same lousy algorithm which should have caused the exact same picture.

Obviously it doesn't. I checked my Audacity settings, it says "Best quality (slowest)"

I assume you used the same.

If noise would be maximum complexity, why is it so much easier to remove noise than it is to remove vocals or instruments?

Because noise is static and everything else is (more or less) erratic.

The more erratic it is, the more the algorithms struggle.


----------



## KeithPhantom

Vamp898 said:


> white noise


Of course, since white noise is more complex than music due to its equal spectral context from DC to Fs/2. Also, it has a mean of 0 and infinite variance. The PDF of a white noise function will give you equal probability with infinite dispersion (standard deviation). Music is usually correlated, and many times, there is collinearity between signals as well. White noise is not collinear. More complex and as hard to describe in terms of pure probability and a Fourier decomposition as white noise there is no signal.


----------



## 71 dB

This is what I get when I try the 96 kHz => 44.1 kHz => 96 kHz test with music (chipmunked from 44.1 kHz to 96 kHz):
Triangle dither used in resampling

.


----------



## 71 dB

Vamp898 said:


> I clicked on "change sample rate" in audacity and changed it there + back.
> 
> You used the exact same method so we're using the same lousy algorithm which should have caused the exact same picture.
> 
> ...


I use Tracks / Resample.  Clearly the sampling theorem works for me and has always worked as it should. For you it doesn't work. I don't know why that is. Something weird is happening in your system. Are you sure the quality setting is for "high quality conversion" and not for "real time conversion?" My Audacity has selection for both. Also I make sure the result of each conversion is 24 bit.

Vocal are much louder than noise and also I think noise removed sounds to human ears better. Technically noise is more complex, theoretically infinitely complex making it totally random.


----------



## KeithPhantom

Vamp898 said:


> If noise would be maximum complexity, why is it so much easier to remove noise than it is to remove vocals or instruments?
> 
> Because noise is static and everything else is (more or less) erratic.


This is something very simple to answer. The problem with removing things such as vocals or instruments is that their respective descriptive functions cannot be generated as fast as an already-predefined white noise function. This is due to latency. You will have to analyze every part of, for example, a voice track, to decompose each frequency in real-time. This can be done, but it requires removing a bunch of latency between components and having enough computing power to quickly calculate the functions that describe the waveform. White noise is already known in terms of the generation function, so this is not an issue. Also, there are known noise shaping methods for white noise, on top of the use of randomly constructive noise (dither) that we can use as well.


----------



## gregorio

Vamp898 said:


> Of course the sampling works much better with white noise or even a plain Sinus curve.
> 
> Thats why i used a complex song with erratic patterns. The more complex and erratic the signal is, the more is it prone to errors.
> 
> ...



Huh?

You don't seen to realise that white noise is random amounts of ALL frequencies, it is therefore more complex and "erratic" than ANY musical signal. According to you, the algorithms should therefore struggle far more with white noise than any music recording!

You also don't seem to realise you are effectively trying to claim that the Nyquist-Shannon Sampling Theorem is wrong. According to Sampling Theory, any and all signals below half the sampling frequency can be captured and reproduced. That's it, there is no more, it's covers ALL audio signals, there is no clause that states "unless it's a complex, erratic music recording"! 

According to sampling theory and countless practical experiments, what you should get from this test is dither noise somewhere near the least significant bit (-138dB) depending on what type of dither you use and of course a large signal above the anti-imaging filter point (around 20kHz). If you're getting a different null result then you're doing something wrong!

G


----------



## Vamp898

I checked all my settings and i found the issue.

When i imported the song, it got imported as 96 KHz and 32bit
When i doubled it, it still had 96 KHz but 16bit.

So the bit rate was changed and i didn't noticed.

So the spectrum analysis from my audacity screenshot contains the lacking dynamic below 20KHz and lacking dynamic + frequencies above 20KHz.

So i changed my audacity settings and checked again and now, the result is similar to what you see.

I cut everything above 20 KHz to better analyze whats there and i ended up with the loudest peak at -80db

If your normalize that, it sounds like using a broken cable that doesn't transfer music anymore but if you wiggle it, you get those distorted nosie peaks  you could get the rhythm of the song if you knew it and new the exact part, but yes, only with normalize. Without, there is just ""silence""

So we went from -65db to around -90db in this song but still, there are distorted peaks up to -80db

You can't take that from me!  i'll stay true to my original statement, if you downsample 96 KHz, you're loosing information. Not that you need those (never said that), but it does happen and you're loosing information below 20 KHz

When i did this a half year ago or so, i lowered the frequency of the ultrasonic sounds to check whats there and ended up with white noise (we assumed that those might have been caused by electric devices in the recording room)

So this is not a proof (and never was) that Hi-Res Audio is useful or makes sense.

My statement was, it is not worth the risk and i stay true do that. Maybe someone in the sound studio had a wrong flag in the wrong settings, maybe they used bad downsampling, maybe they have outdated software or whatever.

And this is not unreal, there are lots of sound studios (especially cheaper ones) where mistakes can happen. Maybe its a bug in the software.

I'm going for Hi-Res Audio because i want to be as close to the original recording as possible.


----------



## audiokangaroo (Aug 17, 2021)

Gregorio, you really didn't understand what I mean because you don't seem to consider what happens in the time domain.
To understand what a waveform is we not only need to consider which signals are audible, but we also have to look at their position in the time domain.
Frequencies are not random, but their position in the time domain is random. This is what makes the waveform more complex than what most people think.
Then, we have to consider that frequencies in the waveform are not really audible. They are not what we use to call sound but only information about sound.
This information has to be computed by the ear to produce actual sound.
We use to think that frequencies in the waveform under 20 KHz are audible and those above 20 KHz are not, but I think this is wrong because all those frequencies are only information about audible sound that is coded into frequency. After it is decoded by the ear everything is band limited again.


----------



## Vamp898 (Aug 17, 2021)

gregorio said:


> Huh?
> 
> You don't seen to realise that white noise is random amounts of ALL frequencies, it is therefore more complex and "erratic" than ANY musical signal. According to you, the algorithms should therefore struggle far more with white noise than any music recording!


I don't think the noise created by Audacity is random, that is why you can simply remove it with the noise filter by almost 100%. It follows a pattern.



> You also don't seem to realise you are effectively trying to claim that the Nyquist-Shannon Sampling Theorem is wrong. According to Sampling Theory, any and all signals below half the sampling frequency can be captured and reproduced. That's it, there is no more, it's covers ALL audio signals, there is no clause that states "unless it's a complex, erratic music recording"!
> 
> According to sampling theory and countless practical experiments, what you should get from this test is dither noise somewhere near the least significant bit (-138dB) depending on what type of dither you use and of course a large signal above the anti-imaging filter point (around 20kHz). If you're getting a different null result then you're doing something wrong!
> 
> G


We already had that. The Nyquist-Shannon Sampling Theorem is a Theorem because it needs a ideal sinc-filter which doesn't exist.

I am not claiming its wrong, i just say its not working (perfectly) in the real world and Wikipedia says the same.


----------



## bfreedma

audiokangaroo said:


> Gregorio, you really didn't understand what I mean because you don't seem to consider what happens in the time domain.
> To understand what a waveform is we not only need to consider which signals are audible, but we also have to look at their position in the time domain.
> Frequencies are not random, but their position in the domain is random. This is what makes the waveform more complex than what most people think.
> Then, we have to consider that frequencies in the waveform are not really audible. They are not what we use to call sound but only information about sound.
> ...




No matter how high you stack your bologna, you won't turn it into filet mignon.  

Please stop posting for a bit and use that time to read some entry level books on digital audio and acoustics.


----------



## audiokangaroo

bfreedma said:


> No matter how high you stack your bologna, you won't turn it into filet mignon.
> 
> Please stop posting for a bit and use that time to read some entry level books on digital audio and acoustics.


Could you suggest a book where it is explained how a waveform is actually produced and how it is decoded by the ear ?
I would be glad to learn about your own bologna.


----------



## bfreedma

audiokangaroo said:


> Could you suggest a book where it is explained how a waveform is actually produced and how it is decoded by the ear ?
> I would be glad to learn about your own bologna.



I did at the beginning of this conversation.  You ignored it.

Start here:  https://www.amazon.com/Sound-Reprod...eakers+and+Room&qid=1629208617&s=books&sr=1-2


----------



## KeithPhantom

Vamp898 said:


> I am not claiming its wrong, i just say its not working (perfectly) in the real world and Wikipedia says the same.


In order for it to work perfectly, we would need infinite sample levels this lowering quantization noise to -infinite. But with 24-bit words, this quantization noise is at the very least -144 dBFS from the fundamental and way way outside the threshold of *absolute* hearing. 16-bit yields you about 96 dB SNR without quantization randomization (dither) or noise shaping. We are not looking for academically perfect results, but practically perfect ones. 



Vamp898 said:


> Theorem because it needs a ideal sinc-filter which doesn't exist.


No, it is a theorem because it is mathematically proven. Not only that, practical filters reduce possible intermodulation of already-filtered frequencies that would be aliased to something like -160 dBFS. Try to hear that. We may not have an academically perfect reconstruction filter, but in practical terms that concern our own absolute limitations, they are perfect.


----------



## audiokangaroo

bfreedma said:


> I did at the beginning of this conversation.  You ignored it.
> 
> Start here:  https://www.amazon.com/Sound-Reproduction-Psychoacoustics-Loudspeakers-Engineering/dp/113892136X/ref=sr_1_2?dchild=1&keywords=Sound+Reproduction:+The+Acoustics+and+Psychoacoustics+of+Loudspeakers+and+Room&qid=1629208617&s=books&sr=1-2


Thank you. I had a look and I could see that there is a chapter called "phase and polarity. do wa hear waveforms ?".
This might be relevant for our discussion. What does the author say about this ?


----------



## bfreedma

audiokangaroo said:


> Thank you. I had a look and I could see that there is a chapter called "phase and polarity. do wa hear waveforms ?".
> This might be relevant for our discussion. What does the author say about this ?



You asked for a recommendation and now want me to recount for you what’s in the book?  Why don’t you read it instead.  You might actually learn something.

Hint - the book doesn’t support your views.


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## gregorio (Aug 17, 2021)

Vamp898 said:


> I cut everything above 20 KHz to better analyze whats there and i ended up with the loudest peak at -80db


That's another testing error! Now you have artefacts of whatever filter you used to cut everything above 20kHz!


Vamp898 said:


> If your normalize that, it sounds like using a broken cable that doesn't transfer music anymore but if you wiggle it, you get those distorted nosie peaks  you could get the rhythm of the song if you knew it and new the exact part, but yes, only with normalize. Without, there is just ""silence"". So we went from -65db to around -90db in this song but still, there are distorted peaks up to -80db


That is either due to the filter you should not have used or is truncation error, which you should have avoided with dither (or not truncated to start with)!


Vamp898 said:


> i'll stay true to my original statement, if you downsample 96 KHz, you're loosing information. Not that you need those (never said that), but it does happen and you're loosing information below 20 KHz


Yes, you are loosing information, you loose all the information above the anti-imaging filter point, as I already said. If you are loosing something else, there is a fault with your test!


Vamp898 said:


> The Nyquist-Shannon Sampling Theorem is a Theorem because it needs a ideal sinc-filter which doesn't exist.
> I am not claiming its wrong, i just say its not working (perfectly) in the real world and Wikipedia says the same.


That's not what wikipedia says! Wikipedia states: "_Instead, some type of approximation of the sinc functions, finite in length, is used. The imperfections attributable to the approximation are known as interpolation error._" - So how much interpolation error do we get? It's actually difficult to measure, you need specialized measuring equipment, it's about 100 times below the noise floor of high quality 24bit converters and about 1,000 times below the ability of your speakers or headphones to reproduce. Notice that on the graph posted by @71 dB there's no sign of this interpolation error even at -145dB. As @KeithPhantom states, for all practical purposes (and indeed significantly beyond practical purposes) we can consider it perfect.

G


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## 71 dB (Aug 17, 2021)

Vamp898 said:


> I checked all my settings and i found the issue.
> 
> When i imported the song, it got imported as 96 KHz and 32bit
> When i doubled it, it still had 96 KHz but 16bit.
> ...


Okay. That explains a lot. Don't be too harsh to yourself. We all do mistakes. I have done mistakes like this many times. We learn from these mistakes. When you double a track or mix track together always check the format is correct because you might have used other settings earlier for something else.



Vamp898 said:


> So the spectrum analysis from my audacity screenshot contains the lacking dynamic below 20KHz and lacking dynamic + frequencies above 20KHz.
> 
> So i changed my audacity settings and checked again and now, the result is similar to what you see.


Great! The Sampling Theorem is working for you after all!



Vamp898 said:


> I cut everything above 20 KHz to better analyze whats there and i ended up with the loudest peak at -80db
> 
> If your normalize that, it sounds like using a broken cable that doesn't transfer music anymore but if you wiggle it, you get those distorted nosie peaks  you could get the rhythm of the song if you knew it and new the exact part, but yes, only with normalize. Without, there is just ""silence""


Yes, makes sense.



Vamp898 said:


> So we went from -65db to around -90db in this song but still, there are distorted peaks up to -80db


All of that is in the end inaudible, because the music itself masks it.



Vamp898 said:


> You can't take that from me!  i'll stay true to my original statement, if you downsample 96 KHz, you're loosing information. Not that you need those (never said that), but it does happen and you're loosing information below 20 KHz.


According to The Sampling Theorem you don't lose information below Nyquist if you do the downsampling properly. Of course the resolution you use limits your accuracy, but doing this in 24 bit gives easily enough accuracy and even 16 bit processing should be fine.

If you don't use dither when resampling you generate distortion. If you use dither you avoid distortion, but you add noise. That is not "loosing" information, because it is related to the bit depth and prosessing used. In theory we can make the bit depth arbitrary large meaning arbitrary small error in processing.



Vamp898 said:


> When i did this a half year ago or so, i lowered the frequency of the ultrasonic sounds to check whats there and ended up with white noise (we assumed that those might have been caused by electric devices in the recording room)


Yes, most stuff above 20 kHz in recordings is noise unless it is a careful hi-rez recording of something that produces a lot of ultrasonics.



Vamp898 said:


> So this is not a proof (and never was) that Hi-Res Audio is useful or makes sense.


Science and careful listening tests suggest the benefits of hi-rez audio are questionable at best. The only reasonable argument for sample rates above 44.1 and 48 kHz is the avoidance of steep anti-alias filtering, but even this argument suggests there is no need for sample rates above about 60 kHz in audio.


----------



## audiokangaroo

71 dB, why do you suggest 60 KHz for signal filtering, and not 80 KHz ou 120 KHz, for instance ?
Is there a good reason for choosing this frequency ?


----------



## Vamp898

KeithPhantom said:


> In order for it to work perfectly, we would need infinite sample levels this lowering quantization noise to -infinite. But with 24-bit words, this quantization noise is at the very least -144 dBFS from the fundamental and way way outside the threshold of *absolute* hearing. 16-bit yields you about 96 dB SNR without quantization randomization (dither) or noise shaping. We are not looking for academically perfect results, but practically perfect ones.
> 
> 
> No, it is a theorem because it is mathematically proven. Not only that, practical filters reduce possible intermodulation of already-filtered frequencies that would be aliased to something like -160 dBFS. Try to hear that. We may not have an academically perfect reconstruction filter, but in practical terms that concern our own absolute limitations, they are perfect.


Nobody ever said its even close of being in the threshold of hearing.

In my very fist statement i already said its inaudible and does not have any practical relevance. Neither the effect on the frequencies below 20KHz, nor the ones above.

So i think we all agree that we're talking about measuring values, not audible sounds.

As said, rhythmic, distorted peaks at -80db is out of question of being relevant.


----------



## Vamp898

71 dB said:


> The only reasonable argument for sample rates above 44.1 and 48 kHz is the avoidance of steep anti-alias filtering


What about recording dolphins or bats?


----------



## gregorio

audiokangaroo said:


> Gregorio, you really didn't understand what I mean because you don't seem to consider what happens in the time domain.


In my last response to you I stated "_It has been explained to you that sound waves loose high/ultrasonic frequency over distance/*time *..._"and previously I stated that "what we sample is amplitude over *time*" so clearly I am considering the time domain and your statement is FALSE!!


audiokangaroo said:


> To understand what a waveform is we not only need to consider which signals are audible, but we also have to look at their position in the time domain.


Again, what we sample is amplitude over time, that's it, nothing else!!


audiokangaroo said:


> Frequencies are not random, but their position in the time domain is random.


So going back to my previous post, what you're effectively saying is that "yes a violin placed on a stage will start (randomly) playing itself but not random frequecies, it will always sound like a violin, not a piano or the spice girls. You still confident in that are you?


audiokangaroo said:


> This is what makes the waveform more complex than what most people think.


No it's not, most people do not think a violin could randomly start playing itself. I don't really know what most people think but I don't need to, the complexity of the waveform can easily be analysed. And don't forget, much of this isn't new science it's some of the oldest, most well established science we have. Pythagoras discovered the mathematical relationship of music notes/harmonics around 2,500 years ago!


audiokangaroo said:


> Then, we have to consider that frequencies in the waveform are not really audible.


Some are, some aren't. If none were audible music wouldn't exist, we wouldn't hear pitch or in fact any sound at all!


audiokangaroo said:


> They are not what we use to call sound but only information about sound.


That obviously cannot be correct because sound is defined by it's frequency content. If a sound has no frequency content then it isn't sound!


audiokangaroo said:


> This information has to be computed by the ear to produce actual sound.


Have you been reading too much metaphysics? Somehow I don't think so. According to you, sound is produced mathematically and randomly, now you're saying the ear produces the sound. So the human ear must be a random mathematical machine! Great, why stop at sound science when we can completely mangle the science of biology and anatomy as well!  And, how can recording work? If it's the ear producing the actual sound then we would have to put our microphones next to the ear rather than next to the musical instrument that's being played. If there's lots of people listening to a concert, who's ears do we record? How much confidence do you have in your assertion? 

G


----------



## 71 dB

audiokangaroo said:


> 71 dB, why do you suggest 60 KHz for signal filtering, and not 80 KHz ou 120 KHz, for instance ?
> Is there a good reason for choosing this frequency ?


About 60 kHz is the "optimal" sample rate for digital audio, because it allows "relaxed" anti-alias filtering, but isn't unnecessarily high. Higher sample rate than that doesn't give any kind of benefits. Using 44.1 kHz and 48 kHz sample rates forces the use of steep anti-alias filters, which isn't really a problem and also file sizes are smaller.

A digital consumer audio format of 54 kHz/13 bit would have almost the same bitrate as 44.1 kHz/16 bit. Anti-alias/reconstruction filtering would be "easier" and 13 bit dynamic range is enough especially if shaped dither is used and since 54 kHz sampling rate has bandwidth up to 27 kHz, most of the shaped dither noise energy could be pushed in the 20-27 kHz frequency band. However, we have the formats we have and fortunately they allow extremely good sound quality.


----------



## 71 dB

Vamp898 said:


> What about recording dolphins or bats?


That's different. That's scientific work. Also, if you want to use ultrasonic sounds in your music played back at lower speed to make them audible for humans high sample rates are beneficial, but _consumer_ audio is a different story.


----------



## audiokangaroo (Aug 17, 2021)

Vamp898 said:


> What about recording dolphins or bats?


Did you notice that the transfer of the audio content of real to real tape onto 44.1/16 results in a very significant loss of audio information  ?
If you want to make and opinion you can the visit the youtube channel called MarioPindaBR. This guy has remastered old tape recordings (some are from very well kown artists to DSD and 96 KHz PCM. Even through youtube transcoding, most DSD masters sound impressive and much better than 44.1/16 or even 96/24.


----------



## audiokangaroo (Aug 17, 2021)

71 dB said:


> About 60 kHz is the "optimal" sample rate for digital audio, because it allows "relaxed" anti-alias filtering, but isn't unnecessarily high. Higher sample rate than that doesn't give any kind of benefits. Using 44.1 kHz and 48 kHz sample rates forces the use of steep anti-alias filters, which isn't really a problem and also file sizes are smaller.
> 
> A digital consumer audio format of 54 kHz/13 bit would have almost the same bitrate as 44.1 kHz/16 bit. Anti-alias/reconstruction filtering would be "easier" and 13 bit dynamic range is enough especially if shaped dither is used and since 54 kHz sampling rate has bandwidth up to 27 kHz, most of the shaped dither noise energy could be pushed in the 20-27 kHz frequency band. However, we have the formats we have and fortunately they allow extremely good sound quality.


This sounds like old Dan Lavry's litterature.
 Do you have any scientific element to decide when an anti-alias filter becomes relaxed in terms of corner and slope  ?


----------



## KeithPhantom

audiokangaroo said:


> This sounds like old Dan Lavry's litterature.
> Do you have any scientific element to decide when an anti-alias filter becomes relaxed in terms of corner and slope  ?


Filter design is based on a tradeoff between steepness and frequency response. You can add ripple to the mix if you want. Assuming a stopband of 22.05 kHz, you can have perfect filtering in terms of steepness, but you have to reduce the upper limit of your bandwidth; that means that you will start filtering a bit before 20 kHz in order to be at least -100 dBFS at 22.05 kHz. This filter may be audible if you start too early to someone with good ears, so there is another approach. The other approach is to have a linear frequency response until the end of the passband, but relax the steepness of the filter by missing by a bit the 22.05 kHz target. Usually, filters that do this have their stopband at 24 kHz.

Even though that we have this filter design, we can also oversample before filtering, thus relaxing the requirements of the filter. The more filtering you need, the higher the order of the filter.


----------



## bfreedma

audiokangaroo said:


> Did you notice that the transfer of the audio content of real to real tape onto 44.1/16 results in a very significant loss of audio information  ?
> If you want to make and opinion you can the visit the youtube channel called MarioPindaBR. This guy has remastered old tape recordings (some are from very well kown artists to DSD and 96 KHz PCM. *Even through youtube transcoding, most DSD masters sound impressive and much better than 44.1/16 or even 96/24*.


At this point, I actually hope you are trolling.  Assessing DSD vs. 44.1 via YouTube streams?  We can’t possibly need to discuss why that isn’t viable, can we?


----------



## 71 dB

audiokangaroo said:


> This sounds like old Dan Lavry's litterature.


Yes it does. 



audiokangaroo said:


> Do you have any scientific element to decide when an anti-alias filter becomes relaxed in terms of corner and slope  ?


No. I don't have _expertise_ on filter design. This is all relative.


----------



## audiokangaroo (Aug 17, 2021)

KeithPhantom said:


> Filter design is based on a tradeoff between steepness and frequency response. You can add ripple to the mix if you want. Assuming a stopband of 22.05 kHz, you can have perfect filtering in terms of steepness, but you have to reduce the upper limit of your bandwidth; that means that you will start filtering a bit before 20 kHz in order to be at least -100 dBFS at 22.05 kHz. This filter may be audible if you start too early to someone with good ears, so there is another approach. The other approach is to have a linear frequency response until the end of the passband, but relax the steepness of the filter by missing by a bit the 22.05 kHz target. Usually, filters that do this have their stopband at 24 kHz.
> 
> Even though that we have this filter design, we can also oversample before filtering, thus relaxing the requirements of the filter. The more filtering you need, the higher the order of the filter.


I read that steep analogue filters could have an impact on phase inside the audible band. Do you think it's true ?


----------



## audiokangaroo (Aug 17, 2021)

bfreedma said:


> At this point, I actually hope you are trolling.  Assessing DSD vs. 44.1 via YouTube streams?  We can’t possibly need to discuss why that isn’t viable, can we?


Do the experience. It may look surprising, but in spite of the youtube transcoding and compression I can clearly hear the difference between DSD and 96/24, at least
when the original analogue master was not or only a little processed. Processing can make things at little blurred.
By the way, I believe that Youtube can go up to 192/24 and their compression process is very clean. 
You can also listen to samples from DSD on highdeftapetransfers.com.


----------



## 71 dB

bfreedma said:


> At this point, I actually hope you are trolling.  Assessing DSD vs. 44.1 via YouTube streams?  We can’t possibly need to discuss why that isn’t viable, can we?


Yeah, the insanity level is astonishing.


----------



## VNandor

Vamp898 said:


> Hey guys, why argue? Measure
> 
> I used リライト from Asian Kung Fu Generation (2016 Version), 96 KHz 24 bit. I changed the bit rate to 44.1 KHz and changed it back to 96 KHz and let Audacity analyze the difference
> 
> ...


Could you upload both the original and your downsampled version somewhere? A short clip like a minute or so is already enough if you don't want to upload the whole files. I could go into details of why the graph doesn't show what you think it shows (it's not precisely the spectrum of the difference) but I would like to try it for myself with your samples before.


----------



## 71 dB

audiokangaroo said:


> I read that steep analogue filters could have an impact on phase inside the audible band. Do you think it's true ?


Not only steep ones. All analog filters have an impact on phase because they are _minimum phase_ filters. In case of anti-alias filters the phase distortion within audible band is largest near the cut off frequency (20 kHz) and gets smaller toward lower frequencies. Phase distortion caused by minimum phase filters is "natural" and doesn't sound that bad so typically it is not very harmful.

The only way to use analog filters without phase distortion is running the signal thru the filter twice, the second time reversed! This cancels the phase distortion.

Digital filters can be designed to be_ linear phase_ meaning they don't produce phase distortion.


----------



## audiokangaroo

71 dB said:


> Yeah, the insanity level is astonishing.


Why do you think that Youtube sonic quality should be much inferior to Apple Music, for instance ?
They are both audio streaming other IP. They both can reach 192/24. Youtube uses a little compression, but this is not very audible. They have made a long way since
5 or 6 years ago, when they had a more or less trashy sound.


----------



## KeithPhantom

audiokangaroo said:


> I read that steep analogue filters could have an impact on phase inside the audible band. Do you think it's true ?


They affect phase due to they being minimum-phase, but there are other bigger issues that affect phase. Enclosures such as rooms and headphones affect phase response due to cancellations and resonances, and these are orders of magnitude greater than anything an analog filter can do. Not only that, we are pretty insensitive to absolute phase shifts, so the audibility factor still needs to completely be assessed.


----------



## 71 dB

audiokangaroo said:


> Why do you think that Youtube sonic quality should be much inferior to Apple Music, for instance ?
> They are both audio streaming other IP. They both can reach 192/24. Youtube uses a little compression, but this is not very audible. They have made a long way since
> 5 or 6 years ago, when they had a more or less trashy sound.


I don't know what the quality is, but I doubt it is CD quality. In 2013 or so it was something like 192 kbps AAC, am I wrong?


----------



## audiokangaroo

71 dB said:


> I don't know what the quality is, but I doubt it is CD quality. In 2013 or so it was something like 192 kbps AAC, am I wrong?


Yes, in 2013, but 8 years are a long time for the Internet. Apple Music is now offering ALAC streaming up to 192/24, which is lossless compression. 8 years ago, Youtube sound was trashy but they have improved quite a lot. If you can listen to a good master on Youtube the sound quality is pretty good.


----------



## vergesslich2 (Aug 17, 2021)

audiokangaroo said:


> [...] Youtube sound was trashy but they have improved quite a lot. If you can listen to a good master on Youtube the sound quality is pretty good.


Get youtube-dl and do "youtube-dl -F URL", without quotes, and you'll see. When you want to see which streams are playing in the browser, you can right click and click stats for nerds. It shows the stream numbers like (251). These numbers are also in the list (at the very left) that youtube-dl outputs, together with samplerates.


----------



## sonitus mirus

The biggest concern with Youtube is there is no provenance for the source.  I've seen a few videos lately using Opus at about 160 kbps for playback. (251)


----------



## Dogmatrix

@castleofargh , @gregorio 
I was becoming concerned the poster had inadvertently waded into deep water so I threw a line out
Sometimes it is safer to disengage than risk pulling someone deeper


----------



## KeithPhantom

audiokangaroo said:


> Even through youtube transcoding, most DSD masters sound impressive and much better than 44.1/16 or even 96/24.


Assessing mono-level modulation by using multi-level modulation and using ears that are better than a spectrum analyzer or an APx555. If you also knew how the broadband spectrum looks for 1-bit noise-shaped modulation you will be scared to death of the amount of quantization noise there is.


----------



## audiokangaroo

KeithPhantom said:


> Assessing mono-level modulation by using multi-level modulation and using ears that are better than a spectrum analyzer or an APx555. If you also knew how the broadband spectrum looks for 1-bit noise-shaped modulation you will be scared to death of the amount of quantization noise there is.


I globally prefer PCM to DSD, but the latter can sound really could under some conditions.
I read  a few articles about sigma delta modulation and noise shaping. Are you interested in talking about DSD ?
Did you study this subject ?


----------



## KeithPhantom

audiokangaroo said:


> Are you interested in talking about DSD ?


Not the thread for this. If you are interested, you can always post a new thread.


----------



## audiokangaroo

KeithPhantom said:


> Not the thread for this. If you are interested, you can always post a new thread.


Why do you think it's not the thread for this ? DSD is related to sampling frequency, isn't it ?


----------



## KeithPhantom

audiokangaroo said:


> Why do you think it's not the thread for this ? DSD is related to sampling frequency, isn't it ?


This thread is about 16 vs 24-bit PCM, not DSD. For a DSD discussion, you need another thread.


----------



## audiokangaroo (Aug 17, 2021)

KeithPhantom said:


> This thread is about 16 vs 24-bit PCM, not DSD. For a DSD discussion, you need another thread.


I guess that you have understood that we have been talking about sampling rate rather than bit depth for a few pages.
As for 16 bit vs 24 bit, I think that 16 bit is enough for playback. Maybe 18 bit could be a little better, but 24 bit certainly don't make sense for audio playback.
In my opinion, sampling rate in the real issue as for achieving high fidelity with digital.
Discussing bit depth is not really worth., because going beyond 16 bit will not improve sound in a significant manner.


----------



## 71 dB

audiokangaroo said:


> I guess that you have understood that we have been talking about sampling rate rather than bit depth for a few pages.
> As for 16 bit vs 24 bit, I think that 16 bit is enough for playback. Maybe 18 bit could be a little better, but 24 bit certainly don't make sense for audio playback.
> In my opinion, sampling rate in the real issue as for achieving high fidelity with digital.
> Discussing bit depth is not really worth., because going beyond 16 bit will not improve sound in a significant manner.


Well nice to see that you at least understand bit depth and its effect on sound quality, allthou I disagree with the "maybe 18 bit could be a little better" part. 16 bit already can provide more dynamic range than needed in consumer audio. I think 13 bits is enough in consumer audio. That is almost 80 dB of dynamic range (~20 dB more than vinyl at best)  even without dither and with shaped dither _perceptual_ dynamic range can be 90-100 dB. So, 16 bit digital audio is already overkill by 3 bits.



KeithPhantom said:


> This thread is about 16 vs 24-bit PCM.


Ironically we haven't been discussing much about bit depth here lately. Bit depth clearly isn't as controversial topic as 44.1/48 kHz vs. 96/192 kHz.


----------



## gregorio

audiokangaroo said:


> Do the experience. It may look surprising, but in spite of the youtube transcoding and compression I can clearly hear the difference between DSD and 96/24, at least
> when the original analogue master was not or only a little processed. Processing can make things at little blurred.
> By the way, I believe that Youtube can go up to 192/24 and their compression process is very clean.


No it can't. If you are a Premium Member, the very highest quality that can be selected is AAC 256 kps. For anything uploaded by someone who isn't a premium member playback is restricted to either AAC 128 or Opus or Vorbis no higher than about 160 kbps, sample rate is typically 44.1kHz and never higher than 48kHz. YouTube doesn't support lossless playback, not even of 44/16.

With regard to the sample rate when uploading music videos, YouTube states: "_44.1kHz recommended. Higher sample rates are accepted but not required (for example, 48kHz or 96kHz)_". - This implies the playback sample rate of music videos will be 44.1kHz, although Youtube doesn't specifically say that. In any event, even with Premium Membership, the playback sample rate cannot be not be higher than 96kHz because that's the highest sample rate supported by the AAC format.

So, what is your experience, that you can clearly hear the difference between a 44.1kHz Opus 160 vs a 44.1kHz Opus 160? Even with the highest quality possible as a Premium Member and even if YouTube supports playback at the highest possible sample rate, the DSD master would have exactly the same sample rate as the 96/24 master!

My experience, from what you've just posted, is that if something says "DSD" and something else says 96/24, then some people will "clearly hear the difference" in these hi-res formats when in fact they are both exactly the same (low-res, lossy) format!!



audiokangaroo said:


> Youtube uses a little compression, but this is not very audible.


So, you can "clearly hear" a sample rate difference when the sample rates are in fact exactly the same but the relatively huge difference from quite heavy lossy compression is "not very audible"?

G


----------



## 71 dB

gregorio said:


> My experience, from what you've just posted, is that if something says "DSD" and something else says 96/24, then some people will "clearly hear the difference" in these hi-res formats when in fact they are both exactly the same (low-res, lossy) format!!
> 
> G


Placebo effect in full action!


----------



## audiokangaroo

71 dB said:


> Well nice to see that you at least understand bit depth and its effect on sound quality, allthou I disagree with the "maybe 18 bit could be a little better" part. 16 bit already can provide more dynamic range than needed in consumer audio. I think 13 bits is enough in consumer audio. That is almost 80 dB of dynamic range (~20 dB more than vinyl at best)  even without dither and with shaped dither _perceptual_ dynamic range can be 90-100 dB. So, 16 bit digital audio is already overkill by 3 bits.
> 
> 
> Ironically we haven't been discussing much about bit depth here lately. Bit depth clearly isn't as controversial topic as 44.1/48 kHz vs. 96/192 kHz.


I read that Philips wanted tu use 14 bit for the CD format and that Sony advocated for 16 bit. 
It was not only for dynamic but also for linearity reasons. 16 bit files are enhanced with a dithering process to improve linearity.
Did you actually experiment with 13 bit to hear how it sounds ?


----------



## audiokangaroo (Aug 19, 2021)

gregorio said:


> No it can't. If you are a Premium Member, the very highest quality that can be selected is AAC 256 kps. For anything uploaded by someone who isn't a premium member playback is restricted to either AAC 128 or Opus or Vorbis no higher than about 160 kbps, sample rate is typically 44.1kHz and never higher than 48kHz. YouTube doesn't support lossless playback, not even of 44/16.
> 
> With regard to the sample rate when uploading music videos, YouTube states: "_44.1kHz recommended. Higher sample rates are accepted but not required (for example, 48kHz or 96kHz)_". - This implies the playback sample rate of music videos will be 44.1kHz, although Youtube doesn't specifically say that. In any event, even with Premium Membership, the playback sample rate cannot be not be higher than 96kHz because that's the highest sample rate supported by the AAC format.
> 
> ...


A 44.1 KHz audio file from a tape or downsampled from a 192 KHz recording will sound clearly better than a native 44.1 KHz recording. This is something that we have
to take into account here. This is the reason why even if Youtube is limited to 96 KHz we can hear adifference throu gh it.
I'm not sure about Youtube beeing limited to 96 KHz or not, as they don't communicate transparently on this matter. What is sure is that they have been focused on
greatly improving audio quality for a few years.
However, if you don't trust Youtube, you can experiment with nativedsd.com. They offer free high quality samples to listen to.


----------



## bfreedma

audiokangaroo said:


> A 44.1 KHz audio file from a tape or downsampled from a 192 KHz recording will sound clearly better than a native 44.1 KHz recording. This is somrthing that we have
> to take into account here. This is the reason why even if Youtube is limited to 96 KHz we can hear adifference throu gh it.
> I'm not sure about Youtube beeing limited to 96 KHz or not, as they don't communicate transparently on this matter. What is sure is that they have been focused on
> greatly improving audio quality for a few years.
> However, if you don't trust Youtube, you can experiment with nativedsd.com. They offer free high quality samples to listen to.



You really struggle to identify the difference between your individual subjective opinions and facts.

Have you ordered the book I recommended yet?  It’s obvious that you desperately need to be better educated on the topics you are attempting to discuss.


----------



## sander99

audiokangaroo said:


> A 44.1 KHz audio file from a tape ... will sound clearly better than a native 44.1 KHz recording.


Do you mean recorded from an analog tape?
Some people may like it, but that would be because they like the distortions, colorations, and speed variations of the analog tape.
The 44.1 KHz 16 bit digitized version of the tape would sound exactly the same.


audiokangaroo said:


> A 44.1 KHz audio file ... downsampled from a 192 KHz recording will sound clearly better than a native 44.1 KHz recording.


In the studio the higher formats may have merits (to keep accumulated degradation by many many processing steps below audibility tresholds). But the end result can be delivered to the consumer in 44.1, 16.
Also recording in 192 KHz (with a higher corner frequency analog filter in the ADC) and downconverting to 44.1 would effectively be comparable to what an oversampling AD converter is actually doing in real time.

As you are kind of saying yourself: the 44.1 kHz 16 bit format retains all these (objective or subjectively preferred) qualities.


----------



## Nickhasarrived

The only reason I listen to 24/192 over 16/44.1 in most cases is because the song is often mastered differently, and you can tell when the artist has some fun in making objects move or adding reflections/reverberations in the 24/192 that the 16/44.1 doesn't have. Does that make one better than the other just from that? No each is a different listening experience.
In terms of actual quality of a song that is mastered identically in 24/192 vs 16/44.1, I thought I heard a difference, but I am nearly 100% sure I couldn't tell in blind testing, and it was a phycological thing. (As long as the dac was bit/sample matched, which is much more important than the sample rate imo)


----------



## castleofargh

audiokangaroo said:


> A 44.1 KHz audio file from a tape or downsampled from a 192 KHz recording will sound clearly better than a native 44.1 KHz recording. This is somrthing that we have
> to take into account here. This is the reason why even if Youtube is limited to 96 KHz we can hear adifference throu gh it.
> I'm not sure about Youtube beeing limited to 96 KHz or not, as they don't communicate transparently on this matter. What is sure is that they have been focused on
> greatly improving audio quality for a few years.
> However, if you don't trust Youtube, you can experiment with nativedsd.com. They offer free high quality samples to listen to.


Yet another example of you making up the reality you like as you go.

If we start with some marketing, or your subjective impression under sighted conditions, the very first action would be to list the possible causes and try to eliminate the non audible ones. Is it marketing bs? How can we verify? Am I feeling that the sound is better with dsd because I know it is dsd that's playing? Is it simply a different mastering of the same song? Am I using a playback system that introduces extra and audible changes to one of the formats? How can I verify all that? And until I have, what the F am I doing posting conclusions of distinct audible improvements, or claiming to know the cause on a forum?????

Same thing for vinyl sounding closer to the original instrument, same thing for the latest tape vs cd statement, and the many other fantasy baits you’ve thrown at us.

Proper diagnotic, testing, and some reliable source of information, would be the logical approach of someone interested in facts. Instead, you keep making statements about sound and the cause of sound differences that stand on thin air and are usually false.
The one piece of supporting evidence for all the crap you have posted, the one and only attempt on your part to demonstrate something, has been that farce about listening to the sound improvements of dsd on youtube.


----------



## 71 dB

audiokangaroo said:


> I read that Philips wanted tu use 14 bit for the CD format and that Sony advocated for 16 bit.
> It was not only for dynamic but also for linearity reasons. 16 bit files are enhanced with a dithering process to improve linearity.
> Did you actually experiment with 13 bit to hear how it sounds ?


Yes, there was discussion about CD format being 14 bit or 16 bit between Philips and Sony. If I have understood it correctly this was due to the difficulty at the time - four decades ago - to make 16 bit analog to digital and digital to analog converters, a problem that has since disappeared as digital audio technology has matured. Philips even used 14 bit DACs in their first CD players despite of the format having 16 bits. The two least significant bits were simply dropped. 

The linearity of ADCs and DACs is one issue affecting the quality of digital audio. Dithering on the other hand is about making quantization process linear by randomizing the quantization error so that it become "transferred" to quantization noise leaving the quantized signal 100 % accurate and linear. In theory properly dithered quantized signals have zero error, zero distortion and zero non-linearity. They are just masked by the quantization (dither) noise. Without dithering quantization error correlates with the signal which means the signal itself is distorted and the result (granulation) sounds much worse than non-correlating dither noise. Dither can be used even to make ADCs and DACs suffering from linearity problems more linear! High level dither can be added before conversion and subtracted from the signal after conversion. However, nowadays converters are so good and linear that there is little need for this anymore. Fun fact: Recording a noisy microphone amp means quantization of "self-dithered" signal resulting in noisy, but potentially more linear result.

Yes, I have tested 13 bit audio, althou the amount of bits comes from calculation how much dynamic range is needed in consumer audio (about 80 dB). The only way to tell 13 bit audio from 16 bit audio is to listen to an extremely quiet part of a song and raising the volume up so that the 18 dB louder dither of 13 bit version becomes audible. Elsewhere it is impossible to tell them apart by listening. I'm not saying world should go to 13 bit. 16 bit works great for consumer audio and is enough with clear (safety) margin. 13 bit is just my view of the limit of what is enough in consumer audio when used optimally (the music uses all 13 bits of the dynamic range and preferably shaped dither is used).


----------



## 71 dB

audiokangaroo said:


> A 44.1 KHz audio file from a tape or downsampled from a 192 KHz recording will sound clearly better than a native 44.1 KHz recording.


Why? If you like the distortions introduced by analog tape that would explain why you think 44.1 kHz digital audio from analog tape would sound "better", but that doesn't explain why 44.1 kHz digital audio from 192 kHz recordings would sound better. If anything, 192 kHz recordings can be considered totally transparent meaning it doesn't matter if you do the 44.1 kHz recording natively or if it is downsampled from a 192 kHz recording.



audiokangaroo said:


> This is something that we have to take into account here.


No, we don't because it doesn't make sense (see above).


----------



## audiokangaroo

71 dB said:


> Why? If you like the distortions introduced by analog tape that would explain why you think 44.1 kHz digital audio from analog tape would sound "better", but that doesn't explain why 44.1 kHz digital audio from 192 kHz recordings would sound better. If anything, 192 kHz recordings can be considered totally transparent meaning it doesn't matter if you do the 44.1 kHz recording natively or if it is downsampled from a 192 kHz recording.
> 
> 
> No, we don't because it doesn't make sense (see above).


I think that tape and 192 KHz are a bit similar because they have both a vey wide frequency response.
We use to think that tape has a frequency response that is not as extended as what CD can provide. This is true if we consider frequency response at -3dB, but measuring this way does not make sense, because the useful amplitude ofvery high frequencies is generally under -40dB. If we measure the frequency response of a tape recoder around -50dB, we will have to consider that tape is indeed a high resolution format with a very extended frequency response.
As of tape recording producing harmonic distortion, It may be considered problematic, but I don't think that it can explain the fact tape sounds more natural that basic
digital audio.


----------



## 71 dB

audiokangaroo said:


> I think that tape and 192 KHz are a bit similar because they have both a very wide frequency response.


All that "very wide frequency response" gets band-limited when you make a 44.1 kHz version of it. How is that different from natively recording at 44.1 kHz? You get the same band-limited frequency response anyway!



audiokangaroo said:


> We use to think that tape has a frequency response that is not as extended as what CD can provide.


Who we? I haven't though such a thing.   



audiokangaroo said:


> This is true if we consider frequency response at -3dB, but measuring this way does not make sense, because the useful amplitude of very high frequencies is generally under -40dB.


What does "useful amplitude" mean in this context?  



audiokangaroo said:


> If we measure the frequency response of a tape recorder around -50dB, we will have to consider that tape is indeed a high resolution format with a very extended frequency response.


Frequency-wise it might be somewhat "high resolution", but otherwise due to the distortions, noise, flutter and non-flat frequency response tape is NOT "high resolution".



audiokangaroo said:


> As of tape recording producing harmonic distortion, It may be considered problematic, but I don't think that it can explain the fact tape sounds more natural that basic
> digital audio.


How do you explain producing harmonic distortion makes something more "natural" than not producing harmonic distortion? My way of explaining this is that producing harmonic distortion the way tape does it makes the sound LESS natural, but MORE pleasing to ears that have gotten used to the distortion. So it is about confusion between more pleasing and more natural.

What you are do is you try to rationalize your subjective sonic preferences with trying to convince yourself and other about the superiority of analog sound because your ego doesn't allow you to admit to yourself you simply prefer LESS accurate and natural sound with distortion. There is nothing wrong with having subjective preferences. The problem is going against the science.


----------



## audiokangaroo

71 dB said:


> All that "very wide frequency response" gets band-limited when you make a 44.1 kHz version of it. How is that different from natively recording at 44.1 kHz? You get the same band-limited frequency response anyway!
> 
> 
> Who we? I haven't though such a thing.
> ...


People who have actually listened directly to an acoustic instrument like a piano or drums know how they sound. When I say this kind of instrument does sound better recorded on tape or high sample rate digital, I don't mean that they sound more pleasing. I mean that they actually sound more like the real thing.
We have the same issue with sound stage. Tape and high  sample rate digital produce a better sound stage than 44.1. Do you think that we could explain this with harmonic 
distortion ?


----------



## 71 dB (Aug 18, 2021)

audiokangaroo said:


> People who have actually listened directly to an acoustic instrument like a piano or drums know how they sound. When I say this kind of instrument does sound better recorded on tape or high sample rate digital, I don't mean that they sound more pleasing. I mean that they actually sound more like the real thing.
> We have the same issue with sound stage. Tape and high  sample rate digital produce a better sound stage than 44.1. Do you think that we could explain this with harmonic
> distortion ?


Why do I keep talking with you? You are not changing your mind so this is waste of time.

(Yes, I have heard acoustics instruments live. To my ears 44.1 does fine job with it. )

I may need to leave this forum not to go insane because all of these analog/hi-rez clowns.


----------



## audiokangaroo

71 dB said:


> Why do I keep talking with you? You are not changing your mind so this is waste of time.
> 
> (Yes, I have heard acoustics instruments live. To my ears 44.1 does fine job with it. )
> 
> I may need to leave this forum not to go insane because all of these analog/hi-rez clowns.


Let's stop talking for a while. I Don't think I'm going to change my mind. 
We would better listen to music and relax.


----------



## 71 dB

audiokangaroo said:


> We would better listen to music and relax.


I certainly agree with that!


----------



## old tech

audiokangaroo said:


> I read that Philips wanted tu use 14 bit for the CD format and that Sony advocated for 16 bit.
> It was not only for dynamic but also for linearity reasons. 16 bit files are enhanced with a dithering process to improve linearity.
> Did you actually experiment with 13 bit to hear how it sounds ?


Well here is a comparison between dithered 8 bit and 16 bit.  As you can hear, the only difference between the two bit depths is noise.  After watching this, how could you not believe that 13 bits is more than enough to capture vinyl?


----------



## old tech

audiokangaroo said:


> I think that tape and 192 KHz are a bit similar because they have both a vey wide frequency response.
> We use to think that tape has a frequency response that is not as extended as what CD can provide. This is true if we consider frequency response at -3dB, but measuring this way does not make sense, because the useful amplitude ofvery high frequencies is generally under -40dB. If we measure the frequency response of a tape recoder around -50dB, we will have to consider that tape is indeed a high resolution format with a very extended frequency response.
> As of tape recording producing harmonic distortion, It may be considered problematic, but I don't think that it can explain the fact tape sounds more natural that basic
> digital audio.


I've even posted the references and results of a controlled test between analog tape and CD recordings of live concerts which demonstrate that a majority of music major subjects rated the CD recordings as higher quality and more natural sounding - on several different speakers and headphones. But just like all the other objective information provided to you by others you simply ignore it and continue rambling on with your anti-science theories. I'm not sure why others continue with the dialogue. As 71 DB observed, you seem to vain to accept that your individual preference is less fidelity and try to dress that up with made up assertions. Be comfortable in your own skin as their is nothing wrong with having subjective preferences.


----------



## 71 dB

old tech said:


> I'm not sure why others continue with the dialogue. As 71 dB observed, you seem too vain to accept that your individual preference is less fidelity and try to dress that up with made up assertions.


Continuing dialogue long after the futility of it has been established is one of my weaknesses. Eventually I get frustrated and angry. I am forced to say to myself the dialogue serves nobody. Maybe in the distant past people practised source criticism, didn't ignore scientific facts  and didn't "choose" truth according to their personal taste. We are unfortunately living in a post factual World of _subjective_ truths. Of course greedy people take advantage of this and that's why there is so much disinformation out there. If you try to fight the disinformation all you get is frustrated and angry. Depressing.


----------



## 71 dB

old tech said:


> Well here is a comparison between dithered 8 bit and 16 bit.  As you can hear, the only difference between the two bit depths is noise.  After watching this, how could you not believe that 13 bits is more than enough to capture vinyl?



In all fairness, audiokangaroo seems to be more openminded to accept that 16 bit or even less can be enough for perfect consumer audio. It is the sample frequency/band-width were the disagreements are. I give credit where credit is due. It is good if audiokangaroo at least knows/understands that going beyond 16 bit in consumer audio doesn't increase accuracy/fidelity, but can only lower the noise floor to be even more inaudible.


----------



## castleofargh

71 dB said:


> Continuing dialogue long after the futility of it has been established is one of my weaknesses. Eventually I get frustrated and angry. I am forced to say to myself the dialogue serves nobody. Maybe in the distant past people practised source criticism, didn't ignore scientific facts  and didn't "choose" truth according to their personal taste. We are unfortunately living in a post factual World of _subjective_ truths. Of course greedy people take advantage of this and that's why there is so much disinformation out there. If you try to fight the disinformation all you get is frustrated and angry. Depressing.


I’m of the opinion that we’re not totally wasting our time. For kanga, it seems to be pointless, now we know it. But it’s a forum. Sometimes, people with doubts need to see irrational in action to reach their conclusions.



71 dB said:


> In all fairness, audiokangaroo seems to be more openminded to accept that 16 bit or even less can be enough for perfect consumer audio. It is the sample frequency/band-width were the disagreements are. I give credit where credit is due. It is good if audiokangaroo at least knows/understands that going beyond 16 bit in consumer audio doesn't increase accuracy/fidelity, but can only lower the noise floor to be even more inaudible.


Obviously. How would a belief in favor of mooooooore bit depth, work with a belief that tape, vinyl, and dsd are the better choices? Once you create that strange line up of ”fidelity” choices, you can't allow yourself to be obsessed with bit depth. Even made up facts would struggle to reconcile those ideas.


----------



## old tech

71 dB said:


> In all fairness, audiokangaroo seems to be more openminded to accept that 16 bit or even less can be enough for perfect consumer audio. It is the sample frequency/band-width were the disagreements are. I give credit where credit is due. It is good if audiokangaroo at least knows/understands that going beyond 16 bit in consumer audio doesn't increase accuracy/fidelity, but can only lower the noise floor to be even more inaudible.


Yes and in fairness that post was in response to his question to you whether you know how 13 bits sound. The video demonstrates that 8 bits dithered has the same accuracy as 16 bits (or 24 bits for that matter) but a higher noise floor. The noise floor of 13 bits is lower than vinyl or analog tape.


----------



## 71 dB

old tech said:


> Yes and in fairness that post was in response to his question to you whether you know how 13 bits sound. The video demonstrates that 8 bits dithered has the same accuracy as 16 bits (or 24 bits for that matter) but a higher noise floor. The noise floor of 13 bits is lower than vinyl or analog tape.


The answer is I do know what 13 bit audio sounds. I have tested it myself. It sounds the same as the 8 bit example of the video except the noise floor is 30 dB lower making it inaudible unless you listen to an extremely quiet part while turning the volume up exceeding reasonable listening levels.



castleofargh said:


> I’m of the opinion that we’re not totally wasting our time. For kanga, it seems to be pointless, now we know it. But it’s a forum. Sometimes, people with doubts need to see irrational in action to reach their conclusions.


You are right. Someone else not even participating in the discussion might be reading and learning.



castleofargh said:


> Obviously. How would a belief in favor of mooooooore bit depth, work with a belief that tape, vinyl, and dsd are the better choices? Once you create that strange line up of ”fidelity” choices, you can't allow yourself to be obsessed with bit depth. Even made up facts would struggle to reconcile those ideas.


Good point.


----------



## bigshot

gregorio said:


> It's not my explanation, it's the proven facts, both mathematically and in practice!



Welcome back Gregorio. The forum has suffered without your participation. I'm glad to see you back.


----------



## danadam

71 dB said:


> making it inaudible unless you listen to an extremely quiet part while turning the volume up exceeding reasonable listening levels.


IMO, it doesn't have to exceed reasonable levels in case of classical music and dither without noise shaping. Here's a sample made from "Hilary Hahn: The Complete Sony Recordings / Mendelssohn, Shostakovich - Violin Concertos / 01. Violin Concerto in E minor, op. 64: I. Allegro molto appassionato". It contains 18s excerpt starting at 6:47 (a quiet passage) and 18s excerpt starting at 11:39 (a loud passage). I also added 3s of silence at the beginning. Every 0.5s the sample has a dither applied at 13 bit level using SoX's "dither" effect with "-p 13" option. With volume level that doesn't make the loud passage unreasonably loud I can hear the the dither in the quiet passage starting at around 13s.
But with noise shaping, yes, it becomes inaudible (at least to me).


----------



## 71 dB (Aug 19, 2021)

danadam said:


> IMO, it doesn't have to exceed reasonable levels in case of classical music and dither without noise shaping. Here's a sample made from "Hilary Hahn: The Complete Sony Recordings / Mendelssohn, Shostakovich - Violin Concertos / 01. Violin Concerto in E minor, op. 64: I. Allegro molto appassionato". It contains 18s excerpt starting at 6:47 (a quiet passage) and 18s excerpt starting at 11:39 (a loud passage). I also added 3s of silence at the beginning. Every 0.5s the sample has a dither applied at 13 bit level using SoX's "dither" effect with "-p 13" option. With volume level that doesn't make the loud passage unreasonably loud I can hear the the dither in the quiet passage starting at around 13s.
> But with noise shaping, yes, it becomes inaudible (at least to me).


Yes, that is as demanding music sample as they come so definitely _shaped_ 13 bit dither (I can hear that non-shaped dither too). Note that this kind of "pulsing dither" of yours is more noticeable than constant dither, because it draws attention to itself. Real 13 bit system wouldn't be able to do this "pulsing".


----------



## gregorio

audiokangaroo said:


> If we measure the frequency response of a tape recoder around -50dB, we will have to consider that tape is indeed a high resolution format with a very extended frequency response.



And have you measured the frequency response of a tape recorder to signals "around -50dB"? If you haven't, then even by your own terms you do NOT have to consider that tape is a high resolution format!

Here are the measurements of pretty much all the most widely used studio tape recorders: http://www.endino.com/graphs/

*Note though:*
1. That these measurements are not in response to a -50dB signal but in response to an optimal (0dBVU) signal, they would obviously be worse for a -50dB signal.
2. These tests were run on freshly calibrated/aligned recorders. Which is NOT the case with in music studio recordings. Pro studio tape recorders were aligned once, at the beginning of each day, not for each recording pass.
3. A music analogue tape master is the result of at least 2 (and typically more) tape "generations", each generation introducing more noise, more distortion and more signal loss.
4. Notice the comparison with the last measurement, the frequency responses of the built-in ADC of a mid 1990's consumer desktop computer!!

It's unlikely that an acoustic ultrasonic signal at -50dB would be above the noise floor on a typical analogue music master.



audiokangaroo said:


> As of tape recording producing harmonic distortion, It may be considered problematic, but I don't think that it can explain the fact tape sounds more natural that basic
> digital audio.



You're right, it doesn't "explain the fact" because there is no scientific explanation of a fact that is false! For example, there is no scientific explanation for the fact that pigs can fly! In addition to the reliable evidence that's already been posted by Old Tech that demonstrates your fact is false, you also refuse to address the obvious question: How does adding unnatural noise and distortion make an acoustic instrument sound more natural?



audiokangaroo said:


> Tape and high sample rate digital produce a better sound stage than 44.1. Do you think that we could explain this with harmonic
> distortion ?



Same again: No, we have no scientific explanation for why figs can fly! The actual fact is that tape has worse soundstage than 44.1, due to crosstalk and other distortions, while higher sample rates have the same soundstage a 44.1.



Nickhasarrived said:


> he only reason I listen to 24/192 over 16/44.1 in most cases is because the song is often mastered differently, and you can tell when the artist has some fun in making objects move or adding reflections/reverberations in the 24/192 that the 16/44.1 doesn't have. Does that make one better than the other just from that? No each is a different listening experience.
> In terms of actual quality of a song that is mastered identically in 24/192 vs 16/44.1, I thought I heard a difference, but I am nearly 100% sure I couldn't tell in blind testing, and it was a phycological thing.



Unfortunately, your first assertion can often be true, because your second assertion is always true! In controlled double blind tests, no one has been able to distinguish 16/44.1 downsampled from a higher resolution master ("controlled" in this context means certain conditions, such as reasonable listening levels, typical filters, etc.). Of course, that's a very inconvenient fact if you want to charge more money for a 24/192 version than the 44.1kHz version. So it's not uncommon for record labels or distributors to change the 44.1kHz version enough so that there is an audible difference.

Incidentally, this "change" is usually just additional audio compression, rather than a change made to the reverberation or positioning, the latter isn't really practical to try and change during mastering as it's already been baked into the mix. However, as additional audio compression can change the freq response and the relative balance between the reverb and the direct signal, it can affect the perception of reverb and/or positioning.

G


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## audiokangaroo (Aug 20, 2021)

Gregorio, it looks like that you are considering audio phenomenons only through measurements and not through actual audition.
You said for instance that tape has worst soundstage than 44.1 because it has worst crosstalk results. However, thruth is that tape has better sound stage
than 44.1 in spite of having worst crosstalk. Listen carefully and you will hear it. In fact soundstage does not only depend on crosstalk but also on the way that the original
waveform is reproduced. This is the reason why frequency response is more important than crosstalk as far as sound stage reproduction is concerned.


----------



## sander99

audiokangaroo said:


> Gregorio, it looks like that you are considering audio phenomenons only through measurements and not through actual audition.





gregorio said:


> In controlled double blind tests, no one has been able to distinguish 16/44.1 downsampled from a higher resolution master





gregorio said:


> I for example, have done myself or been involved with numerous ABX tests and read about and/or discussed with other pro engineers numerous other unpublished ABX tests.


...
...
...
...
...


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## bfreedma (Aug 20, 2021)

audiokangaroo said:


> Gregorio, it looks like that you are considering audio phenomenons only through measurements and not through actual audition.
> You said for instance that tape has worst soundstage than 44.1 because it has worst crosstalk results. However, thruth is that tape has better sound stage
> than 44.1 in spite of having worst crosstalk. Listen carefully and you will hear it. In fact soundstage does not only depend on crosstalk but also on the way that the original
> waveform is reproduced. This is the reason why frequency response is more important than crosstalk as far as sound stage reproduction is concerned.



Your ability to repeatedly ignore actual facts while constantly manufacturing “facts” that don’t exist outside of your imagination is truly impressive.

Everyone here is wrong.  Known science is wrong.  Published testing is wrong.  But you, who have stated you don’t understand these topics well, are right…


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## gregorio (Aug 20, 2021)

audiokangaroo said:


> Gregorio, it looks like that you are considering audio phenomenons only through measurements and not through actual audition.


Hang on, YOU are the one that stated "If we *measure* the frequency response of a tape recorder..."! And, Contrary to your false assertion, I have actually auditioned professional studio tape recorders in professional studios, many times in fact! Have you?

If you haven't, what is it that you're considering? It's not "actual audition" or the measurements, which you're deliberately ignoring, so what's left? Your imagination, marketing or other made up nonsense?



audiokangaroo said:


> You said for instance that tape has worst soundstage than 44.1 because it has worst crosstalk results. However, thruth is that tape has better sound stage
> than 44.1 in spite of having worst crosstalk. Listen carefully and you will hear it. In fact soundstage does not only depend on crosstalk but also on the way that the original
> waveform is reproduced. This is the reason why frequency response is more important than crosstalk as far as sound stage reproduction is concerned.


I have listened very critically (to pro studio tape recorders and digital recorders) and heard the opposite, better sound stage with digital 44.1kHz. And, if frequency response is more important than crosstalk, then as the measurements prove, an extremely cheap 1990's built-in ADC has far more accurate frequency response then even the best studio tape recorders costing thousands of times more!

Again, how have you arrived at your "truth" (which is false)? What actual auditions have you done with studio tape recorders and/or what measurements?

G


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## Davesrose

audiokangaroo said:


> Gregorio, it looks like that you are considering audio phenomenons only through measurements and not through actual audition.
> You said for instance that tape has worst soundstage than 44.1 because it has worst crosstalk results. However, thruth is that tape has better sound stage
> than 44.1 in spite of having worst crosstalk. Listen carefully and you will hear it. In fact soundstage does not only depend on crosstalk but also on the way that the original
> waveform is reproduced. This is the reason why frequency response is more important than crosstalk as far as sound stage reproduction is concerned.


I'm not following this logic.  44.1 has better frequency response measurements and tape has more crosstalk.  If you state frequency response is more important than crosstalk for soundstage reproduction, then how is it tape is better?  It seems you have your preference for whatever analog setup vs whatever digital system you listen to (and then are filtering what data you'll accept vs considering changing your opinion if such data goes against your reasoning).  I think the simple explanation is that digital is better at accurate reproduction (completely flat frequency response within all auditory ranges): and analog will always have some distortions.  You prefer these distortions (people can prefer vinyl or tubes due to certain frequencies being rolled off).


----------



## castleofargh

@audiokangaroo is out of this thread for a week. I think we all deserve a break.


----------



## bigshot

Tape does not have better soundstage, because it has worse crosstalk. Crosstalk is directly linked to stereo imaging.


----------



## sonitus mirus

Troll or not, the recent replies to another "new" subscriber is a wonderful opportunity to clearly show how false assumptions and unscientific ideas are easy to refute.  I'm not sure if kangaroo actually believes readers are frustrated and he is "getting" to everyone or not, but the joke is on him, if there is a joke.


----------



## old tech

bigshot said:


> Tape does not have better soundstage, because it has worse crosstalk. Crosstalk is directly linked to stereo imaging.


I think sometimes, some people confuse accurate soundstage with phase errors or other distortions.

I remember a while back having a debate with a work colleague regarding his assertion that vinyl has a better, 'more 3D' sound stage than digital media.  I knew what he was on about as I have experienced a wider and, seemingly more enveloping sound stage when playing my vinyl. Putting aside mastering differences, it was clear to me that the wider sound stage was fake and due to incorrect alignment of the cart rather than reproducing the mix more accurately.  Most of this wider sound stage disappeared when I correctly aligned the cart.  I also noticed this with some of the cheaper turntables and cartridges.

I liken it to the sound I get from my clock radio when I press the "Wide" button. It does make the sound appear wider than the size of the clock radio, but obviously fake.  If Kanga prefers the fake or manufactured sound stage over what the artist/mixing/mastering engineer intended then there are better digital solutions with DSP.  Alternatively, he can stick with analog media/playback equipment and hook it up to a pair of Bose 901s and he'll be in heaven.


----------



## bigshot

It might be an error masquerading as the effect on timing of speakers in a room. But there is a lot more going on in a room when it comes to sound location.


----------



## 71 dB

old tech said:


> I think sometimes, some people confuse accurate soundstage with phase errors or other distortions.


People especially seem to confuse sonic accuracy with more pleasing sound.



old tech said:


> I remember a while back having a debate with a work colleague regarding his assertion that vinyl has a better, 'more 3D' sound stage than digital media.  I knew what he was on about as I have experienced a wider and, seemingly more enveloping sound stage when playing my vinyl. Putting aside mastering differences, it was clear to me that the wider sound stage was fake and due to incorrect alignment of the cart rather than reproducing the mix more accurately.  Most of this wider sound stage disappeared when I correctly aligned the cart.  I also noticed this with some of the cheaper turntables and cartridges.


The interesting thing about vinyl is that lateral movement of the needle corresponds "mid" -channel while vertical movement corresponds "side" channel. It is pretty clear that these two directions produce different distortions meaning the "mid" channel contains different extra harmonics due to non-linearities than the "side" channel and this creates the "smeared" and "rich" three-dimensional soundstage cloud people seem to love. The fact that elliptical filtering is used to avoid extreme vertical movement of the needle makes the sound more mono-like at low frequencies supporting the pleasantness of the soundstage (ILD increasing with frequency is "natural").

This kind of pleasantness can be taken advantage of also in the digital world: It is easy to process "mid" and "side" channels separately for this effect. Analog world can teach a lot about what is pleasant for the ear and we can apply that on the digital world. Best of both worlds! Digital recordings might be unnaturally "accurate" for simple reasons such as the microphones not moving at all while a person listening in a room will move/turn his/her head a little bit. The small inaccuraties and fluctuations of analog recorders might remove this "unnatural accuracy" while the transparency and accuracy of digital recorders do nothing about it. Real life sounds are always more or less "colorized". There are micro-fluctuations caused by the acoustic environment and the movements of the listener. This means there must be an expected level for accuracy for our hearing and digital recordings can surpass this level becoming "too accurate" which translates to "clinical cold sound". The beauty of digital audio is that we can control the accuracy of sound very precisely. We can for example saturate the sound just the amount we want to make it "naturally accurate." Or we can fluctuate the gain just a little bit to remove the clinical cold feel, but still retain full perceived accuracy.


----------



## gregorio

old tech said:


> I think sometimes, some people confuse accurate soundstage with phase errors or other distortions.



Soundstage is a perception and therefore only exists within each individual's brain. As with most other perceptions, it can be influenced by several different actual audio properties, freq response (obviously within the audible band), phase, crosstalk, etc. 

The logical black hole that @audiokangaroo can't avoid is that tape is less accurate than 16/44 for every single one of these audio properties! If tape were better for even just one of those properties, then it would be a more tricky argument but as it isn't, he hasn't got a leg to stand on. I'm sure that wouldn't stop him from trying though 

G


----------



## bigshot

Like the black knight in Monty Python and the Holy Grail... "It's just a flesh wound!"


----------



## BobG55




----------



## theaudiologist1

people talking about bits and kHz while in the real world the only important number that matters is the dynamic range. anything less than DR10 is garbage. And that's one great thing about SACD's: you're almost always guaranteed a great recording, or at the very worst an "OK" recording, but no brickwalled abominations. No need to waste days or weeks checking the dynamic range of every release and talking about which CD to get on Steve Hoffman.


----------



## bigshot

I’ve gotten bad SACDs. Not because of the format, but because the music was remixed poorly.


----------



## old tech

theaudiologist1 said:


> people talking about bits and kHz while in the real world the only important number that matters is the dynamic range. anything less than DR10 is garbage. And that's one great thing about SACD's: you're almost always guaranteed a great recording, or at the very worst an "OK" recording, but no brickwalled abominations. No need to waste days or weeks checking the dynamic range of every release and talking about which CD to get on Steve Hoffman.


The DR is an indicator but unless it is an extremely low number it cannot be assumed that a higher number is necessarily better.

A high DR number may say something about the dynamics of the recording but it doesn't say anything about the other qualities, eg quality of the source master, EQ choices, use of no-noise and so on.

One example is the double CD of best of 10cc.  It has a DR of 8 but it is a huge improvement over some of the earlier 10cc releases that have a DR of 13. The former is a bit louder but it is also clearer, cleaner and with more punch, without being fatiguing.


----------



## gregorio

theaudiologist1 said:


> people talking about bits and kHz while in the real world the only important number that matters is the dynamic range. anything less than DR10 is garbage. And that's one great thing about SACD's: you're almost always guaranteed a great recording, or at the very worst an "OK" recording, but no brickwalled abominations. No need to waste days or weeks checking the dynamic range of every release and talking about which CD to get on Steve Hoffman.


I would say pretty much the opposite, that in the real world, the DR number doesn't really matter at all. Firstly, it's a pretty suspect measurement in the first place. For example, it's been demonstrated that an LP can have a DR measurement 2-4 higher than the digital master from which it was cut. Secondly, it also depends on the DR we have to start with; let's say we have a recording of a composition that without any compression at all reads DR9 and a symphony recording has had a large amount of compression, say 16dB, which results in a DR11 reading. Is the first garbage and the second great?

In critical listening conditions, not having a really hammered recording is generally better but the DR measurement is not necessarily an indicator of whether the recording has been hammered with compression and some music genres are designed from the outset to be hammered any way!

G


----------



## theaudiologist1

bigshot said:


> I’ve gotten bad SACDs. Not because of the format, but because the music was remixed poorly.


The only bad one for me was the Alice In Chains Greatest Hits SACD.



old tech said:


> The DR is an indicator but unless it is an extremely low number it cannot be assumed that a higher number is necessarily better.
> 
> A high DR number may say something about the dynamics of the recording but it doesn't say anything about the other qualities, eg quality of the source master, EQ choices, use of no-noise and so on.
> 
> One example is the double CD of best of 10cc.  It has a DR of 8 but it is a huge improvement over some of the earlier 10cc releases that have a DR of 13. The former is a bit louder but it is also clearer, cleaner and with more punch, without being fatiguing.


There is no way that DR8 will sound good, specially with Rock or Classical. Maybe DR10 will sound better than DR13 if mastered well.



gregorio said:


> I would say pretty much the opposite, that in the real world, the DR number doesn't really matter at all. Firstly, it's a pretty suspect measurement in the first place. For example, it's been demonstrated that an LP can have a DR measurement 2-4 higher than the digital master from which it was cut. Secondly, it also depends on the DR we have to start with; let's say we have a recording of a composition that without any compression at all reads DR9 and a symphony recording has had a large amount of compression, say 16dB, which results in a DR11 reading. Is the first garbage and the second great?
> 
> In critical listening conditions, not having a really hammered recording is generally better but the DR measurement is not necessarily an indicator of whether the recording has been hammered with compression and some music genres are designed from the outset to be hammered any way!
> 
> G


DR8-12 are what will depend on the genre, but less than DR8 is objectively horrible no matter what genre it is, and sadly modern recrodings have dynamic ranges of 4-5 and some I saw even 2! It's impossible to be a non-classical audiophile these days.


----------



## old tech

You have no idea ^


----------



## gregorio

theaudiologist1 said:


> There is no way that DR8 will sound good, specially with Rock or Classical. Maybe DR10 will sound better than DR13 if mastered well.
> 
> DR8-12 are what will depend on the genre, but less than DR8 is objectively horrible no matter what genre it is, and sadly modern recrodings have dynamic ranges of 4-5 and some I saw even 2! It's impossible to be a non-classical audiophile these days.



I would think that a completely uncompressed recording of John Cage's 4:33 could have a DR of about 1, certainly less than 8! Would that be objectively horrible? Subjectively you might not like the piece but it wouldn't be objectively horrible and the same could be said of many pieces. Even with more traditional classical music, massive amounts of compression can be subjectively better, the hugely compressed Classic FM is the only way I can listen to classical music when driving for example but I certainly wouldn't find it subjectively better when listening critically in a more controlled environment.

The DR measurement can be an indicator of too heavily applied compression/limiting but not always and even when it is, it's not always relevant any way!

G


----------



## 71 dB

theaudiologist1 said:


> The only bad one for me was the Alice In Chains Greatest Hits SACD.


SACD has practically two advantages (compared to CD):

1) Multichannel support
2) Recording and production for SACD releases are _often_ carefully done meaning even the stereo downmixed SACD/CD versions on the disc sound great.

That's about it. If you are re-realising old stereophonic recordings on stereo only SACD you don't have these benefits. Of course you can make a new awesome sounding stereo remaster of the old stuff and have a good sounding disc, but you don't need SACD for that. CD is fine. Only if you create a new multichannel mix of old material does it make sense to release it on SACD. I don't know how Alice in Chains Greatest Hits is. Not my type of music. As a general rule, classical music on SACD is awesome thanks to new multichannel recordings done in places of natural acoustics. 



theaudiologist1 said:


> There is no way that DR8 will sound good, specially with Rock or Classical. Maybe DR10 will sound better than DR13 if mastered well.


Well, nobody in their right mind would reduce the dynamic range of a classical music recording into DR8. Meanwhile, modern pop music uses small dynamic range skilfully (dynamic compression is crucial part of sound design) and DR6 can sound pretty dynamic. The same DR6 applied to a symphony would give comical results.



theaudiologist1 said:


> DR8-12 are what will depend on the genre, but less than DR8 is objectively horrible no matter what genre it is, and sadly modern recrodings have dynamic ranges of 4-5 and some I saw even 2! It's impossible to be a non-classical audiophile these days.


As I mentioned above, modern pop music uses dynamic compression in the sound design and mixing cleverly so that it works at lower DR values. Commercial music tends to be more compressed while less commercial, more "artsy" genres of music tend to offer bigger DR values. Classical music is not the only option, but one has to dig deeper beyond the commercial surface.


----------



## bigshot

Do any of us know how those dynamic range numbers are assigned? It seems to me that there is a lot more to dynamics than can be reduced to a number. And there are situations in music where something dynamic might not hit the algorithm right to reflect it. Judging dynamics is something that would be hard to measure without subjective judgement I would think.


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## theaudiologist1 (Aug 30, 2021)

71 dB said:


> 1) Multichannel support



I actually don't care much for multichannel since

1) I'm a headphone audiophile.
2) The extracted 5.1 DSD files are HUGE. (8GB+ for an album in DSD64)
3) I don't know a headphone DAC/AMP that can do 5.1.
4) The LFE channel is completely useless on headphones

So I guess I'm part of the <1% who cares about the stereo layer.


71 dB said:


> As I mentioned above, modern pop music uses dynamic compression in the sound design and mixing cleverly so that it works at lower DR values. Commercial music tends to be more compressed while less commercial, more "artsy" genres of music tend to offer bigger DR values. Classical music is not the only option, but one has to dig deeper beyond the commercial surface.


My problem is that even for rock even remasters of old songs are brickwalled to death and sound terrible. It's one thing to compress it a bit it's another to brickwall it. All the CD masters from the 80s up to the mid 90s had good DR scores. Only since then we've gotten terrible results. All the albums that sound good with a low DR sound good in spite of, not because of, the low DR.


71 dB said:


> 2) Recording and production for SACD releases are _often_ carefully done meaning even the stereo downmixed SACD/CD versions on the disc sound great.



wait the stereo versions are downmixes of the 5.1? You can't do downmixing in the DSD realm. That means...


----------



## 71 dB

theaudiologist1 said:


> I actually don't care much for multichannel since
> 
> 1) I'm a headphone audiophile.
> 2) The extracted 5.1 DSD files are HUGE. (8GB+ for an album in DSD64)
> ...


So why care about SACD at all?



theaudiologist1 said:


> My problem is that even for rock even remasters of old songs are brickwalled to death and sound terrible. It's one thing to compress it a bit it's another to brickwall it. All the CD masters from the 80s up to the mid 90s had good DR scores. Only since then we've gotten terrible results. All the albums that sound good with a low DR sound good in spite of, not because of, the low DR.


Get the old releases cheap/used. People have "upgraded" their CDs with the newer brickwalled ones so the old ones are looking for a new home.



theaudiologist1 said:


> wait the stereo versions are downmixes of the 5.1? You can't do downmixing in the DSD realm. That means...


Of course they are. They record in DSD format, then transform (integrate) it into PCM to "do the magic that you can't do to DSD" including making the stereo downmixes and the go back to DSD for the SACD disc.


----------



## sander99

theaudiologist1 said:


> 3) I don't know a headphone DAC/AMP that can do 5.1.


A very convincing binaural simulation of loudspeakers in a room is possible with DSP. But for good results you need personalisation to match your hrtf (head related transfer function).

Two examples:

Smyth Realiser A16 (has DACs and amps but you can also just feed the processed digital signal to your own DACs and amps).
It can use headtracking to keep the virtual speakers at fixed positions. It can decode dolby atmos up to 16 or 24 (newer version) channels.

Impulcifer (free software by @jaakkopasanen, but you will need in-ear-mics) to do measurements and create a personal preset that can be used with other software, for example HeSuVi (also free). With this you can do 7.1 channels max. and no headtracking.
https://github.com/jaakkopasanen/Impulcifer
https://www.head-fi.org/threads/recording-impulse-responses-for-speaker-virtualization.890719/

Both these solutions require measurements of real speakers in a room (and of the headphones that you want to use) with in-ear-mics.


----------



## bigshot (Aug 30, 2021)

It's fine if you can't do multichannel. My point was that when they did a multichannel mix they also cleaned up the stereo versions. That is the master used for your LP release. You say that the CD and streaming versions of the Doors albums aren't as good as the collectors' LPs. That isn't true. The exact same remaster you have is available on Apple Music streaming, as well as SACD and CD in an audibly transparent format (LPs are not transparent.)

It's not just the Doors catalog that you are wrong about. Your blanket assessment of CDs as being "brick walled to death" just isn't true. Some are, some sound fantastic. It's actually gotten better since the 2000s. But we've gone over and over that before and you insist on being a broken record of misinformation. It's the only subject you ever talk about and you never listen to anything anyone else says. Tiresome.

In addition to the Smyth Realiser, Apple is working on developing Dolby Atmos for AirPod headphones. They currently have both the spatial audio and remastered stereo versions of the Doors albums in Apple Music. Spatial audio isn't terribly spatial yet, but the Doors tracks in the Apple streaming service sound fantastic.


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## danadam

bigshot said:


> Do any of us know how those dynamic range numbers are assigned?


The original algorithm (called TT DR Offline Meter) is described here: https://web.archive.org/web/2018091....de/sites/default/files/Measuring DR ENv3.pdf
The new one (called MAAT DROffline) is not revealed, AFAIK.


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## bigshot (Aug 30, 2021)

Math isn't my strongest subject, but I'm figuring that the amount of time the peak occupies has a big impact on the rating. A classical piece may have a huge dynamic range from low level to a big crescendo, but if the work consists mostly of low level, it would rank as a low dynamic number. In something like Dark Side of the Moon, individual songs would have completely different dynamic ratings, and an overall rating for the whole album would be averaged so much, you might not know if there really are wide peaks and valleys in there. I'm guessing this scale works best with fairly constant dynamics distributed equally throughout a whole album side, or with albums that have long stretches of uninterrupted peak and long stretches of uninterrupted quiet.

In short, this is fine as a ballpark figure, but individual circumstances of the way the peaks and valleys are laid out in time can make it vary a lot. One album rated as a six might sound much more dynamic than a different album rated as a six. So you can't necessarily say that a six is more dynamic than a five with different albums, but you can with two releases of the same album that measure that way. Am I right?


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## 71 dB (Aug 30, 2021)

bigshot said:


> Math isn't my strongest subject, but I'm figuring that the amount of time the peak occupies has a big impact on the rating. A classical piece may have a huge dynamic range from low level to a big crescendo, but if the work consists mostly of low level, it would rank as a low dynamic number. In something like Dark Side of the Moon, individual songs would have completely different dynamic ratings, and an overall rating for the whole album would be averaged so much, you might not know if there really are wide peaks and valleys in there. I'm guessing this scale works best with fairly constant dynamics distributed equally throughout a whole album side, or with albums that have long stretches of uninterrupted peak and long stretches of uninterrupted quiet.
> 
> In short, this is fine as a ballpark figure, but individual circumstances of the way the peaks and valleys are laid out in time can make it vary a lot. One album rated as a six might sound much more dynamic than a different album rated as a six. So you can't necessarily say that a six is more dynamic than a five with different albums, but you can with two releases of the same album that measure that way. Am I right?


The music is cut into 3 seconds long blocks. The the _rms_ and _peak_ values are calculated for each block. Histograms of the rms and peak values on dB scale are built. Then the rms sum of the 20 % largest rms values are compared to the 2nd largest peak value, and this gives the DR value for each channel. The final DR value is the average of channel DR values rounded up to integer value. So, this 20 % rule does affect things.


----------



## bigshot

How does that relate to different kinds of music? Does a 6 for 1812 Overture with a huge cannon blast peak far above the music sound like the same dynamics as a 6 heavy metal song that is loud through most of the track?


----------



## 71 dB

bigshot said:


> How does that relate to different kinds of music? Does a 6 for 1812 Overture with a huge cannon blast peak far above the music sound like the same dynamics as a 6 heavy metal song that is loud through most of the track?


If the cannon blast happen inside one 3 second frame it only affects the largest peak. If it happen between two frames, the other frame become the 2nd largest and the cannon affects DR.


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## bigshot (Aug 31, 2021)

Hmmm... And I'm assuming that a single song on an album can have a quite different rating than the album as a whole.


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## 71 dB

bigshot said:


> Hmmm... And I'm assuming that a single song on an album can have a quite different rating than the album as a whole.


Yes, of course if the album has been made that way. Just as the tempo of the songs can vary a lot. Some artists may want to have commercial radio-friendly tracks (Typically lower DR value) to sell the album (they even have a sticker on the album cover saying "includes the hits songs..."), but also more artistic and less commercial tracks (typically larger DR value).


----------



## bigshot

I'm guessing that professional engineers don't use a semi-arbitrary rating system like this to master the dynamic ranges. They do it by ear creatively, and I bet different parts of the music have different dynamic ranges by design.


----------



## 71 dB

bigshot said:


> I'm guessing that professional engineers don't use a semi-arbitrary rating system like this to master the dynamic ranges. They do it by ear creatively, and I bet different parts of the music have different dynamic ranges by design.


That's what the professional engineers would do if they had creative freedom, but often they don't have. The clients want "LOUD" mixes, because for most people louder means better.


----------



## bigshot

But they judge it by that numerical rating, not by ear?

There is creativity in creating mixes for different demographics and purposes. Engineers don't mix solely to their own taste. They are serving an audience. The creativity comes with balancing the practical aspects with the expressive ones. Admittedly though, most of the creativity comes with creating the music, not engineering it.


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## VNandor

71 dB said:


> That's what the professional engineers would do if they had creative freedom, but often they don't have. The clients want "LOUD" mixes, because for most people louder means better.


As far as I know, all the major streaming platforms have started using their own idea of loudness normalization. I sure hope people will eventually catch up to that fact. If the track was too loud it would be simply turned down to a certain maximum accepted loudness. Spotify also has a feature on by default that actually turns up and limits tracks that's below their target range. This can be fortunately turned off though.


----------



## danadam

VNandor said:


> Spotify also has a feature on by default that actually turns up and limits tracks that's below their target range. This can be fortunately turned off though.


For some time already the limitting is only done in the "Loud" preset, which is not a default one. From https://artists.spotify.com/help/article/loudness-normalization


> *Positive gain* is applied to softer masters so the loudness level is -14 dB LUFS. We consider the headroom of the track, and leave 1 dB headroom for lossy encodings to preserve audio quality.
> *Example:* If a track loudness level is -20 dB LUFS, and its True Peak maximum is -5 dB FS, we only lift the track up to -16 dB LUFS.
> ...
> 
> ...


----------



## Davesrose

71 dB said:


> That's what the professional engineers would do if they had creative freedom, but often they don't have. The clients want "LOUD" mixes, because for most people louder means better.


From what I've gathered, it also depends on genre.  Certainly one would think popular rock is the genre that has progressively gotten compressed and "LOUD".  When I did collect SACD, it was primarily classical (and then jazz).  If dynamic range was for the entire song, then I've always been impressed by Ravel's Bolero: where it goes from barely a register of instruments to huge symphonics.  Then also I have my speaker system for music and movies.  Good movie tracks also have a dynamic range of quiet passages to possibly quite loud effects in volume and bass.  One of my favorite movies for plot and sound is Master and Commander.  There's plenty of scenes with just dialogue and ambient ship noises....but then it's interspersed with loud passages of canon fire.  There's a reason why it won the academy award for sound design....and I'd automatically buy it if it makes it to 4K with Atmos.


----------



## bigshot

It has to do with the intended audience and the situation the music will be heard in. You don't want to mix an album the same for listening in a silent living room as you do a car or in earbuds on a train. The stuff that is compressed is the music intended for portable use. I've noticed in the past couple of years that CDs aren't hot mastered as much any more. Perhaps some genres are more apt to be played while on the go. You wouldn't master a Sibelius symphony to sound good on a bus.


----------



## Davesrose

bigshot said:


> It has to do with the intended audience and the situation the music will be heard in. You don't want to mix an album the same for listening in a silent living room as you do a car or in earbuds on a train. The stuff that is compressed is the music intended for portable use. I've noticed in the past couple of years that CDs aren't hot mastered as much any more. Perhaps some genres are more apt to be played while on the go. You wouldn't master a Sibelius symphony to sound good on a bus.


Environment seems to be irrelevant towards the trend of hotness (IE rock continuing to get compressed).  Having mixed media existed in the 80s: where you had your album going to CD and vinyl, but also Walkman (and say cheap headphones) and cassette tape in car.  So with digital masters, it's always been a mixed bag when it comes to delivery.  I would say that the CDs I have from the 80s that I really cherish are classical music: where I am isolating to good conditions whether it being my good headphone setup or speaker system.  Then again, some of them I did get when I was a kid then (and I played them via cheaper headphones over a Sony boombox).  I got into the performance, and it's great that uncompressed wave should be archival.

When it does come to cars, though.....I have had tried keeping a Sony Discman working that is good about a "surround" DSP with headphones, and also has another DSP for cars (which tries to have an EQ good for compensating with background car noises).  Hard, though, now....as it could skip (and later generations had ESP to have some buffer memory to not pick up the skip).

When it comes to mastering, I'm not involved with sound....but I am familiar with graphics.  Yes, when you're authoring, you're not accounting for least common denominator , but you might account for some.  For me, I try to author content on a calibrated monitor and set color profiles....and I know there will be plenty of people who will be viewing on a display that's not calibrated.  However, these days, it's better that given monitors and systems are getting more neutral in color.  It's not anything like say my master's project in grad school: which was showing 3D animations with my content expert.  When I first loaded them on his NEC montior, we couldn't see anything.  His contrast settings were so atrocious, that it was just black!

Anyway, I also just responded about Apple Music getting more songs being lossless format: these arguments about the smallest lossy formats validity are getting outdated.


----------



## gregorio

Davesrose said:


> From what I've gathered, it also depends on genre. Certainly one would think popular rock is the genre that has progressively gotten compressed and "LOUD".


Popular rock and pretty much all the rock sub-genres were always loud compared to the other genres of their day. In fact fact for some of them, loudness was an overt selling point, bands competed to be the loudest for quite a long time. The problem started in the 1990's, when new genres evolved which incorporated new technology, like sampling. Some/Many of these genres were built from the ground up to incorporate heavy compression, they achieved a "dynamic range" by orchestration rather than a by loud/quieter passages. EG. A verse could be little more than just a vocal and say a bass/kick drum. Even if you compress/limit it to death, it's still going to be perceived as much quieter than the choruses, which are richer in freq content, even though the peak levels are the same. This presented rock genres with a problem, they had to compete on loudness with genres specifically designed for the latest loudness technology (of the 1990's) and what was an acceptable amount of compression/limiting for most of these new genres, was way too much for the much older rock genres (that were designed for the technology of the 1960's and '70's).


Davesrose said:


> Environment seems to be irrelevant towards the trend of hotness (IE rock continuing to get compressed).


Not irrelevant, it's a contributing factor, along with various other factors. In fact many years ago in this very thread, a representative of a Hi-res distributor (it might have been Linn Records, HDTracks or another company, I can't remember) specifically stated that they require the mastering engineer to add extra compression to their CD versions, because they expected the CD to be ripped to a lossy format and played on an iPod in poor listening environments. Although this is just an excuse IMO, to introduce an audible difference between the CD and Hi-res versions to justify the higher price. And of course, even the very start of the loudness wars was largely environmental, getting a louder mix on Duke Boxes and on the radio. EDM for example is specifically designed for packed night clubs, hardly an ideal listening environment!

G


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## VNandor

danadam said:


> For some time already the limitting is only done in the "Loud" preset, which is not a default one. From https://artists.spotify.com/help/article/loudness-normalization


Damn just when I told off some people for not keeping up with the times.


----------



## theaudiologist1

71 dB said:


> So why care about SACD at all?


Because SACD versions are usually definitive masters, regardless of format. Look at the DSoTM SACD. And also I don't want to compare 10000 CD masters to see which one is good.


71 dB said:


> Get the old releases cheap/used. People have "upgraded" their CDs with the newer brickwalled ones so the old ones are looking for a new home.


I do this, but new music only has new masters, so there is not much choice. It's one thing keeping me from trying new music (another that new music is garbage).



bigshot said:


> But they judge it by that numerical rating, not by ear?
> 
> There is creativity in creating mixes for different demographics and purposes. Engineers don't mix solely to their own taste. They are serving an audience. The creativity comes with balancing the practical aspects with the expressive ones. Admittedly though, most of the creativity comes with creating the music, not engineering it.


The numbers tell 90% of the story. Don't tell me a mix with DR6 is going to sound good. That's low even for electronica.

I'm just curious why they keep compressing CD's now that the "normal" crowd moved on to iPods or whatever new thing there is now.


----------



## bigshot

OK


----------



## gregorio

theaudiologist1 said:


> [1] The numbers tell 90% of the story. Don't tell me a mix with DR6 is going to sound good. That's low even for electronica.
> 
> [2] I'm just curious why they keep compressing CD's now that the "normal" crowd moved on to iPods or whatever new thing there is now.


1. You keep changing your tune, you started by saying "anything less than DR10 is garbage", now you're saying DR6. You seem to be making it up as you go along, which isn't acceptable in the sound science forum! It is entirely possible a DR6 mix could "sound good", for some genres in some listening environments.

2. You answered that question yourself and I answered it in my previous post. "_Because SACD versions are usually definitive masters_" and as SACD is a largely dead format, other so called hi-res masters are often the definitive masters because they can charge significantly more for a definitive master in a so calle hi-res container than they could for the same definitive master in 16/44 format and the reason they can do that is because the audiophile community has fallen for the Hi-res BS marketing. And finally, the quickest, easiest, cheapest and most easily justifiable way to make a 16/44 version sound worse (in critical listening environments) than the hi-res version is simply to add more compression to it.

G


----------



## castleofargh

Isn't it great to know that a song is bad by looking at one single DR value calculated in a pretty arbitrary way? From a similar logic, I would also suggest to avoid music that doesn't use all the notes, or music that only relies on a handful of instruments.







https://dr.loudness-war.info/album/list?artist=Ed+Sheeran&album=
https://dr.loudness-war.info/album/list?artist=rihanna&album=
Rihanna has apparently sold about as many albums as Pink Floyd. Proof that people really love bad sounding music. Or something, I'm not good at math.


----------



## bigshot (Sep 3, 2021)

I have great trust in numbers to define fidelity. How much different is the sound than the way it is supposed to be? That can be answered.

But I've worked with enough artists to know that creative decisions cannot be quantified like that. Musical dynamics, like many other aspects of balancing a mix are creative. There is an intent that can't be quantified. For one piece of music, a 4 might be perfect. For another you need the 11 on the scale of 1 to 10. You can't run Sgt Pepper through an algorithm to determine if the Beatles were brilliant musicians.

A creative work can be destroyed by ham handed rethinking. Or it can be improved by a revision that heightens the emotional impact. I've heard SACDs that do both of those things. You can't judge music by numbers or formats. You can only judge fidelity that way. And sometimes there are things that are more important than fidelity. Like expression. So you listen, analyze what you hear, and decide for yourself if it works or not. No machine will think for you. Relying on some arbitrary number spit out by an algorithm to judge a creative decision is dumb.

All that said, mastering is a process that serves a purpose. How well it serves the purpose is what matters, not whether it measures the same as when it was mastered for a completely different purpose. This has been explained many times to our friend and he doesn't listen. We spend too much of our time explaining things to people who don't listen.


----------



## 71 dB

theaudiologist1 said:


> Because SACD versions are usually definitive masters, regardless of format. Look at the DSoTM SACD. And also I don't want to compare 10000 CD masters to see which one is good.


Definitive masters? Well, I suppose sometimes they are. All my SACDs are classical music especially from BIS label, but from other labels also such as CPO. I suppose for them the SACD is the "definitive" master and to my experience the CD layer sounds exactly the same as the stereo SACD layer and the only difference to multichannel version is the amount of channels. These labels typically only release the hybrid SACD so there is no choosing.



theaudiologist1 said:


> I do this, but new music only has new masters, so there is not much choice. It's one thing keeping me from trying new music (another that new music is garbage).


Life is easy when there is less choice. Also, "new" music typically suffers less from low DR values than "old" music because of the ways it is constructed. If you think new music is garbage (some of it is, but not all just as some of old music is also junk) then just listen to old music using old high DR releases.


----------



## bigshot (Sep 3, 2021)

71 dB said:


> to my experience the CD layer sounds exactly the same as the stereo SACD layer and the only difference to multichannel version is the amount of channels.



Multichannel mixes are by definition new mixes. The balances, application of filters and reverbs and even sometimes the takes used are different. The only reason that they sound the same is because when they do a remix for multichannel, they export a 2 channel fold down too. It doesn't sound like the original album. Do a direct A/B comparison of the multichannel mix to the original album mix on a CD and you'll see. I have a Rolling Stones SACD where the multichannel version doesn't include the brass section that is on the CD layer.


----------



## Ryokan

There's a lot of talk about buying a Dap that can un-fold MQA files to their fullest extent, but I'm guessing, like the Hi-Res files I own, I wouldn't hear a difference from the equivalent 256- 320kbps Mp3?


----------



## bigshot

There's evidence that MQA isn't even as good as high data rate lossy.


----------



## Ryokan

bigshot said:


> There's evidence that MQA isn't even as good as high data rate lossy.



I actually leave off large hi-res files now, which I once gave preference to when buying, in favour of 320 mp3 or flac, they're a third of the size, or less, and the player has to do less work so I get more out of the battery.


----------



## 71 dB

bigshot said:


> Multichannel mixes are by definition new mixes. The balances, application of filters and reverbs and even sometimes the takes used are different. The only reason that they sound the same is because when they do a remix for multichannel, they export a 2 channel fold down too. It doesn't sound like the original album. Do a direct A/B comparison of the multichannel mix to the original album mix on a CD and you'll see. I have a Rolling Stones SACD where the multichannel version doesn't include the brass section that is on the CD layer.


I am talking about new recordings of _classical music_ where the multichannel version is THE original version and it is just downmixed to stereo SACD and CD layers. Outside classical music hardly any of my favorite music has ever been released on SACD. Tangerine Dream has one, "Rubycon", but because I didn't get into Tangerine Dream until 2008 this release was already OOP and really expensive, so I got myself the CD version. Generally I am not into rock music, but I enjoy King Crimson, which to my knowledge doesn't have SACD releases. Most of my non-classical music favourites are more or less marginal stuff so it is no wonder there never was SACD. For marketing reasons the "the biggest names" got their SACDs, but I am not into "the biggest names". I am into "music I like". Very rarely that means "big names". King Crimson and Tangerine Dream are among the "biggest names" among my favourites and even they are marginal obscure groups for many. Classical music has always been so much better served on SACD, because it makes so much more sense and on that side I'm into many "big names". Obscure composers are releases on SACD too! So for me SACD is 100 % a classical music format.


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## bigshot

The early Tangerine Dream catalog was release on blu-ray in multichannel mixes in a box set. It's fantastic. The later albums were released in 2 channel and that box is only useful for the live recordings.

Just about all of King Crimson is released in multichannel. They do DVD-A.

Most releases are DVD or blu-ray, not SACD any more.


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## Sonic Defender

I have a DAC that is perfectly audibly transparent, Gustard X16 and I am now thinking about getting some kind of digital transport (DVD or Blu-ray) that can output the digital stream unadulterated which I could then pass through my DAC. Sadly I know nothing about such devices as I went right from CD transports to ripping my collection for computer based audio. Any suggestions for a DVD or Blu-ray player that I might consider? Almost exclusively for audio, not sure if I would ever use it for video, but probably a little.


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## bigshot (Sep 4, 2021)

Blu ray and many DVDs are encrypted. If you want to rip them use an app to make mkv. You’ll need a Blu-ray drive for your comp. it’s a lot easier just to play the disk using a standalone player.

EDIT: Aha! The question changed after I answered it.


----------



## theaudiologist1

gregorio said:


> 1. You keep changing your tune, you started by saying "anything less than DR10 is garbage", now you're saying DR6. You seem to be making it up as you go along, which isn't acceptable in the sound science forum! It is entirely possible a DR6 mix could "sound good", for some genres in some listening environments.
> 
> 2. You answered that question yourself and I answered it in my previous post. "_Because SACD versions are usually definitive masters_" and as SACD is a largely dead format, other so called hi-res masters are often the definitive masters because they can charge significantly more for a definitive master in a so calle hi-res container than they could for the same definitive master in 16/44 format and the reason they can do that is because the audiophile community has fallen for the Hi-res BS marketing. And finally, the quickest, easiest, cheapest and most easily justifiable way to make a 16/44 version sound worse (in critical listening environments) than the hi-res version is simply to add more compression to it.
> 
> G


1)I probably exaggerated. But a release with a dynamic range less than 10 means it's compressed, specially for rock, jazz and classical. The minimum I accept is 10 for rock, 12 for Classical and 8 for electronica. DR6 is just bad.

2)Many "hirez" releases from HDTracks are also just upsamples from 16/44.1. Not only that but they are heavily compressed (yes even the hi-res versions).


bigshot said:


> There's evidence that MQA isn't even as good as high data rate lossy.


Yeah it sounds horrible. It's also a proprietary DRM format so you should stay away.


71 dB said:


> I am talking about new recordings of _classical music_ where the multichannel version is THE original version and it is just downmixed to stereo SACD and CD layers.


Where did you get that the stereo DSD mix is a downmix of the multichannel?


----------



## 71 dB

bigshot said:


> The early Tangerine Dream catalog was release on blu-ray in multichannel mixes in a box set. It's fantastic. The later albums were released in 2 channel and that box is only useful for the live recordings.


Okay, but when I started collecting Tangerine Dream in 2008 CDs were the only ones available and even on CD a lot of Tangerine Dream is hard to find. I collected about 90 CDs of Tangerine Dream during the time period 2008, 2009 and 2010, but after that it has been more difficult except for the new releases and I am not a billionaire who is willing to pay $50 or $100 on eBay for an OOP rarity. 



bigshot said:


> Just about all of King Crimson is released in multichannel. They do DVD-A.


Yes, I have the "Radical Action..." boxset with one _Blu-ray_ disc. I happened to discover King Crimson also in 2008. At that time I used a lot of time and energy to explore what 70's has to offer (for me) and Tangerine Dream and King Crimson were the result of this. I was never educated about this stuff in my childhood, because my dad listened to Jazz only and my friends in school listened to 80's rock/pop music and personally I didn't really actively listen to music until in late 80's in high-school. Because I am so introverted and do not interact much with "outer world", it takes me a long time to discover stuff I like (before the internet at least) and also my taste tends to be more eclected and "weird" than for average people, because I don't identify myself as member of any "group" listening to certain type of music. I just listen to anything I like from Buxtehude's cantatas to Fauré´s chamber music to Gato Barbieri's latin jazz to Ke$ha's pop to Jonny L's drum'n'bass to Autechre's abstract sonic sound-worlds. That's why I felt myself alien on Tangerine Dream discussion board. Other TD fans tend to listen to electronic/synth music only and do not care about classical music at all nor do they care about electronic dance music for that matter. So if I mention to them Buxtehude, Fauré, Barbieri, Ke$ha, Jonny L and Autechre they are like "what?" and someone may tell Autechre's 3 first albums "Incunabula", "Amber" and "Tri Repetae" are kind of cool, but the others are too "difficult."



bigshot said:


> Most releases are DVD or blu-ray, not SACD any more.


Yes, except for classical music which still sees SACD releases. Of the six newest releases of BIS no less than five are hybrid SACDs and only one is a regular CD.


----------



## 71 dB

theaudiologist1 said:


> Where did you get that the stereo DSD mix is a downmix of the multichannel?


What else could it be?


----------



## sander99

71 dB said:


> What else could it be?


Just another mix from the multi_track_ recordings, but now in stereo?


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## 71 dB

sander99 said:


> Just another mix from the multi_track_ recordings, but now in stereo?


Why make another mix when you can just downmix stereo versions from the multichannel version?


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## bigshot

They generally do two mixes when they do a multichannel mix... multichannel and stereo. The stereo mix is secondary. They have the mix up on the board, so they just adapt it to work on stereo too. It doesn't take a long time to do, but it is a little more than just a fold down. Not a big deal though. It's not like they are mixing it from scratch.


----------



## gregorio

Sonic Defender said:


> [1] I have a DAC that is perfectly audibly transparent, Gustard X16 and [2] I am now thinking about getting some kind of digital transport (DVD or Blu-ray) that can output the digital stream unadulterated which I could then pass through my DAC. Any suggestions for a DVD or Blu-ray player that I might consider? Almost exclusively for audio, not sure if I would ever use it for video, but probably a little.


1. So does pretty much everyone.

2. Any half decent BR player should do the job perfectly, assuming you're outputting the digital signal to a DAC. A BR player is obviously more future proof than a DVD player but you'd need an AVR to take advantage of all the audio formats that BR supports.

G


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## gregorio

theaudiologist1 said:


> 1)I probably exaggerated. [1a] But a release with a dynamic range less than 10 means it's compressed, specially for rock, jazz and classical. The minimum I accept is 10 for rock, 12 for Classical and 8 for electronica. DR6 is just bad.
> 2)Many "hirez" releases from HDTracks are also just upsamples from 16/44.1. Not only that but they are heavily compressed (yes even the hi-res versions).
> [3] Where did you get that the stereo DSD mix is a downmix of the multichannel?


1. You seem to consistently overlook the fact that this is the Sound Science subforum and science does NOT allow exaggeration! We can be a bit more forgiving here than science allows but still your assertions either need to be accurate (NOT exaggerated) to start with or you need to qualify your assertion; as a guess, opinion or a possible/probable exaggeration.
1a. Rock music is ALWAYS compressed and with say classical music, a release with a dynamic range of less than about DR20 probably means it's compressed. Although of course that depends on the piece and it's orchestration, an unaccompanied piano obviously doesn't have the same dynamic range as a full symphony orchestra. The question is therefore not if it's compressed but compressed by how much and whether it's an inappropriate amount? And that depends on the genre and individual composition. Again, DR6 is not "just bad", it may represent an appropriate amount of compression for some pieces, although probably relatively few.

2. Within in the range of human hearing, the highest resolution possible is 16/44 and "Hi-res" is nothing more than an invented audiophile marketing term. As such, it can mean pretty much anything any company (or audiophile) wants it to mean. Upsampled 16/44 is therefore "Hi-res", even a terrible old cassette tape from the 1970's, where the music is barely recognizable due to signal loss and hiss, could legitimately be described as "Hi-res" if you digitized it at say 24/96. So, pretty much anything goes, which is great if you're a distributor who wants to be able to sell a wide catalogue of "hi-res" material (at inflated prices)!

3. Just about every mix of the last 60+ years is a downmix, that's what the term "to mix" means; to mix channels together to end up with fewer output channels. It makes logical sense to start with the mix going to the most output channels and work your way down to the fewest (2 channel stereo), as it takes far less time and gives more consistent results than the other way around. There maybe some special case exceptions but that's the workflow in the vast majority of cases. I would be very surprised (though I don't know for certain) if there were not at least some stereo versions that were just automated downmixes from the multichannel original. When I work on multichannel and stereo versions I start with an automated mixdown from the multichannel, very occasionally that works perfectly, most of the time it works perfectly with a number of tweaks and very occasionally an almost complete remix is necessary.

G


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## theaudiologist1

bigshot said:


> Most releases are DVD or blu-ray, not SACD any more.





gregorio said:


> 1. You seem to consistently overlook the fact that this is the Sound Science subforum and science does NOT allow exaggeration! We can be a bit more forgiving here than science allows but still your assertions either need to be accurate (NOT exaggerated) to start with or you need to qualify your assertion; as a guess, opinion or a possible/probable exaggeration.
> 1a. Rock music is ALWAYS compressed and with say classical music, a release with a dynamic range of less than about DR20 probably means it's compressed. Although of course that depends on the piece and it's orchestration, an unaccompanied piano obviously doesn't have the same dynamic range as a full symphony orchestra. The question is therefore not if it's compressed but compressed by how much and whether it's an inappropriate amount? And that depends on the genre and individual composition. Again, DR6 is not "just bad", it may represent an appropriate amount of compression for some pieces, although probably relatively few.
> 
> 2. Within in the range of human hearing, the highest resolution possible is 16/44 and "Hi-res" is nothing more than an invented audiophile marketing term. As such, it can mean pretty much anything any company (or audiophile) wants it to mean. Upsampled 16/44 is therefore "Hi-res", even a terrible old cassette tape from the 1970's, where the music is barely recognizable due to signal loss and hiss, could legitimately be described as "Hi-res" if you digitized it at say 24/96. So, pretty much anything goes, which is great if you're a distributor who wants to be able to sell a wide catalogue of "hi-res" material (at inflated prices)!
> ...


1)I wasn't against all compression. I was against BRICKWALLING A SONG TO DEATH. If a rock or metal release had a CD in the 80s-early 90s it usually had a dynamic range of 10+, and the same release "remastered" now gets a dynamic range of 5-6. THAT IS BRICKWALLING. It will usually sound worse and more fatuiging than the original. In the case of rock, if a song has a DR of less than 10 and the original had a high DR, it will sound overcompressed.

2) That is true, but that doesn't mean you can sell someone a 16/44.1 recording, upsample it, anc call it hi-res. All it takes is taking the WAV/FLAC/WV file in Audition or Audacity and checking the frequency cutoff.

3)Aren't the majority of recording out there done in stereo?


----------



## bigshot (Sep 4, 2021)

The majority of music is stereo.

I judge mixing and mastering by listening to it. A number can't describe what I am able to hear.


----------



## 71 dB

theaudiologist1 said:


> 1)I wasn't against all compression. I was against BRICKWALLING A SONG TO DEATH. If a rock or metal release had a CD in the 80s-early 90s it usually had a dynamic range of 10+, and the same release "remastered" now gets a dynamic range of 5-6. THAT IS BRICKWALLING. It will usually sound worse and more fatuiging than the original. In the case of rock, if a song has a DR of less than 10 and the original had a high DR, it will sound overcompressed.


I think pretty much everyone agree with this, but as mentioned before in these cases it is often wise to get one of the older and more dynamic releases. If anything, you have the choice between DR10+ and DR6, don't you?

However, newer music is likely to be produced with DR6 in mind rather than DR10+ and that's why it can sound good even at DR6. I think one of the secrets is to play with spectral content: The overall level of the music doesn't vary much, but what happens inside narrow frequency band does. So, if we analyse say the octave band 250-500 Hz,  it turns out to be much more dynamic than DR6. Or if we investigate individual tracks many of them have large dynamic range. It just that putting everything together in the mix results in DR6.

Also, rock seems to be the most brickwalled music genre meaning other (originally dynamic) genres are less brickwalled. 



theaudiologist1 said:


> 2) That is true, but that doesn't mean you can sell someone a 16/44.1 recording, upsample it, anc call it hi-res. All it takes is taking the WAV/FLAC/WV file in Audition or Audacity and checking the frequency cutoff.


They can do that, because placebo effect makes the upsampled version sound better for audiophools.



theaudiologist1 said:


> 3) Aren't the majority of recording out there done in stereo?


Depends on the genre I guess. Popular music tends to favour stereo while classical music productions do a lot of multichannel recordings.


----------



## Sonic Defender

gregorio said:


> 1. So does pretty much everyone.
> 
> 2. Any half decent BR player should do the job perfectly, assuming you're outputting the digital signal to a DAC. A BR player is obviously more future proof than a DVD player but you'd need an AVR to take advantage of all the audio formats that BR supports.
> 
> G


1. Yes I know, DACs have been able to be audibly transparent probably back in the mid 80s.

2. I am not interested in full audio format coverage, I would be only bothering with basic audio recordings, not worried about the whole coverage thing for formats.


----------



## bigshot

Most blu-ray players play most formats. The two that they may or may not play are SACD and DVD-A. Sony makes blu-ray players that play those audio formats.


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## Davesrose (Sep 6, 2021)

bigshot said:


> Most blu-ray players play most formats. The two that they may or may not play are SACD and DVD-A. Sony makes blu-ray players that play those audio formats.


Or if all you're concerned about is audio transports (for stereo DAC), there are DVD players that fit that bill.  I have an Oppo DVD player (upscaling HDMI DVD player) plugged into my DAC for handling CDs and SACDs.  Can also handle DVD-A obviously.  With SACD, it's converting to PCM for digital output.  I also have an Oppo blu-ray player attached to my home theater setup that's serving for multi-channel SACD and region free blu-ray (though it's a pain that I have to switch my HDMI option in my 4K receiver to not have extended color space).  Anyway, just bringing this up as used DVD players like my Oppo should be a cheap option for universal audio transport (unless you are one of the ones bent on digital DSD output).  I'm also a cinephile, and have found blu-ray or new UHD blu-ray players are more for audio formats carried via HDMI for multi-channel surround sound.


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## bigshot

A standard DVD player can't access the lossless DVD-A tracks, only the compressed standard ones.


----------



## gregorio

theaudiologist1 said:


> 1)I wasn't against all compression. I was against BRICKWALLING A SONG TO DEATH. [1a] In the case of rock, if a song has a DR of less than 10 and the original had a high DR, it will sound overcompressed.
> 
> 2) That is true, but that doesn't mean you can sell someone a 16/44.1 recording, upsample it, anc call it hi-res.
> 
> 3)Aren't the majority of recording out there done in stereo?


1. Who isn't against "brickwalling a song to death"? The problem is that's it's difficult to define what "to death" means, it's different for different songs/genres and in different listening environments.
1a. Now you're qualifying your assertion, "_in the case of rock_" and if the "_original had a high DR_", that's quite different from a blanket assertion that anything with DR10 or less is garbage.

2. Unfortunately, that's exactly what it means! There's no law or rule against it, the only thing that would likely stop them is the opinion of the vast majority of their customers but even that's not required or guaranteed.

3. No, virtually no recordings are recorded with just a stereo mic/pair, they are recorded with multiple mics which have to be mixed down to the distribution formats. Most commercial music is mixed down to stereo and no other distribution format but as mentioned, there are exceptions: Classical music is more likely to be mixed down to a multichannel format as well as stereo, EDM and other dance music is usually mixed to both stereo and mono and all genres of music destined for use in film and TV are usually multichannel.

G


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## Sonic Defender

bigshot said:


> A standard DVD player can't access the lossless DVD-A tracks, only the compressed standard ones.


Are there quite a few albums that have been put on DVD-A or are you mostly thinking about concerts and the like?


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## Sonic Defender

Davesrose said:


> Or if all you're concerned about is audio transports (for stereo DAC), there are DVD players that fit that bill.  I have an Oppo DVD player (upscaling HDMI DVD player) plugged into my DAC for handling CDs and SACDs.  Can also handle DVD-A obviously.  With SACD, it's converting to PCM for digital output.  I also have an Oppo blu-ray player attached to my home theater setup that's serving for multi-channel SACD and region free blu-ray (though it's a pain that I have to switch my HDMI option in my 4K receiver to not have extended color space).  Anyway, just bringing this up as used DVD players like my Oppo should be a cheap option for universal audio transport (unless you are one of the ones bent on digital DSD output).  I'm also a cinephile, and have found blu-ray or new UHD blu-ray players are more for audio formats carried via HDMI for multi-channel surround sound.


Thank you, my brother has one of the Oppo's, probably 10 years old now, but still pretty darn good sound, generally speaking. I know that it was one model below another and you had to step up to the $1000 model one above his at the time to get a Sabre DAC section. Are Oppo still making any of their players or are they all from the used, or old new stock marketplace?


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## bigshot (Sep 6, 2021)

DVD-A isn’t used a lot, but there are still multichannel mixes being released in the format. Steven Wilson’s prog rock multichannel releases are on DVD-A. (Yes, XTC, Jethro Tull, etc.)

Oppo has stopped making players. Sony makes a high end machine that is comparable.


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## Sonic Defender

bigshot said:


> DVD-A isn’t used a lot, but there are still multichannel mixes being released in the format. Steven Wilson’s prog rock multichannel releases are on DVD-A. (Yes, XTC, Jethro Tull, etc.)
> 
> Oppo has stopped making players. Sony makes a high end machine that is comparable.


Thank you. Cheers.


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## Sonic Defender

Sounds like it would be a huge waste of money to get a player unless movies were important to me and they really aren't. I can't imagine why anyone would pay the prices that are being asked to buy current movies. Some of the prices are outrageous and certainly for somebody like me who only watches movies once, maybe twice, far better to pay for single stream rentals.

Down the road I will look into things if my needs change.


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## 71 dB

Sonic Defender said:


> Sounds like it would be a huge waste of money to get a player unless movies were important to me and they really aren't. I can't imagine why anyone would pay the prices that are being asked to buy current movies. Some of the prices are outrageous and certainly for somebody like me who only watches movies once, maybe twice, far better to pay for single stream rentals.
> 
> Down the road I will look into things if my needs change.


What kind of prices are you talking about? Sure, new 4K UHD releases are very expensive, but I am into Blu-rays which to my eyes and on my small tv screen give perfect picture if the release is good. Blu-rays can also be pricy when released, but they tend to get cheaper fast, at least the movies targeted to masses. Some movies can be ridiculously expensive. This year I have spend about 30 euros (about $35) for two Blu-rays of French movies: _Les Parapluies de Cherbourg_ and _Irréversible_. On the other hand I buy other Blu-rays much cheap (e.g. 3 or 4 Blu-rays for 20 euros ($24)).


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## Sonic Defender (Sep 6, 2021)

71 dB said:


> What kind of prices are you talking about? Sure, new 4K UHD releases are very expensive, but I am into Blu-rays which to my eyes and on my small tv screen give perfect picture if the release is good. Blu-rays can also be pricy when released, but they tend to get cheaper fast, at least the movies targeted to masses. Some movies can be ridiculously expensive. This year I have spend about 30 euros (about $35) for two Blu-rays of French movies: _Les Parapluies de Cherbourg_ and _Irréversible_. On the other hand I buy other Blu-rays much cheap (e.g. 3 or 4 Blu-rays for 20 euros ($24)).


Way to expensive for me. I have zero interest in collecting movies and the few people I know who have gone down that rabbit hole have eventually regretted how much room they took and how much money went into the collection. I see people liquidating their once prized collections at yard sales for pennies on the dollar so it keeps my desire to collect movies down to zero.

This isn't directed at you 71 dB. I think that some people amass these big movie collections and build a theatre believing that friends and family want to come over and spend their time in a room watching a movie, probably played too loud and with the host pausing to alert them to this great effect coming up that they just have to see and hear, or asking what they think of the video and sound quality. The guests politely agree and on their drive home talk about how into the gear the host was and that next time they should suggest maybe a quieter evening where people actually talk.

For me, the very last thing I would ever want to do when I am in a social setting that isn't a movie theatre, is watch a movie. I can watch television and movies anytime and when I do get together with friends I prefer talking or playing cards, movie watching is not very social and it makes people quiet and sleepy. Again, these are just my opinions, nothing more, nothing less. An exception of course is being at home with your partner. Once you're no longer in that lusty phase what better way to kill off the time than staring at a screen until bed?


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## Davesrose

Sonic Defender said:


> Are Oppo still making any of their players or are they all from the used, or old new stock marketplace?


Oppo stopped making players, so if you're looking for a good price, you can find them used.  Pioneer Elite series has comparible specs for audiophile/videophile features that Oppo did (Oppo's uhd blu-ray player is still sought after in the used market).


bigshot said:


> A standard DVD player can't access the lossless DVD-A tracks, only the compressed standard ones.


Well my Oppo DVD player says it does support DVD-A.  It was made for audiophiles in mind (and also upscaling video to HDMI).


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## 71 dB

Sonic Defender said:


> Way to expensive for me. I have zero interest in collecting movies and the few people I know who have gone down that rabbit hole have eventually regretted how much room they took and how much money went into the collection. I see people liquidating their once prized collections at yard sales for pennies on the dollar so it keeps my desire to collect movies down to zero.
> 
> This isn't directed at you 71 dB. I think that some people amass these big movie collections and build a theatre believing that friends and family want to come over and spend their time in a room watching a movie, probably played too loud and with the host pausing to alert them to this great effect coming up that they just have to see and hear, or asking what they think of the video and sound quality. The guests politely agree and on their drive home talk about how into the gear the host was and that next time they should suggest maybe a quieter evening where people actually talk.
> 
> For me, the very last thing I would ever want to do when I am in a social setting that isn't a movie theatre, is watch a movie. I can watch television and movies anytime and when I do get together with friends I prefer talking or playing cards, movie watching is not very social and it makes people quiet and sleepy. Again, these are just my opinions, nothing more, nothing less. An exception of course is being at home with your partner. Once you're no longer in that lusty phase what better way to kill off the time than staring at a screen until bed?


I didn't mean so say everyone needs to collect movies. Each to their own. My point was that if someone collects movies on physical format, there are ways to keep the average price much lower. Personally I don't like paying for streaming services (too little control of the content), but I enjoy collecting physical media (both movies and music). I am very introverted and I always watch movies alone. I have my own weird movie taste.


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## megabigeye

Sonic Defender said:


> I have a DAC that is perfectly audibly transparent, Gustard X16 and I am now thinking about getting some kind of digital transport (DVD or Blu-ray) that can output the digital stream unadulterated which I could then pass through my DAC. Sadly I know nothing about such devices as I went right from CD transports to ripping my collection for computer based audio. Any suggestions for a DVD or Blu-ray player that I might consider? Almost exclusively for audio, not sure if I would ever use it for video, but probably a little.


I think any Blu-ray or DVD player with S/PDIF should work equally well.

I’ve been using a cheap-o Sony BDP-S370 hooked up to whatever DAC I happen to have on hand, and I think it sounds fabulous. My very unscientific way of testing was to plug other transports into the DAC’s other inputs (e.g., a Mac Mini into the USB port) and try to detect a difference. I couldn’t.
I was also recently surprised to realize that the RCA output also sounds pretty good, though my method of “testing” was even less scientifically rigorous than previous tests. Plug, unplug, repeat.
My only recommendation based on my experience would be to look for a player with fast load times and something that either runs cool or else is fanless. Some CDs can take a minute or so to load on my player, and the fan comes on after an hour or so of playback*.

*Now that I’m thinking about it, the fan may have been coming on because the player was in a drawer with other equipment that may have been making it warmer than normal.


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## bigshot (Sep 6, 2021)

Try and find a wide range of classic movies on streaming services… Physical media is just about the only option for cinephiles.

I’d rather spend money on books, music and movies than over priced audiophile stereo equipment, that's for sure!

When I put a screening room in my house, I learned the difference between a home theater and a TV set. Most people have TV sets in their family room and it's on a lot of the time, even when just one person is watching it. It becomes background noise. With a theater, especially projection, the lights go down and you watch a single movie or program. You don't binge watch or background with a theater. It's a different way of viewing movies than TV. When I have friends over, we have dinner and chat and have a glass of wine and enjoy each other's company. Then we go to the theater, watch a movie for 90 minutes, then after it's over, relax and chat about the movie. It's a nice way to spend an evening. I do that every weekend.


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## Sonic Defender

bigshot said:


> Try and find a wide range of classic movies on streaming services… Physical media is just about the only option for cinephiles.
> 
> I’d rather spend money on books, music and movies than over priced audiophile stereo equipment, that's for sure!
> 
> When I put a screening room in my house, I learned the difference between a home theater and a TV set. Most people have TV sets in their family room and it's on a lot of the time, even when just one person is watching it. It becomes background noise. With a theater, especially projection, the lights go down and you watch a single movie or program. You don't binge watch or background with a theater. It's a different way of viewing movies than TV. When I have friends over, we have dinner and chat and have a glass of wine and enjoy each other's company. Then we go to the theater, watch a movie for 90 minutes, then after it's over, relax and chat about the movie. It's a nice way to spend an evening. I do that every weekend.


That actually sounds like a very nice way to go about it, and I do also get the distinction that you are drawing between a TV based setup and a projection. I can see that making a significant difference. I also like that you include the social component after of discussing, far more grown up sounding than sadly the majority of people I have met that have invested in their home theatre. They become obsessed with it and they seem to want everybody who ever enters their home for a social gathering to want to make their big loud cinema setup the centre of everything, and then talk about the minutia of formats and processing until your ready to either fall asleep or start cutting yourself.

Seriously, I know I am exaggerating for effect, but not by much. To all people remember, whatever your obsession with your rig just don't be that guy that can't stop talking about it. No matter how amazing it is, the VAST majority of people will really not care a great deal, some would rather watch grass grow than listen to you explain your system and a very small minority will actually care. It pained me to realize that absolutely nobody in my social circle gave a rats ass about audio reproduction. They couldn't care less if the music at a party comes from a laptop's built in speaker, screaming with distortion or a tricked out 2.2 system that digs deep and is glorious to hear and behold. So sad. I would love to make a friend who is into audio. My brother is, but only to talk about it, and even then only briefly, never to sit down and listen. He would rather talk about stuff in the Manosphere (groan).


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## Davesrose (Sep 6, 2021)

Sonic Defender said:


> That actually sounds like a very nice way to go about it, and I do also get the distinction that you are drawing between a TV based setup and a projection. I can see that making a significant difference. I also like that you include the social component after of discussing, far more grown up sounding than sadly the majority of people I have met that have invested in their home theatre. They become obsessed with it and they seem to want everybody who ever enters their home for a social gathering to want to make their big loud cinema setup the centre of everything, and then talk about the minutia of formats and processing until your ready to either fall asleep or start cutting yourself.
> 
> Seriously, I know I am exaggerating for effect, but not by much. To all people remember, whatever your obsession with your rig just don't be that guy that can't stop talking about it. No matter how amazing it is, the VAST majority of people will really not care a great deal, some would rather watch grass grow than listen to you explain your system and a very small minority will actually care. It pained me to realize that absolutely nobody in my social circle gave a rats ass about audio reproduction. They couldn't care less if the music at a party comes from a laptop's built in speaker, screaming with distortion or a tricked out 2.2 system that digs deep and is glorious to hear and behold. So sad. I would love to make a friend who is into audio. My brother is, but only to talk about it, and even then only briefly, never to sit down and listen. He would rather talk about stuff in the Manosphere (groan).


I'm more of that obsessed guy about technology, but I won't "nerd out" unless that person is asking about 4K, visual effects, or something cinematography related (some of the items I'm pretty knowledgable about).  I haven't considered a projection system, but have an OLED TV for superior picture quality and is more conducive for my townhouse.  It takes up a good viewing angle when I and a friend are watching in my two recliners watching a movie (and my Atmos speaker setup is calibrated to the seating area of my recliners).  If entertaining for a party, I've got a couch and other chairs that people will sit and I can have music playing through my speakers.  For someone first viewing on my setup, they may praise the image quality or ask me about my speaker setup.  As to why I spent some time considering gear, I see it as a means for getting the best experience out of a movie: but I appreciate the inherit merits of the movie first.  With 4K, we're now at a threshold where older movies are being restored in a picture and sound format that can surpass the original screening (where with film projection, there were limits with the film negative and also the print getting dust and scratches).

For movie collecting, if you're collecting a range of movies then you might have to be format agnostic.  I do like that some classics are making it to 4K (as that does support the best resolving power 35mm film can approach).  However, some older movies made it to DVD and went out of print: also some with blu-ray.  Now you also have independent studios that can more easily distribute movies via streaming.  My dad amassed a huge collection of DVDs and blu-rays, which he's almost done ripping them to hard drives to serve up on Plex.  I've started using Plex as well for my collection of movies I'm saving to hard drive.


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## Sonic Defender

Davesrose said:


> I'm more of that obsessed guy about technology, but I won't "nerd out" unless that person is asking about 4K, visual effects, or something cinematography related (some of the items I'm pretty knowledgable about).  I haven't considered a projection system, but have an OLED TV for superior picture quality and is more conducive for my townhouse.  It takes up a good viewing angle when I and a friend are watching in my two recliners watching a movie (and my Atmos speaker setup is calibrated to the seating area of my recliners).  If entertaining for a party, I've got a couch and other chairs that people will sit and I can have music playing through my speakers.  For someone first viewing on my setup, they may praise the image quality or ask me about my speaker setup.  As to why I spent some time considering gear, I see it as a means for getting the best experience out of a movie: but I appreciate the inherit merits of the movie first.  With 4K, we're now at a threshold where older movies are being restored in a picture and sound format that can surpass the original screening (where with film projection, there were limits with the film negative and also the print getting dust and scratches).
> 
> For movie collecting, if you're collecting a range of movies then you might have to be format agnostic.  I do like that some classics are making it to 4K (as that does support the best resolving power 35mm film can approach).  However, some older movies made it to DVD and went out of print: also some with blu-ray.  Now you also have independent studios that can more easily distribute movies via streaming.


Sounds great, absolutely if people ask then answer, but are you also ok at reading people's body and non-verbal cues? Most people ask to be polite, and may be interested to a point, but not want to go down the rabbit hole and have it all explained with the pros and cons. Certainly some do, but if there is mixed company, being the host that focuses on the exception guest who cares and takes the gathering down a path that the majority may be less interested in is worth considering. I only say this as I have seen so many potentially lovely gathers killed by a host with an agenda that does not really revolve around the guests. The hosts wants the guests to love what they love and are sure that if they just force things on the guests they will see what they have been missing.

Sometimes I am sure they may be right, but really, I have always found that the best gatherings are when technology is really in the background and the foreground is the people and the conversation. Music should be quietly playing in the background, preferably with the system out of sight (although many of us including myself don't have enough room). Ideally my stereo would be in-wall and out of sight, even though I love the look of all components, but for entertaining, nope, I would focus on the room and the people, not the technology. End of rant.


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## Davesrose

Sonic Defender said:


> Sounds great, absolutely if people ask then answer, but are you also ok at reading people's body and non-verbal cues? Most people ask to be polite, and may be interested to a point, but not want to go down the rabbit hole and have it all explained with the pros and cons. Certainly some do, but if there is mixed company, being the host that focuses on the exception guest who cares and takes the gathering down a path that the majority may be less interested in is worth considering. I only say this as I have seen so many potentially lovely gathers killed by a host with an agenda that does not really revolve around the guests. The hosts wants the guests to love what they love and are sure that if they just force things on the guests they will see what they have been missing.
> 
> Sometimes I am sure they may be right, but really, I have always found that the best gatherings are when technology is really in the background and the foreground is the people and the conversation. Music should be quietly playing in the background, preferably with the system out of sight (although many of us including myself don't have enough room). Ideally my stereo would be in-wall and out of sight, even though I love the look of all components, but for entertaining, nope, I would focus on the room and the people, not the technology. End of rant.


Yes, my default hosting people is not talk about my sound system or OLED TV.  My DVD cabinets also have doors, so they don't see my physical media.  You have a point about inconspicuous speakers if you don't ever want to have a conversation get into equipment.  I have tower speakers, and cherry and black surrounds and height speakers...they do take up room, but I try to treat it like furniture (especially my large subwoofer).  I have had instances of people being interested in seeing concert blu-rays. So sometimes in that instance they'll indicate how they hear the benefit of a surround system vs stereo or portable speakers.  Guys tend to be more about gear and want to talk about it.  I don't talk tech with the women I date.  But when I first got my 7.1 speaker setup, my girlfriend at the time said that even though she had doubts about the size of speakers, she could really appreciate the sound quality.


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## Sonic Defender

Davesrose said:


> Yes, my default hosting people is not talk about my sound system or OLED TV.  My DVD cabinets also have doors, so they don't see my physical media.  You have a point about inconspicuous speakers if you don't ever want to have a conversation get into equipment.  I have tower speakers, and cherry and black surrounds and height speakers...they do take up room, but I try to treat it like furniture (especially my large subwoofer).  I have had instances of people being interested in seeing concert blu-rays. So sometimes in that instance they'll indicate how they hear the benefit of a surround system vs stereo or portable speakers.  Guys tend to be more about gear and want to talk about it.  I don't talk tech with the women I date.  But when I first got my 7.1 speaker setup, my girlfriend at the time said that even though she had doubts about the size of speakers, she could really appreciate the sound quality.


Awesome to hear, clearly you are a socially aware person. I also try to treat my somewhat large tower speakers (and dual SVS subs) as furniture as best possible. It isn't like the hosts has no rights, it is our homes after all so we do also count. My point was about the host who is "guest blind" and in their mind scripts their perfect evening without even really considering what the guests perfect evening is, which is the job and hopefully the pleasure of the host. Sounds like you get it.


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## bigshot

I think the problem isn’t that your friends’ home theaters are boring. Your friends are boring!


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## Davesrose

Sonic Defender said:


> Awesome to hear, clearly you are a socially aware person. I also try to treat my somewhat large tower speakers (and dual SVS subs) as furniture as best possible. It isn't like the hosts has no rights, it is our homes after all so we do also count. My point was about the host who is "guest blind" and in their mind scripts their perfect evening without even really considering what the guests perfect evening is, which is the job and hopefully the pleasure of the host. Sounds like you get it.


Yes, and you have a good point about not focusing on your sound setup with general socializing.  I think I also learned that at an early age since I was into unpopular subjects in high school: computers, photography, and model kit making (and visual effects with movies).  With adulthood, its included beer brewing.  So my friends know to seek me out for their computer help, and I'll get into some of the science of brewing if someone is asking about fermentation.  Then I might get into water chemistry and grain makeup if they are asking about differences with beer styles.


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## bigshot

I cook and I bring my friends into the kitchen and cook for them. No one complains because the food is good!


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## castleofargh




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## theaudiologist1

Are DVD-A's still being made? I'd much rather have DVD-A over DRM-filled hard-to-rip Bluray's.


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## bigshot (Sep 11, 2021)

Yes, Steven Wilson who produces the lion's share of multichannel mixes right now prefers the DVD-A format. Not sure why.

Also, if you collect multichannel music, you're dealing with out of print stuff in all kinds of formats. So being able to play anything is a distinct advantage.

It's possible to rip blu-rays and even SACDs.


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## tinkererGCH

excellent initial post with great explanations. Now my simple question is if it makes sense using a DAC AMP Stack for the laptop when listening to CD-rips in 16bit? or is it enough to just plug in the headphone to the audio port of the laptop?


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## bigshot

That depends on the impedance and sensitivity of your headphones. They might benefit from amping.


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## 71 dB

bigshot said:


> That depends on the impedance and sensitivity of your headphones. They might benefit from amping.


Plus the output impedance of the laptop.


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## ostuni

tinkererGCH said:


> excellent initial post with great explanations. Now my simple question is if it makes sense using a DAC AMP Stack for the laptop when listening to CD-rips in 16bit? or is it enough to just plug in the headphone to the audio port of the laptop?


With my 300 ohm HD800s, I need a certified MQA dac amp to unwrap MQA from Tidal.


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## tarikuz

Please check:

https://cdn.shopify.com/s/files/1/0321/7609/files/ASPECTS_OF_SAMPLING.pdf?267

Enjoy!


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## gregorio

tarikuz said:


> Please check:
> 
> https://cdn.shopify.com/s/files/1/0321/7609/files/ASPECTS_OF_SAMPLING.pdf?267
> 
> Enjoy!


A very decent write-up, considering it’s nearly 30 years old! The audibility of jitter  question was put to bed within about 5 years or so of the publication of the document but the rest of it is still accurate. 

G


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